ReleaseNotes8:Firmware: Difference between revisions

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= V8 Hotfix  1 (10-80500.01) =
= V8 Hotfix  6 (80500.20) =
Changes included in Version 8 hotfix1
Changes included in Version 8 hotfix6
[http://mantis.innovaphone.com/view.php?id=47564 Definition]
[http://mantis.innovaphone.com/view.php?id=54972 Definition]


== New Features ==
== New Features ==
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=== Show presence note (if available) instead of activity during ringback  ===
=== Modified interface for OEM password complexity ===


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Problem: When calling another PBX user, it's presence activity is displayed at the callers phone screen. Presenting called user's presence note is more appropriate.<br/><br/>Solution: Prefer 'note' over 'activity'.<br/><br/>Files: app_disp.cpp<br/><br/>Products affected: H.323 Phones<br/><br/>Risk: No risk.<!---->
OEMs can now implement a module for checking password complexity<!---->
''Status:''
files: <br/>./common/lib/lib.mak <br/>./common/interface/interface.mak <br/>./common/interface/pwd_complex_api.h <br/>./common/interface/pwd_complex_api.cpp <br/>./ascom/pwd_complex/pwd_complex.h <br/>./ascom/pwd_complex/pwd_complex.cpp <br/>./box/command/command.h <br/>./box/command/command.cpp <br/>./dect/users/dectusers.cpp <br/><br/>
      
      
=== Config download must be supressed in phone training mode ===
=== OEM password complexity for Kerberos users ===


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problem: Config download is not supressed in phone training mode.<br/> <br/>solution: fix in code<br/> <br/>files: phone_save_hdr.xml<br/><br/>products: all phones<br/><br/>risks: None<br/><!---->
The Kerberos module can now check the complexity of user passwords if this is implemented by the OEM software.<!---->
''Status:''
''Status:''
checked in to 9.00,8.00,09-80500
files:<br/>kerberos_db.cpp
      
      
=== provide uptime and local time of trap in debug log ===
=== Simplified administration UI for some OEMS ===


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problem: sometimes it's helpful to know at which time of day and how long after boot a trap occured<br/><br/>solution: fix in code<br/> <br/>files: box.cpp<br/><br/>products: all<br/><br/>risks: None<br/><!---->
Some items in the adminstration user interface can now be hidden by setting special xml-modes (admin-basic,admin-advanced).<!---->
''Status:''
''Status:''
checked in to 9.00,8.00,09-80500
files:<br/>- ./dect/users/dectusers.cpp<br/>- ./dect/master/dectmaster.cpp<br/>- ./platform/platform.mak<br/>- ./platform/asc_diagnostics_basic.xml<br/>- ./platform/asc_diagnostics_hdr_basic.xml<br/>- ./platform/dect_hdr.xml<br/>- ./platform/eth0_hdr.xml<br/>- ./platform/left_menu.xml<br/>- ./box/httpfiles/reset_hdr.xml<br/>- ./common/platform/ip1201.cpp<br/>- ./box/command/command.h<br/>- ./box/command/command.cpp
      
      
=== PBX Broadcast Conference Option ===
=== Hide some pages and items on admin UI while OEM provisioning is running ===


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problem: It should be possible to configure if the last remaining user in a conference call should be disconnected or not.<br/><br/>solution: New configuration option implemented.<br/><br/>files: pbx_bc_conf.h, pbx_bc_conf.cpp, pbx_edit_bc_conf.xsl.<br/><br/>products affected: All devices with PBX.<br/><br/>risk: Minimal risk of collateral damage. <!---->
While the provisioning module of an OEM is active, special xml-modes are set that can be used to hide items from the administration interface.<!---->
''Status:''
files:<br/>./ascom/httpfiles/asc_ntp.xsl<br/>./ascom/httpfiles/asc_dectfty.xsl<br/>./common/platform/ip1201.h<br/>./common/platform/ip1201.cpp<br/>./common/service/ntp/ntp.cpp<br/>./dect/fty/dectfty.cpp
      
      
=== IP-DECT OEM user database import/export ===
=== IP-DECT OEM location monitor function change ===


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problem: New functions for OEM import/export of user data like export filter, ipei checksum or error messages.<br/><br/>solution: Functionality implemented.<br/><br/>files: dectusers_if.h, dectusers.h, dectusers.cpp, OEM xsl files.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
For OEM modules the location monitor is changed.<!---->
''Status:''
dectmaster.cpp
      
      
=== IP-DECT handset product id/software version ===
=== DTMF feature call completion can be also used for no response ===


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problem: There is no possibility to see the handset product id and software version.<br/><br/>solution: Tool-tip with the product id and software version of the handset has been added for the IPEI item in the GUI user list. It will be available after restart of the handset.<br/><br/>files: dectmaster.cpp, dectradio.cpp, dect_users_right.xsl(OEM).<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
The feature is not only usable after a busy call, but also after a call with no response.<!---->
      
      
=== Presence: Added overlay activity attribute ===
=== Update client option for short URL ===


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Problem: External applications want to set/reset "on-the-phone" activity for a PBX user.<br/><br/>Solution: Added overlay activity attribute for each PBX user.<br/><br/>Files: pbx.cpp/h<br/><br/>Products affected: PBX<br/><br/>Risk: No risk. <!---->
For OEM http server the update client should not append additional options to the update server URL.<!---->
''Status:''
update.h, update.cpp
      
      
=== Master PBX to obtain licenses from another Master ===
=== SIP: Detect remote party identity change ===


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problem: There are configurations in which centralized licensing is desired but otherwise independent Master PBXs are needed.<br/><br/>solution: Configuration option added to allow a slave to register at a master to obtain licenses, but act as master in all other respects<br/><br/>files: pbx.cpp, pbx.h, pbx_general.xsl<br/><br/>products: all with PBX<br/><br/>risks: Minimal<!---->
Remote party update did not work in all cases:<br/>If initial INVITE got no identity header, but re-INVITE contains identity header.<!---->
''Status:''
sip.cpp/h
      
      
== Bug Fixes ==
=== IP-DECT OEM configuration options for registration speed ===
 
 
   
=== SIP: Fax and audio offer was rejected with 488 ===


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Problem: A combined SDP offer (fax and audio) was rejected with 488.<br/><br/>Solution: Answer with 200/OK and provide audio answer.<br/><br/>Files: sip.cpp/h<br/><br/>Products affected: SIP gateways<br/><br/>Risk: No risk. <!---->
For an OEM PBX it is necessary to configure the user's registration speed to this PBX. Used only in the OEM DECT device.<!---->
''Status:''
dectmaster.h, dectmaster,cpp.
      
      
=== Potential trap accessing NULL pointer in PBX Waiting Object ===
=== SIP: Added Microsoft propriatary extension "ms-acceptedby" for OCS compatibility ===


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problem: There is a small chance of a NULL pointer access trap when doing confuguration changes on a Waiting Queue object right when a call is cleared<br/><br/>solution: Check for NULL pointer added<br/><br/>files: pbx_wait.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
A forked call that is accepty elsewhere is counted as "missed call" by OCS unless Microsoft specific extension is add to Reason header.<br/> Reason: SIP;cause=200;text="OK";ms-acceptedby="sip:user@domain.com"<br/>According to [MS-SIPRE].pdf<!---->
      
      
=== AD Replication. Only 10x In-Maps per Source Attribute Configurable ===
=== A DHCP client with  "/keep on" should not fall back to dicsover mode if the lease is due ===


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Problem: Only 10x In-Maps per Source Attribute Configurable<br/><br/>Solution: Adjust to 40. Form submit method now POST (was GET).<br/><br/>Files: ldapmap.cpp, ldaprep.xsl<br/><br/>Products affected: PBX products<br/><br/>Risk: No risk. <!---->
"/keep on" forces reusing the remembered lease if no DHCP server is responding after boot. But if the server failed to respond to the final rebind request for a regularly obtained lease a new recovery was started.<br/>Now in this case the lease is used further, a request for the lease and an ARP requests to check if the IP address is not assigned to another device are sent in regular intervals. <!---->
      
      
=== SIP: INVITE rejected with 407 ===
=== SIP: Hide product information in reject responses ===


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Problem: Registered SIP interfaces reject incoming calls with 407 if the INVITE comes from a remote source addr/port that doesn't match addr/port where the REGISTER was sent to.<br/><br/>Solution: Do not check remote source port.<br/><br/>Files: siptrans.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk.<!---->
Don't be kind to SIP scan tools.<!---->
''Status:''
siptrans.cpp
      
      
=== Do not allow special characters for Kerberos realm names ===
=== Include modes into configuration page of update client ===


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Problem: On the General/Kerberos page the name of the server realm was not checked. Only domain style names should be allowed.<br/><br/>Solution: Allow only A-Z a-z 0-9 . -<br/><br/>Files: kerberos_db.cpp<br/><br/>Products affected: Gateways<br/><br/>Risk: No risk<!---->
Needed for OEM specific XSL.<!---->
      
      
=== malloc must always run disabled ===
=== Phone: Problems with 'Presence' Fkey ===


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problem: usually malloc is called only from disabled state. in some extremely rare cases (for example before a DRAM upload) it is called from enabled state and then an interrupt may cause assignment of the same memory chunk to different callers.<br/> <br/>solution: fix in code<br/> <br/>files: os.cpp<br/><br/>products: all products<br/><br/>risks: None<!---->
Presence Fkey requires working presence subscription.<br/>Presence subscription may fail from time to time due to several reasons.<br/>Reliable re-establishment is required.<!---->
''Status:''
''Status:''
checked in to 9.00,8.00,09-80500
phonesig.cpp
      
      
=== reduce phone firmware size by excluding unused LDAP components ===
== Bug Fixes ==
 
 
   
=== SIP: Media-negotiation after call transfer failed (no audio) ===


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problem: references from flashdir module to fdirui object (flash dir user interface) force the inclusion of objects which are not used in the phone.<br/> <br/>solution: conditional compilation of flashdirui.cpp to prevent references, conditional linking of objects<br/> <br/>files: common/service/ldap/flashdir.cpp, common/service/ldap/ldap.mak, phone_inca.mak, ip72.mak, phone_coldfire.mak<br/><br/>products: all products<br/><br/>risks: None<br/><!---->
Re-negotiation after call transfer failed.<br/>Results into no-audio condition.<!---->
''Status:''
''Status:''
checked in to 9.00,8.00,09-80500
sip.cpp/h
      
      
=== SIP: Handling of weird simple-message-summary ===
=== send busy tone from PBX dtmf object for not working cf with diversion filter ===


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Problem: "Messages-Waiting:yes;Voice-Message:/0" turned MWI off. "Messages-Waiting:no;Voice-Message:8/0" turned MWI on.<br/><br/>Solution: "Messages-Waiting:yes;Voice-Message:/0" turned MWI on. "Messages-Waiting:no;Voice-Message:8/0" turned MWI off.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP Phones<br/><br/>Risk: No risk.<!---->
If a diversion filter is set on a user and the dialed diversion to the pbx dtmf object is not allowed, a busy tone and a reject cause is now sent by the dtmf object.<!---->
      
      
=== incorrect pointer assignment in submit_config of dtmf/icp object and false configuration possibility ===
=== IP-DECT Master call list OEM link and call state ===


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problem: a) local buffer assigned to a given pointer<br/>         b) it was possible to configure just one of two needed e164 and if configured just one (which makes sense for pickup), the code was not correctly shown as enabled in configuration window.<br/><br/>solution: fix in code<br/> <br/>files: pbx_dtmf.h, pbx_dtmf.cpp, pbx_edit_dtmf-ctrl.xsl, pbx_icp.cpp<br/><br/>products: all pbx products<br/><br/>risks: None <!---->
For OEM devices the call clear link doesn't work.<br/>Call state for the outgoing party is shown as "off-hook".<!---->
''Status:''
dectmaster_call.xsl, dectmaster.cpp
      
      
=== permit to control the display format of names from local/PBX directory the same way as for external directories ===
=== No Media event was generated even everything was normal for unanswered CC exec on IP-DECT ===


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problem: some users want to control the name display format for inbound and outbound calls separately and to reorder/omit parts of a name. this already works for external directories but names from local or PBX directory were not displayed if the first name attribute was not configured for display because such names have only one, the 'cn' attribute.<br/><br/>solution: by default the full 'cn' is displayed for entries from local or PBX directory. if the format string starts with an asterisk ('*') 'cn' is tokenized and the tokens are ordered according to the requested format.<br/>  config add PHONE APP /name-display-in <format-in> /name-display-out <format-out>"<br/>'format-...' selects the name attributes to be displayed and their order.<br/>The default format is "123", i.e. all names are displayed as configured.<br/>"3" displays only the third attribute of a name from an external directory but the complete 'cn' from local or PBX directory.<br/>"*3" displays only the third attribute of a name from an external directory and only the third token (if any) of a 'cn' from local or PBX directory.<br/> <br/>files: app_call.cpp<br/><br/>products: all phones<br/><br/>risks: None<br/><!---->
Could happen for other traffic cases as well like rejected CC exec<!---->
''Status:''
''Status:''
checked in to 9.00,8.00,09-80500
dectradio.cpp, media.cpp
      
      
=== IP22 Interop with devices that do not support T38 redundancy and retransmits ===
=== Point to Multipoint ISDN Maps need to set Type ISDN for CGPN-Out Map ===


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problem: Some fax gateways (e.g. old avaya equipment) do not accept T.38 packet with redundancy or do no accept resent packets with the same sequence number<br/><br/>solution: DSP config added, use http://addr/AC-DSP0/info.xml?xsl=dsp.xsl to edit the settings.<br/><br/>files: ac_dsp3.cpp ac_dsp3.h ac_dsp3.mak dsp.xsl<br/>products: IP2x IP30x<br/>risks: Low <!---->
If not the mapping does not work for some networks and always the default number is used for outgoing calls as calling party number<!---->
''Status:''
gk.cpp
      
      
=== prevent creation of pbx dtmf object with char '#' in long name ===
=== SIP: Digest authentication is rejected if username contains non-us-ascii characters ===


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problem: it was possible to create a dtmf object with long name dtmf#join_group. If one then creates another dtmf object with long name dtmf and feature code join group enabled, it traps, because the object tries to create another user with dtmf#join_group as long name without check.<br/><br/>solution: disallow char '#' in long name of dtmf object<br/> <br/>files: pbx_dtmf.cpp<br/><br/>products: all pbx devices<br/><br/>risks: None<!---->
Digest authentication is rejected if username contains non-us-ascii characters.<br/>Expected special characters to be URL encoded, but most clients send it UTF8 encoded.<!---->
      
      
=== IP-DECT idle display update ===
=== H.323: Cause received with PROGRESS message got lost ===


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problem: Idle display update does not work for DECT handsets.<br/><br/>solution: Functionality implemented, but it must be enabled over the GUI and must not be used if foreign handsets are used.<br/><br/>files: dectmaster.h, dectmaster.cpp, dectmaster.xsl.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
This could result in calls to busy subscribers in a QSIG PBX to terminate with "recovery on time expiry" instead of "user busy"<!---->
''Status:''
h323sig.cpp
      
      
=== IP-DECT GUI Authentication Code ===
=== SIP: Outgoing call (early, not connected) was not canceled (sometimes) on ISDN interworking scenario ===


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problem: The configuration option for the System Authentication Code is not shown.<br/><br/>solution: Fixed.<br/><br/>files: dectusers.xsl.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. Only GUI change.<!---->
An incoming DISCONNECT with progress indicator did not caused the outgoing SIP call to be canceled.<!---->
''Status:''
sip.cpp
      
      
=== IP-DECT RTP stream of second hold call ===
=== Gateway: divertingLeg2 was not passed in some cases ===


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problem: The RTP stream of the second call is not stopped if the call is hold and an unattended call transfer is initiated.<br/><br/>solution: RTP stop event added.<br/><br/>files: dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
divertingLeg2 got lost during re-routing in Gateway.<br/>E.g. routing each call over TONE caused the divertingLeg2 to disappear.<!---->
      
      
=== dtmf feature code set presence used wrong argument to toggle the mobility ===
=== Webdav: Handling of failed TCP when writing to file ===


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problem: the feature code argument for toggling mobility was swapped<br/><br/>solution: use correct argument index<br/> <br/>files: pbx_dtmf.cpp<br/><br/>products: all pbx devices<br/><br/>risks: None <!---->
Webdav client needs handling of TCP error when writing to file<!---->
      
      
=== SIP: Handling of transfer to different ip address ===
=== TEL interface: '#11' not callable if feature codes enabled ===


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Problem: MS Exchange transfers fax calls to external fax servers with REFER.<br/><br/>Solution: Follow this transfer and send new INVITE to destination address.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk<!---->
If feature codes are enabled for a TEL interface, the number '#11' without anything else can not be dialled.<br/>To fix please submit gateway's general page with the OK button or do a factory reset.<!---->
''Status:''
config.h, relay_general.xsl
      
      
=== Missing Tooltips in web ui for licenses ===
=== ARP requests/replies returned to the sender should be ignored ===


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Problem: Missing Tooltips in web ui for licenses.<br/><br/>Solution: Added tooltips.<br/><br/>Files: license.xsl<br/><br/>Products affected: All Gateway/PBX devices<br/><br/>Risk: No risk.<!---->
It was observed that in WLAN environments broadcasted ARP requests/replies may be received by the sender again. This results in some problems when DHCP checks if an IP address is not used by another device via ARP. Now returned requests/replies are simply ignored. <!---->
      
      
=== IP-DECT OEM module software update ===
=== T.38 doesnt work if the call is transferred from a IP-Phone to a fax device ===


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problem: There are new software versions of the OEM software modules: BMC interface software, MSF module and Skinny protocol.<br/><br/>solution: OEM modules replaced.<br/><br/>files: DECT files, MSF files, Skinny files, config.h, fty.h, fty.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Normal risk of collateral damage. Updated from improved OEM branch.<!---->
Affects IP2x IP30x fax gateways, the ipphone needs no update <!---->
''Status:''
ac_dsp3.cpp<br/>v7:<br/>ac494004.h <br/>ac498004.h
      
      
=== SIP: Problem with media negotiation after 488 ===
=== DECT: Trap while initiating blind transfer when using SIP as PBX protocol ===


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|[http://mantis.innovaphone.com/view.php?id=55581 55581]
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Problem: After re-INVITE client transaction was rejected (e.g. 488) the next re-INVITE was not send. May result in one-way-audio.<br/><br/>Solution: Cleanup when handling reject for re-INVITE.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk<!---->
0:0246:363:3 - GK-CALL free error 9481a58c<br/>0:0246:363:4 - last free=DECTMASTER-RADIO len=6<br/>0:0246:363:4 - caller=0x943796d0<br/>0:0246:363:4 - HEXDUMP<br/>     00000000 - 05 80 38 30  31 31                                  ..8011          <br/>0:0246:363:4 - BUFFER-FREE: obj at 0x9481a574 inconsistent<br/>0:0246:363:4 - HEXDUMP<br/><br/>Fixed in dectmaster.cpp<!---->
      
      
=== use flashman erase on "reset to factory defaults" ===
=== Kerberos problem with encrypted password data containing null bytes ===


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|[http://mantis.innovaphone.com/view.php?id=55692 55692]
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problem: when resetting telephones to factory defaults, first default registration survives the reset<br/><br/>solution: use flashman erase now<br/><br/>files: phone/admin/phone_admin.cpp<br/><br/>products: all telephones<br/><br/>risks: none known<br/><br/><br/><br/><!---->
Encrypted Kerberos passwords that are stored using LDAP may contain null bytes. Therefore they must not be handled as strings but as binary data when reading them.<!---->
''Status:''
files: kerberos_ldap.cpp
      
      
=== IP-DECT MSF module login ===
=== Phone: Make PBX-initiated calls don't look like transferred calls ===


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|}
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problem: It is not possible to login to the MSF module.<br/><br/>solution: Function signature changed with the new MSF module version.<br/><br/>files: dectmsf.h, dectmsf.cpp, telnet.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. Updated from improved OEM branch. <!---->
Do not send CT_SETUP.<!---->
      
      
=== editing phone user config with IE failed for users with non-ascii chars in long name ===
=== "Join Group" function key lost state after a PBX reboot when the phone config was stored on the PBX ===


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|[http://mantis.innovaphone.com/view.php?id=55790 55790]
|}
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problem: the long user name was patched latin1 encoded into a xml file with encoding="utf-8"<br/> <br/>solution: fix in code<br/> <br/>files: pbx_phone.cpp<br/><br/>products: all PBX<br/><br/>risks: None<br/><!---->
The Join Group function key lost it's state and did not work anymore after a PBX reset because the the phone config sent by the PBX after reregistration was not evaluated at the phone again. <!---->
''Status:''
checked in to 9.00,8.00,09-80500
      
      
=== ip2000/ip6000 - start ETH2 (virtual network connction for Linux) only when Linux is enabled ===
=== flash variables may get lost after reboot (because  of an earlier  trap in the critical phase of flash garbage collection) ===


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problem: ETH2 was always started with the preconfigured fixed address 192.168.2.1/24. routing problems may occur if this network is used otherwise already. <br/> <br/>solution: start start ETH2 only when Linux is enabled, do not preconfigure an IP address on ETH2 <br/> <br/>files: linux_eth_drv.cpp, config.h<br/><br/>products: ip2000/ip6000<br/><br/>risks: None <!---->
Two valid segments bearing the same data are left back when a fragmented segment is compacted into a new one and the box traps after the new segment has been validated but before the old segment has been marked invalid.<br/>Because of a wrong comparison this situation was not resolved after reboot. Instead of deleting one of the segments the new segment was used until completely filled. Therafter all further allocations failed. This situation could only be cleared by a reset to factory defaults.<br/>Now, if the flash user is permitted to use only one segment (for example VARS on most boxes) the old segment is invalidated and the new compacted segment remains. If the flash user is permitted to use more segments (for example LDAP) the new segment is invalidated because it's not known which of the old segments was compacted.<br/> <br/> <!---->
''Status:''
checked in to 8.00,09-80500,9.00
      
      
=== Reject of ectLinkIdRequest not handled ===
=== PBX potential trap when parsing SOAP XML ===


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|[http://mantis.innovaphone.com/view.php?id=55812 55812]
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Problem: When trying to pass a call transfer of two ISDN calls to the ISDN network by means of ECT ("External Transfer"), a reject was not handled properly.<br/><br/>Solution: Decode an provide error code to gateway.<br/><br/>Files: fty.h/cpp q950.cpp relay.cpp<br/><br/>Products affected: ISDN gateways<br/><br/>Risk: No risk.<!---->
No child element found in SOAP XML<br/><!---->
      
      
=== SIP: Media negotiation problem at Alcatel Omni PCX ===
=== Possible buffer overrun when reading/writing fat volumn id ===


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|[http://mantis.innovaphone.com/view.php?id=55858 55858]
|}
|}
Problem: Several provisional responses with changing remote RTP addresses may cause RTP to be sent to wrong destination.<br/><br/>Solution: Fix handling of updated SDP answers.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk.<!---->
There was a possible buffer overrun when reading/writing the fat volumn id.<!---->
      
      
=== default language setting from bootcode must be respected when setting phone default configuration ===
=== SIP: Display name contained bad characters in some cases ===


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problem: the default language for texts displayed on the phone may be set at manufacturing time. it's saved in the bootcode and evaluated at boot time. But because of some changes in config processing it was not included when gathering the basic configuration data and thus the phone alway started in german language.<br/> <br/>solution: fix in code<br/> <br/>files: phone_config.cpp<br/><br/>products: all phones<br/><br/>risks: None<br/><!---->
Uninitialized buffer content presented as name identification.<!---->
''Status:''
   
checked in to 9.00,8.00,09-80500
= V8 Hotfix 7 (80500.27) =
Changes included in Version 8 hotfix7
[http://mantis.innovaphone.com/view.php?id=56817 Definition]
 
== New Features ==
 
 
      
      
=== Kerberos for PBX-Realms did not work on IP302 and IP305 ===
=== SIP: Distinctive ring tones ===


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Problem: The processing of LDAP search results in the Kerberos realm tree was erroneous and therefore failed on devices of the IP28 platform.<br/><br/>Solution: Fix processing of LDAP search results.<br/><br/>Files affected: kerberos_db.cpp, kerberos_ldap.cpp<br/><br/>Risks: none<!---->
Handling of "Alert-Info: internal".<br/>Triggers special ring tone.<!---->
''Status:''
sip.cpp
      
      
=== Timing problem at the first request to a Kerberos server ===
=== SIP: Send P-Asserted-Identity header in 180/Ringing ===


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Problem: Currently there is a timing problem when a Kerberos server receives its very first ticket request. To answer this (and any following) requests the server needs its own secret key. The current flow is that the server calculates this key, answers the request and then writes the key to the database. Until the key has been written all following requests will be answered with a Kerberos error message. As a consequence the very first try to join the Kerberos realm fails.<br/><br/>Solution: The server has to wait until the key has been written to the database before it answers the first request.<br/><br/>Files affected: kerberos_ldap.h, kerberos_ldap.cpp, kerberos_db.cpp<br/><br/>Risk: small<!---->
Some UAC do not show called party's display name when added to To header by UAS.<br/>We now provide PAI header in provisional responses also containing the called party's display name.<!---->
''Status:''
siptrans.cpp/h<br/>sip.cpp
      
      
=== "vars del" does not delete additional administrator accounts ===
=== Gatway: Call completion interworking on called side did not work ===


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|[http://mantis.innovaphone.com/view.php?id=48400 48400]
|[http://mantis.innovaphone.com/view.php?id=56214 56214]
|}
|}
Problem: The "vars del" command preserves all VARS with the name prefix CMD0. Therefore the additional administrator accounts and other module configuration are not reset. This problem occurs on factory resets and config updates.<br/><br/>Solution: Preserve only "CMD0/AUTH" when executing the "vars del" command.<br/><br/>Files affected: command.cpp<br/><br/>Risks: none<!---->
Call completion on called side did not work<br/><br/>Thanks to Georg Hartwig for giving us his precious support during developent!<!---->
''Status:''
relay.cpp/h<br/>q950.cpp/h<br/>q931.cpp/h<br/>q931_nt.cpp<br/>q931_te.cpp<br/>nt_tbl.tbl<br/>te_tbl.tbl<br/>fty.cpp/h
      
      
=== Setup wizard does not accept new XML formatted license files ===
=== SIP Interworking: CGPN in display name of From URI ===


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|}
|}
Problem: When trying to upload an XML formatted license file while stepping through setup wizard no licenses are accepted. No error indicated neither.<br/><br/>Solution: Delegate upload to new XML style license handler.<br/><br/>Files: setup_lics.xsl<br/><br/>Products affected: All devices.<br/><br/>Risk: No risk.<!---->
SIP Interworking: Get CGPN from display name of From URI<!---->
      
      
=== Trap when configuring empty realm name for Kerberos server ===
=== A DHCP client with  "/keep on" should send DISCOVER requesting the last assigned address after boot (not a REQUEST) ===


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|}
|}
Problem: On the General/Kerberos page the box traps when the user removes the realm name and clicks the Ok button.<br/><br/>Solution: Fix processing of form.<br/><br/>Files: kerberos_db.cpp<br/><br/>Risks: none<!---->
In WLAN networks with more than one DHCP Server REQUESTing the last assigned address after boot needs more time to switch to a new server if the server providing this address has gone. <!---->
      
      
=== SIP: Interoperability to Aastra endpoints ===
=== Configuration Option to keep Routes over a PPP interface always active ===


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|}
|}
Problem: Calls broadcasted by WQ cannot be accepted by Aastra phones. Aastra phones sending 180 Ringing with SDP offer while requesting PRACK with SDP answer. SDP answer cannot provided at this early stage.<br/><br/>Solution: Send dummy answer in PRACK. Re-Negotiation will happen anyway after connect.<br/><br/>Files: sip.cpp/h<br/><br/>Products affected: PBXs serving SIP Phones<br/><br/>Risk: No risk. <!---->
To guarantee that certain connections are only established over a virtual private network, routes over a PPP interface need to be kept active in routing table even while the PPP interface is down. This is done now by checking<br/>   "Configuration/IP/PPP-Config/PPP<n>/Always keep Routes active"<br/>For enabled PPP interfaces which are not up the current routing state (active/skipped) is displayed in addition to the interface state under<br/>   "Configuration/IP/Routing" <br/><br/><!---->
   
== Bug Fixes ==
 
 
      
      
=== IP-DECT GUI MWI numbers ===
=== Gateway: Trap if Name Out or other fields with very long content ===


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|}
|}
problem: The configuration options for MWI numbers are not shown.<br/><br/>solution: Fixed.<br/><br/>files: dectfty.xsl.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. Only GUI change.<!---->
A buffer overrun could happen if very long strings were used as input values<!---->
''Status:''
gk.cpp
      
      
=== IP-DECT SARI variable ===
=== PBX: Unknown filter did not work anymore in version 8 ===


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|}
|}
problem: The SARI variable is updated every configuration change, but is not needed in OEM modules.<br/><br/>solution: Update condition fixed.<br/><br/>files: dectlocalusers.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage.<!---->
The unknown filter could be configured, but was not applied to calls made by endpoints registered as unknown.<!---->
''Status:''
pbx.cpp
      
      
=== SIP: No switch from local ringback tone to inband ringback tone ===
=== Firmware update failure on ip4001 ===


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Problem: No switch from local ringback tone to inband ringback tone, because 183/Progress response was not handled after 180/Ringing.<br/><br/>Solution: Handle 183 after 180.<br/><br/>Files: sipstate.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk.<!---->
On the IP4001 the hwbuild string is computed using the boot flags to see if the box is in production mode. This causes a flash access conflict if the info screen is shown during a flash write ( firmware upload ). <!---->
''Status:''
cpu.cpp cpu.h
      
      
=== SIP: Problems with DTMF when interworking v5 devices to SIP ===
=== Gateway: Overlap Dialing routes did not work as expected ===


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|[http://mantis.innovaphone.com/view.php?id=56006 56006]
|}
|}
Problem: A call initiated by a v5 device that is interworked to SIP may have problems with DTMF.<br/><br/>Solution: Send SDP offer in INVITE as one single media description.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP gateway and PBXs<br/><br/>Risk: No risk.<!---->
- sometimes '#' was added to the outgoing call even if 'Add #' was not configured<br/>- enbloc calls were terminated by a route with '.' as incomplete if not enough digits, even if matching routes followed<!---->
''Status:''
relay.cpp, gk.cpp
      
      
=== One-way-voice after retrieve together with SRTP, H323 and Registration with password ===
=== IP2x IP30x: Missing tones on BRI interface with SIP implementations that send RTP prior to coder negotiation ===


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|}
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problem: One-way-voice could happen after retrieve, if SRTP is used on H.323 call, which uses a refistration with password. This could also happen when switching to a 3-pty conference (retrieve done on one leg). This is caused by an unencrypted SRTP key sent with the media renegotiation.<br/><br/>solution: Do encryption of the SRTP key.<br/><br/>files: h323sig.cpp<br/><br/>products: all<br/><br/>risks: Collateral damage with media negotiation<!---->
This is the problematic scenario:<br/>The IP302 BRI interface is registered on a SIP proxy.<br/>An outgoing call is placed, the SIP proxy sends a STATUS 180 Ringing without SDP information. <br/>The remote side sends RTP data (with inband information) to the IP302.<br/>This switches off the IP302 generated tone, but the remote tone is cannot be used since the SDP is missing in the STATUS 180 message.<br/><br/>Now we ignore RTP with unknown coder for switching off the tone.<!---->
''Status:''
ac_dsp3.cpp
      
      
=== no presence note set if no activity has been selected ===
=== SIP: Switch to fax did not work in some cases ===


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|}
problem: if the activity is empty, but a note given, no note was set for this presence index<br/><br/>solution: use note even if no activity is given<br/> <br/>files: pbx_dtmf.cpp<br/><br/>products: all pbx devices<br/><br/>risks: None <!---->
Sometimes switch to audio occured immediately after switch to t.38<!---->
''Status:''
sip.cpp
      
      
=== handle calls from master/slave user in dtmf object if calling number is found as mobility fork ===
=== Call Completion on Busy to diverted destination failed ===


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problem: if a call comes from the master/slave user, the dtmf object cancels the call, even if there is a user which has the calling number as mobility fork<br/><br/>solution: check mobility users for incoming master/slave calls and if a user is found, use this one<br/> <br/>files: pbx_dtmf.cpp<br/><br/>products: all pbx devices<br/><br/>risks: None<!---->
with the call rejection no informtion about the final destination (leg1 info) was sent, so the call completion was tried with the original called destination.<!---->
''Status:''
pbx.cpp
      
      
=== VM: Trap, Double-Free after sending Email with Body ===
=== PBX: Multiple mobility destinations with delay not handled optimal ===


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problem: VM, trap, double free after sending email with body<br/><br/>solution: don't send such emails or apply fix<br/> <br/>files: smtp_mta.cpp<br/><br/>products: all pbx devices<br/><br/>risks: None <!---->
- if no local phone was registered, all mobility destinations were called right away. Now the destination with the shortes delay is called right away and the others later according difference in delay<br/><br/>- if local phone was busy the mobility destinations was only called after delay. The one with the shortes delay should be called first and then the others.<!---->
''Status:''
pbx_mobility.cpp, pbx_mobility.h, pbx.cpp, pbx_api.h
      
      
=== Display of hardware id missing for UNKNOWN Registrations ===
=== PBX: Groups could not be configured for objects with empty PBX setting ===


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|}
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problem: The Hardware Id is not displayed for Unknown Registrations on the Registrations page and is not copied into the edit page<br/><br/>solution: User Interface fixed<br/><br/>files: pbx_regs.xsl, pbx_edit_object.xsl<br/><br/>products: All with PBX<br/><br/>risks: None<!---->
Empty PBX setting means the object is handled as it has the local PBX set. So the local groups should be selectable<!---->
''Status:''
pbx_admin.cpp, pbx.cpp
      
      
=== When pressing apply on PBX Objects Node editor, the editor changed to a PBX editor ===
=== Always allow local authentication in boot mode ===


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|[http://mantis.innovaphone.com/view.php?id=56396 56396]
|}
|}
problem: After apply the wrong .xsl was used (pbx_edit_loc.xsl instead of pbx_edit_node.xsl)<br/><br/>solution: Use correct .xsl<br/><br/>files: pbx_edit_node.xsl<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
As Kerberos does not work in boot mode, the disable local authentication flag must be ignored there.<!---->
''Status:''
Files: command.cpp
      
      
=== Configuration of _KADMIN_ password ===
=== SIP: Switch to t.38 was answered with audio instead of 488 reject ===


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|}
|}
Problem: On the PBX/Security page the _KADMIN_ password can only be deleted by removing it from the first one of the two input fields. This is inconsistent with the configuration of the other users that can also be deleted by removing the password from both input fields.<br/><br/>Solution: Implement the deleting of the _KADMIN_ password in a consistent way: Removing the password from the first or both fields removes the _KADMIN_ user.<br/><br/>Files: pbx_admin.cpp, pbx_password.xsl<br/><br/>Risks: Small risk of collateral damage<!---->
In case t.38 is not enable, a switch to t.38 was not rejected with 488.<br/>SDP answer with currently active audio coder was send instead.<!---->
      
      
=== SIP: CSeq not correct inside dialog ===
=== PBX: Errors when creating or changing Mobility objects were not displayed ===


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|}
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Problem: Requests within a dialog MUST contain strictly monotonically    increasing and contiguous CSeq sequence numbers (increasing-by-one).<br/><br/>Solution: Keep a private CSeq counter at each call object, subscription object and registration object.<br/><br/>Files: sip.cpp/h siptrans.cpp/h<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk. <!---->
If an error was detected (e.g. duplicate number) saving of the object was prohibited, but no error message as for other objects was displayed<!---->
''Status:''
pbx_edit_mobility.xsl
      
      
=== Trap if dialing wrong number from mobile phone ===
=== PBX-SOAP: Admin function could not be used to configure some new parameters ===


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|}
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problem: A trap happens if a mobile endpoint dials a wrong number and continues to dial when the busy tone is already played.<br/><br/>solution: Better handling of DTMF during playing busy<br/><br/>files: pbx_mobility.cpp<br/><br/>products: all with PBX<br/><br/>risks: Minimal risk of collateral damage<!---->
like phone-config, description, ...<!---->
''Status:''
pbx.cpp, pbx.h
      
      
=== PBX: Trap if filter next config too long ===
=== IP-DECT R-key handling for OEM protocol ===


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|}
|}
problem: If a PBX Filter is configured with a 'next filter' of a length more then 15 characters the PBX traps<br/><br/>solution: Check for length<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: none<!---->
The R-key for an OEM protocol does not work.<!---->
      
      
=== Presence display on fkey 'Partner' stops updating ===
=== Support for packetization up to 80ms ===


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|Id
|[http://mantis.innovaphone.com/view.php?id=48569 48569]
|[http://mantis.innovaphone.com/view.php?id=56566 56566]
|}
|}
Problem: Phone gives up on trying to to establish/re-establich presence subscription.<br/><br/>Solution: Don't stop trying to establish/re-establich presence subscription.<br/><br/>Files: phonesig.cpp<br/><br/>Products affected: Phones<br/><br/>Risk: No risk.<!---->
60ms was the limit before<!---->
''Status:''
h323ch.cpp
      
      
=== If for one user two mobile endpoints are configured, no mobile endpoint could be called ===
=== IP-DECT FTY with TSIP and SIPS ===


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problem: Calls to a user with two mobile endpoints resulted in busy. If call-waiting was turned on, only one mobile endpoint was called.<br/><br/>solution: Algorythm to check if a call already exists for a mobile endpoint fixed<br/><br/>files: pbx_mobility.cpp, pbx_mobility.h, pbx.cpp, pbx.h, pbx_api.h, pbx_xml.cpp, pbx_dtmf.cpp<br/><br/>products: all with PBX<br/><br/>risks: Small risk of collateral damage with mobile endpoints<!---->
The feature codes do not work with TSIP, the local cf does not work with TSIP and SIPS.<!---->
      
      
=== IP 22/24/28/302/305 - provide more dynamic memory by disabling memory guard ===
=== IP-DECT: No Audio was received during call waiting ===


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problem: memory guarding requires 8 byte per malloc'd item. For 50000 items as already seen on active boxes the overhead is 400000 byte. Because of the limited memory it's better to disable guarding.<br/><br/>solution: fix in code<br/> <br/>files: ip24.mak<br/><br/>products: IP 22/24/28/302/305<br/><br/>risks: None<br/><!---->
This was another collateral damage from<br/><br/>fix: #55177: No Media event was generated even everything was normal for unanswered CC exec on IP-DECT<!---->
''Status:''
checked in to 9.00,8.00,09-80500
      
      
=== Call-Completion display at called phone not deleted if calling/called in same but not root node ===
=== Changing the do-not-disturb user setting has no effect if do-not-disturb function key configured and present ===


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problem: The Call-Completion waiting display at called endpoint was not deleted after execution of call completion if calling and called endpoint in same but not root node. The mwiDeactivate used to deleted the display contained the wrong number. In a previous fix the mwiActivate which was also wrong was fixed, so before this fix it worked, because both contained the same wrong number<br/><br/>solution: Fix number in mwiDeactivate<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
problem: Changing the do-not-disturb user setting has no effect if do-not-disturb function key configured and present<br/><br/>solution: fixed in code<br/><br/>files: phone/user/phone_user.cpp<br/><br/>products: all IPxxx telephones<br/><br/>risks: none<br/><br/><!---->
      
      
=== Phone: Handling of failed attended transfer ===
= V8 Hotfix 8 (80500.28) =
Changes included in Version 8 hotfix8
[http://mantis.innovaphone.com/view.php?id=56818 Definition]
 
== New Features ==
 
 
   
=== Gatway: Do not pass through SRTP key if "Enable SRTP" not activated ===


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Problem: An attended transfer may fail. If an error is encountered the call legs may disappear in the background. This happened regularly when a phone tried to transfer calls bound to different SIP registrations or to a SIP and a H323 registration.<br/><br/>Solution: Terminate both calls if attended transfer failed. Don't initiate transfers which which cannot be handled by SIP protocol.<br/><br/>Files: sip.cpp, phonesig.cpp, app_call.cpp<br/><br/>Products affected: Phones<br/><br/>Risk: No risk.<!---->
Pass through SRTP key only if "Enable SRTP" is activated<!---->
''Status:''
channel.h<br/>sip.cpp<br/>gk.cpp<br/>h323ch.cpp
      
      
=== Progress Indicator 'Call not end to end ISDN' sent with each H.323 alert ===
=== PBX: Only 8 IP Filters possible, no indication if maximum reached ===


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problem: The PI 'call not end to end isdn' was sent with each H.323 alert. But this progress code also indicates that in-band info is available, so a local ringback is sometimes turned of because of it.<br/><br/>solution: Do not send 'call is not end to end isdn'<br/><br/>files: relay.cpp<br/><br/>products: all gateway products<br/><br/>risk: Could be that some endpoints require this PI<!---->
Number increased to 32. If 32 Filters are configured no field to enter a new one is displayed<!---->
''Status:''
pbx.cpp, pbx.h, pbx_api.h, pbx_admin.cpp, pbx_global.xsl
      
      
=== feature codes are not always working for mobility, if multiple mobility objects exist ===
=== PBX: Filters to even restrict registration with password ===


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problem: if there are more than one mobility objects, mobility feature codes didn't work for users with the second (etc.) mobility object<br/><br/>solution: recursivly go through mobility objects when trying to find a user with a certain fork number<br/> <br/>files: pbx_dtmf.cpp, pbx.cpp, pbx.h, pbx_api.h<br/><br/>products: all pbx devices<br/><br/>risks: None <!---->
The existing filters only restricted registration to the PBX without password. Now in addition to this registration with password can be restricted as well.<!---->
''Status:''
pbx.cpp, pbx.h, pbx_api.h, pbx_admin.cpp, pbx_global.xsl, pbx_admin_hdr.xml
      
      
=== Gateway Routing Table: Routing a call to a Gatekeeper registration using Name Out did not work anymore ===
=== DTMF facilities: new MWI modes for an OEM protocol ===


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problem: When Name Out in a route was used to address a Gatekeeper registration this Name Out was sent with the call. This prohibited further routing of the call on the receiving side with overlap sending.<br/><br/>solution: Do not send Name Out with call if the name is used to identify Gatekeeper reg.<br/><br/>files: relay.cpp<br/><br/>products: all Gateway products<br/><br/>risks: Small risk that this name needs to be transmited for some special configurations<!---->
New modes for message waiting indication added in the DTMF facility module. There are used for an OEM protocol in OEM IP-DECT devices.<!---->
      
      
=== PBX/Node was added to config object when configuration was changed ===
=== SIP: Allow to receive messages larger than 2560 bytes ===


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|[http://mantis.innovaphone.com/view.php?id=57081 57081]
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problem: When the configuration of a config object was changed (e.g. a Function key added to the phone config), the Node and PBX was set to the local PBX. The config object should never have a Node/PBX. The Node/PBX config could not be removed anymore.<br/><br/>solution: Do not add Node/PBX in this case<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
There was a limitation for incoming SIP messages at 2560 bytes.<!---->
      
      
=== enabling mobility/cw didn't work for 2 stage dialing over mobility object ===
=== IP-DECT: anonymous login; master id checks/traces ===


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problem: if a mobil client tries to call its mobility object first and then dials the corresponding feature code for mobility on/off (and mobility cw), it didn't work<br/><br/>solution: separate method for toggling mobility/cw on calling user object<br/> <br/>files: pbx_dtmf.cpp, pbx.cpp, pbx.h, pbx_api.h<br/><br/>products: all pbx devices<br/><br/>risks: None <!---->
For anonymous handsets login additional master id checks and traces added.<!---->
      
      
=== Modem interop problem ===
=== make function keys on the phone-ui unmodifiable and unviewable ===


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problem: The modem bypass function does not work with some modems.<br/><br/>solution: Option to disable modem bypass added, pcm trace option added, new DSP code<br/><br/>files: ac_dsp3.cpp ac_dsp3.h dsp.xsl trace.xsl ac49?004.h ac49x_drv_*.h<br/><br/>products: ip22 ip24 ip28 ip302 ip305<br/><br/>risk: low risk<br/> <!---->
problem: by setting a function key readonly mask (config change PHONE USER /funclock-ro-mask <mask> or web-ui: Phone->Protect->Function keys not modifiable on the phone-> <mask>), one can now determine a set of function key types which can only be set thru a web-ui and can only be viewed but not modified through phone-ui (see http://wiki.innovaphone.com/index.php?title=Howto:Disable_Function_Key_Modification_On_Phone_UI)<br/><br/>solution: fixed in code<br/><br/>files: phone/user/*<br/><br/>products: all telephones<br/><br/>risks: none<br/><!---->
      
      
=== Missing normalization of received diverting leg info ===
=== SIP: Use registration's Contact-URI as Request-URI on calls to endpoints only ===


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problem: A diverting leg info received on a trunk or gateway object must be normalized to the PBX internal representation (number from root with all prefixes but without escapes) to be displayed correctly when sent to an endpoint. This normalization was missing.<br/><br/>solution: Normalization added.<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Registered gateways get a Request-URI containing the destination number<!---->
      
      
=== PBX: Dialing local objects without PBX config was not possible from different PBX ===
=== Automated Kerberos configuration triggered by a special VAR ===


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|[http://mantis.innovaphone.com/view.php?id=57330 57330]
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problem: If an object is defined without PBX and Node, it is handled on each PBX as if it was configured for the local PBX/Node. Thus it should be possible to dial it on a given PBX by using the node prefix of this PBX. This was not possible if the local flag on the object was set as well.<br/><br/>solution: PBX routing fixed<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h, pbx_dtmf.cpp<br/><br/>products: all with PBX<br/><br/>risks: Collateral damage within routing to 'local' objects<br/><br/><!---->
A box can now be advised to join a Kerberos realm by writing an XML-Command to variable CMD0/KCMD.<!---->
''Status:''
command.h<br/>command.cpp<br/>command.xsl<br/><br/>http://wiki.innovaphone.com/index.php?title=Howto:How_to_configure_Kerberos_using_commands#Automated_Client_Configuration_.28V8_Hotfix8_and_later.29
      
      
=== allow empty search base in directory search object ===
=== IP-DECT: Kerberos configuration options for radio device configuration ===


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|[http://mantis.innovaphone.com/view.php?id=57339 57339]
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problem: ldap server needs an empty search base, but the directory search object doesn't allow an empty one<br/><br/>solution: disable check for empty search base<br/> <br/>files: pbx_dirsearch.cpp<br/><br/>products: all pbx devices<br/><br/>risks: None <!---->
Now it is also possible to configure the Kerberos client if the radio device in discovery mode is configured by the master. The new feature #57330 is used.<!---->
      
      
=== Trap if T.38 timer expiry during closing of T.38 session ===
=== IP-DECT: Messaging options and XML message type support ===


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problem: If a T.38 timer expired right within the fraction of a ms which it takes to close the T.38 session, a trap happened<br/><br/>solution: Stop timer before closing session<br/><br/>files: media.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
New configuration page "DECT - Messaging" for the IP-DECT messaging alert signal options. The enable option replaces the IP Master option "Enable messaging to PBX".<br/>The XML message type is supported now. With XML messages it is possible to change the alert signal message dependent.<br/>The message priority can be considered if enabled: the SIP priority "emergency" changes the alert signal to alarm and the priority "non-urgent" changes it to silence.<!---->
      
      
=== Memory Leak ===
=== Decoding of special XML entities ===


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Problem: Lost memory every time a registered SIP interface is deactivated.<br/><br/>Solution: Free allocated memory for authentication data.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP endpoints<br/><br/>Risk: No risk.<!---->
Implement decoding of the following entities: &amp;lt; &amp;gt; &amp;quot; &amp;apos; &amp;amp;<!---->
''Status:''
files: xml.cpp
      
      
=== PBX: As diverting number the real number was sent even if Send Number configured ===
=== IP-DECT: log messages for MSF calls ===


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problem: The 'Send Number' should be sent as diverting number if configured<br/><br/>solution: Send 'Send Number' as diverting number if configured<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Log messages for MSF calls added.<!---->
      
      
=== Remove memory leak in kerberos client ===
=== IP-DECT: MSF module option disable ===


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Problem: When the decryption of a Kerberos ticket fails a memory leak is left. Also the Kerberos client does not report this to the application.<br/><br/>Solution: Fix protocol implementation and Kerberos client.<br/><br/>Files: kerberos_prot.cpp, kerberos_client.cpp<br/><br/>Risk: none<!---->
With the option /disable it is possible to disable the DECT MSF module.<!---->
      
      
=== PBX: Reject external calls did not work as desired together with call forward ===
=== VM, URL parameter "$_noctl=true" allows to reject control-calls ===


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problem: An endpoint with 'Reject Ext. Calls' set should not be able to forward external calls. An external call forwarded to an endpoint with 'Reject Ext. Calls' set should succeed.<br/><br/>solution: fixed<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: Could be that the old behaviour was desired in some installations<!---->
Control calls may reach a VM object unintentionally. Such calls can now be rejected.<!---->
      
      
=== IP240-1000 - network connectivity lost  after restart of the physical layer controller because of spurious errors ===
=== Gateway: If Moh Mode is configured set 'exclusive coder' checkmark as well on UI ===


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problem: the physical layer controller (PHY) of an IP240-100 is checked for spurious errors any second. If such errors are detected the controller is restarted. After restart data transfer from the phone CPU to the network was blocked. Data reception from network and transfer between LAN and PC port did work.<br/><br/>solution: fix in code<br/> <br/>files: inca_drv.cpp<br/><br/>products: IP 240-1000<br/><br/>risks: None<br/><!---->
The MOH Mode implies that exclusive coders are used<!---->
''Status:''
''Status:''
checked in to 9.00,8.00,09-80500
relay_edit_phys.xsl
      
      
=== Presence subscription to external user failed when forwarded through PBX object "Gateway" ===
=== Phone: Show presence note on 'partner' fkey label ===


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Problem: "Gateway" object did not remove its prefix when forwarding subscription.<br/><br/>Solution: Remove prefix when forwarding presence subscription.<br/><br/>Files: pbx.cpp<br/><br/>Products affected: PBX<br/><br/>Risk: No risk.<!---->
Show presence note (if availbale) on 'partner' fkey label.<br/>If no text note is avalable, activity is shown (as usual).<!---->
      
      
=== Leaks with DECT signalling using TLS ===
=== update service 'provision' option to request earlier and faster polling in provisioning mode ===


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Problem: Sometimes the TLS sockets that are used by DECT signalling are not deleted because of a shutdown event collision in the TLS socket<br/><br/>Solution: Fix TLS shutdown flow<br/><br/>Files: tls.cpp<br/><br/>Risk: Collateral damage with other applications using TLS<!---->
In provisioning mode the update service should start polling the update server as soon as possible and not use the default delay.<br/>This can be configured now by<br/><br/>    config add UP1 /provision <n><br/><br/><n> defines the delay in seconds of the first poll, subsequent polls start after (previous delay * 2) seconds. The maximum delay between polls is 60 seconds.<br/><br/>    config add UP1 /provision 0<br/>or<br/>     config rem UP1 /provision<br/><br/>switches back to the default or the configured polling interval<br/><br/><!---->
      
      
=== RTP is sent to wrong destination ===
== Bug Fixes ==
 
 
   
=== PBX: Checking if a call matches an pending call-completion request was wrong ===


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Problem: A single packet causes the RTP stream to be redirected to a new destination (workaround for NAT). May cause no-media in case of late packet arrival after call transfer.<br/><br/>Solution: Only a continuous packet stream can cause the RTP redirection.<br/><br/>Files: media.cpp/h<br/><br/>Products affected: All devices<br/><br/>Risk: No risk.<!---->
If a call completion is pending and the user calls the destination with the pending CC or the user retries successfully the call independent of the pending CC, we want to avoid to signal this CC. For this we match any calls to pending CCs. Sometimes this resulted in matches even if there was none and pending CCs were cleared which shouldn't<!---->
''Status:''
pbx.cpp
      
      
=== Potential trap when reconfiguring Gateway interfaces ===
=== PBX: Trap if duplicate "Long Name" in Database ===


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problem: If a gateway interface is reconfigured while calls are active a trap could happen later on.<br/><br/>solution: Better handling of reconfiguration<br/><br/>files: signal.cpp, signal.h<br/><br/>products: all<br/><br/>risks: none<!---->
It may happen that on a replicated PBX temporarily multiple objects with the same Long Name (cn) exist. In the case the PBX restarted.<!---->
''Status:''
pbx.cpp
      
      
=== Gateway: Trap if reconfiguring an analog interface registration with Feature Code support ===
=== PBX: CFNR configured at Waiting not executed correctly on transfer to Waiting ===


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problem: If the reegistration configuration of an analog interface (e.g. changing registration name/no) was changed a trap happened when the registration was up again. Happened only with enabled Feature Codes.<br/><br/>solution: Better handling of reconfiguration<br/><br/>files: dtmffty.cpp, gk.cpp, relayfty.cpp, relayfty.h, relayfty_if.h<br/><br/>products: all with analog interfaces<br/><br/>risks: Minimal<!---->
under some circumstances not executed at all and sometime without waiting for No Response Timeout<!---->
''Status:''
pbx.cpp
      
      
=== IP-DECT facility entity memory leak ===
=== Gatway: Suspend/Resume on call completion interworking ===


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problem: Facility entity objects aren't deleted in rare situations.<br/><br/>solution: Facility entity delete function call fixed.<br/><br/>files: dectmaster.cpp, dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
Suspend/Resume signaling on call completion interworking did not interwork<!---->
      
      
=== Trust manufacturer root certificate by default if there is no certificate in flash ===
=== PBX Mobility: Trap if call to mobile phone scheduled for recall is cleared and SOAP monitoring is on ===


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Problem: New devices are equipped with a certificate chain stored in first flash segment. The devices will add the root certitificate to the trust list at first boot. If there is no certificate in flash this will not happen. Hence staging with HTTPS-based update scripts did not work on legacy devices.<br/><br/>Feature: Add a OEM specific manufacturer certificate in the product_info in the firmware. If present, this certificate will be trusted after factory reset, if there is no certificate in flash.<br/><br/>Files: box.cpp, box.h, os.h, ipXXX.cpp, x509.cpp<br/><br/>Risk: Collateral damage on the X509 module and product_info mechanism<!---->
If call is put on hold by the mobile phone and then the mobile phone hangs up, the PBX tries to recall the mobile phone. If the held party hangs up in this situation with SOAP monitoring of the mobile phone active, a trap happens<!---->
''Status:''
pbx_mobility.cpp
      
      
=== Presence note ignored if presence activity has been set ===
=== Trap on call completion with mobility over dtmf object ===


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problem: Presence note ignored if presence activity has been set, the calling party expects to see both the activity and the note<br/><br/>solution: fixed in code<br/><br/>files: phone/app_disp.cpp<br/><br/>products: all telephones<br/><br/>risks: none<br/><br/><br/><!---->
When using call completion with mobility over the dtmf object, the PBX crashed.<br/>Now call completion over mobility is rejected.<br/><!---->
      
      
=== PBX: Device configuration was lost, if PBX object was changed with SOAP Admin function ===
=== Disconnect from DTMF/ICP/Directory search object didn't work with mobility ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=48874 48874]
|[http://mantis.innovaphone.com/view.php?id=56883 56883]
|}
|}
problem: The <device/> tag could not be written with the SOAP Admin function, so this information was lost, when an object was changed or created using this method.<br/><br/>solution: Allow <device/> tag<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
The disconnect from the DTMF, ICP and Directory search objects didn't work with mobility, as it was wrongly called.<!---->
      
      
=== H450 debug info ===
=== PBX Mobility: CLIR did not work correctly ===


{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=48889 48889]
|[http://mantis.innovaphone.com/view.php?id=56899 56899]
|}
|}
problem: Additional debug messages are needed for the h450_entity object.<br/><br/>solution: Debug info added.<br/><br/>files: h450.h, h450.cpp.<br/><br/>products affected: All devices with H323.<br/><br/>risk: No risk of collateral damage. <!---->
A call was sent without number, but it should have been sent with Number Presentation restricted option set.<!---->
''Status:''
ep_lib.cpp
      
      
=== "mod cmd UP0 scfg TFTP://..." does not work ===
=== SIP: Keep ringing calls longer than 3 min ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=48911 48911]
|[http://mantis.innovaphone.com/view.php?id=56901 56901]
|}
|}
problem: saving config to a TFTP server fails because the update module was not triggered to send the next data chunk.<br/> <br/>solution: fix in code<br/> <br/>files: httpclient_i.cpp httpclient_i.h<br/><br/>products: all<br/><br/>risks: None<!---->
An INVITE client transaction was canceled 180 secs after "180 Ringing" have been received.<br/><!---->
''Status:''
checked in to 9.00,8.00,09-80300,09-80500 
      
      
=== prevent to link useless exit code from library ===
=== IP-DECT: Load sharing for trunks (OEM protocol) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=48937 48937]
|[http://mantis.innovaphone.com/view.php?id=56942 56942]
|}
|}
problem: for some some static objects the constructor registers exit handlers calling some library function. this is useless because we never call exit().<br/> <br/>solution: add dummy function to code<br/> <br/>files: box.cpp<br/><br/>products: all<br/><br/>risks: None<br/><!---->
Load sharing for trunks does not work. It is used for an OEM protocol.<!---->
''Status:''
checked in to 9.00,8.00,09-80500
      
      
=== PBX BC conference object TAPI information ===
=== Trap: When handling call completion request from ISDN ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=48940 48940]
|[http://mantis.innovaphone.com/view.php?id=57113 57113]
|}
|}
problem: The broadcast conference object does not generate information for TAPI.<br/><br/>solution: Monitor connector added which generates the TAPI information.<br/><br/>files: pbx_bc_conf.h, pbx_bc_conf.cpp.<br/><br/>products affected: All devices with PBX.<br/><br/>risk: Minimal risk of collateral damage. <!---->
Trap: When handling call completion request from ISDN<!---->
''Status:''
relay.cpp<br/>q931.cpp<br/>pppif.cpp<br/>signal.cpp/h
      
      
=== IP-DECT GUI user search ===
=== Qsig Leg2 Info decoding could fail ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=48946 48946]
|[http://mantis.innovaphone.com/view.php?id=57126 57126]
|}
|}
problem: A wrong URL is generated if a question mark is typed in the user search field.<br/><br/>solution: Search field text encoding fixed.<br/><br/>files: dect_users_left.xsl.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. Only GUI change. <!---->
Qsig Leg2 Info decoding could fail<!---->
      
      
=== "mod cmd UP0 prot TFTP://..." does not work ===
=== Protect TLS socket against collision of SOCKET_RECV and SOCKET_SHUTDOWN ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=48957 48957]
|[http://mantis.innovaphone.com/view.php?id=57130 57130]
|}
|}
problem: firmware upload from a TFTP server fails because of a missing 'complete' indication in last packet.<br/> <br/>solution: fix in code<br/> <br/>files: httpclient_i.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
It was possible that a collision of SOCKET_RECV from the application and SOCKET_SHUTDOWN from the TLS socket occured. This could lead to a trap because the application was already deleted when the SOCKET_RECV_RESULT was sent.<!---->
''Status:''
''Status:''
checked in to 9.00,8.00,09-80300,09-80500
tls.cpp
      
      
= V8 Hotfix  2 (80500.04) =
=== Missing "Recall possible" text in status line ===
Changes included in Version 8 hotfix2
[http://mantis.innovaphone.com/view.php?id=49093 Definition]
 
== New Features ==
 
 
   
=== IP-DECT IP-Master in IP6000 device ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51509 51509]
|[http://mantis.innovaphone.com/view.php?id=57196 57196]
|}
|}
problem: For big DECT systems the IP-DECT IP-Master should be hosted in IP6000.<br/><br/>solution: IP-DECT module with IP-Master added to IP6000 firmware. Usable only with IP-DECT multi-cell license.<br/><br/>files: dectuser.cpp, config.h, ip6000.h, ip6000.cpp, ip6000.mak, left_menu.xml, trace.xsl, new: dect module files without dect radio files, dect_hdr.xml, dect_admin_hdr.xml, dect.xml, dect_admin.xml.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
problem: Missing "Recall possible" text in status line <br/><br/>solution: fixed in call<br/><br/>files: phone/app/app_cc.cpp [box/phone]/forms/[lcd/]forms_gen.cpp<br/><br/>products: all telephones<br/><br/>risks: none<br/><br/> <br/><!---->
      
      
== Bug Fixes ==
=== PBX: Call from mobile endpoint could not be picked up with DTMF group pickup ===
 
 
   
=== RELAY: Remove config parameter "mask" from GUI ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=48127 48127]
|[http://mantis.innovaphone.com/view.php?id=57204 57204]
|}
|}
ENUM/SIP interfaces shall accept all call sources (no filtering).<br/>No-Reg-IFs shall use addr/mask as filter for call sources.<br/>(Remove old mask logic for outgoing calls in gk.cpp)<!---->
pickup was rejeceted<!---->
''Status:''
pbx.cpp
      
      
=== PBX: Reference to Config Template lost, when opening User from Registrations page ===
=== v9 Replication Compliance ===


{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49089 49089]
|[http://mantis.innovaphone.com/view.php?id=57274 57274]
|}
|}
problem: If a user object was opened from the Registrations page a configured config template was not displayed. By pressing Save or Apply the user object was written without the config template<br/><br/>solution: Display config template when opening user object from registrations page<br/><br/>files: pbx_regs.xsl<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Fixes addressing UTF-8 conversions<!---->
      
      
=== SIP: Handling of re-INVITE collision ===
=== SIP: Some interop tweaks did not work ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49135 49135]
|[http://mantis.innovaphone.com/view.php?id=57354 57354]
|}
|}
Problem: After re-INVITE collision at Mitel-PBX, every incoming re-try was rejected with 491 until outgoing re-try was successful.<br/><br/>Solution: Accept incoming re-try while having a postponed re-INVITE client transaction.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk.<!---->
Some module options did not work after reboot:<br/> /no-hr-notify<br/> /prefer-pai<br/><!---->
      
      
=== Need to configure 'Route Root-Node External Calls to' in case of 'License Only' on Slave ===
=== SIP: Fix for video calls through broadcast user ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49137 49137]
|[http://mantis.innovaphone.com/view.php?id=57504 57504]
|}
|}
problem: A Slave or Standby-Slave PBX configured as 'License Only' did not allow to configure a Root Node Extern destination<br/><br/>solution: Allow configureation of Root-Node Extern<br/><br/>files: pbx_general.xsl<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
When initiating a video call towards broadcast user, an offer/offer collision may occur in the PBX.<br/>The PBX must select the video coder (not only audio coder) in this case.<!---->
      
      
=== Media Renegotiation from H.323 Slowstart to H.323 EFC failed accross multiple PBXs ===
=== IP-DECT: Pickup, caller id update ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49147 49147]
|[http://mantis.innovaphone.com/view.php?id=57509 57509]
|}
|}
problem: The message with the new FeatureSet, which indicated a switchover from non-EFC to EFC was not forwarded by the PBX. This happens if a slow-start endpoint located at a slave transfers a call originating from another slave so that both new endpoints are EFC. The master in this case did not forward the switchover FeatureSet.<br/><br/>solution: forward FeatureSet<br/><br/>files: h323ch.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
Fix for the caller id display update after call pickup.<!---->
      
      
=== Gateway: Feature Code Support Configuration fixed ===
=== SIP: Decoding of special Contact-URIs ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49152 49152]
|[http://mantis.innovaphone.com/view.php?id=57523 57523]
|}
|}
problem: Feature Code fieldset was displayed even if not Feature Code support available. Sometimes empty Feture Code Fieldset<br/><br/>solution: Better checking in UI<br/><br/>files: relay_edit_phys.xsl<br/><br/>products: all gateway products<br/><br/>risks: None<!---->
sip:2031;phone-context=cdp.udp@dpp.nortel:5070;maddr=47.166.92.207;transport=udp<br/>The port information was not extracted from phone-context parameter.<br/>Format used by Nortel only.<!---->
      
      
=== Memory Leak ===
=== SIP: SDP attribut annexb=no was missing ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49164 49164]
|[http://mantis.innovaphone.com/view.php?id=57533 57533]
|}
|}
Problem: When closing a SIP interface a small buffer containing the proxy name was not freed.<br/><br/>Solution: Free proxy name buffer.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP endpoints<br/><br/>Risk: No risk.<!---->
If G.729 Annex B was disabled it must be explicitely announced,<br/>because no mentioning annexb is interpreted as annexb=yes.<br/><!---->
      
      
=== Gateway: CGPN Map at route was executed even if the call using this route failed ===
=== Tones: Ringback cadence for Ireland not correct ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49175 49175]
|[http://mantis.innovaphone.com/view.php?id=57545 57545]
|}
|}
problem: A CGPN Map at a route was executed even if the call using this route failed. This was confusing if rerouting was configured in case a destination was not available.<br/><br/>solution: CGPN map not executed if call failed, so rerouting could be done with the same CGPN<br/><br/>files: gk.cpp<br/><br/>products: all gateway products<br/><br/>risks: Could be that there are configs depending on old, wrong behaviour<!---->
Ringing tone - Ireland<br/>Freq: 400+450<br/>Cadence: 0.4 on 0.2 off 0.4 on 2.0 off<br/><br/><!---->
      
      
=== Cleanup gateway interface config ===
=== PBX: Trap when handling presence subscription for VM object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49181 49181]
|[http://mantis.innovaphone.com/view.php?id=57578 57578]
|}
|}
Problem: Config option 'mask' could not be used as filter for incoming calls.<br/><br/>Solution: Accept configured 'mask' and use it as filter (together with 'addr') for incoming calls on interfaces without registration.<br/><br/>Files: gk_if.h gk.h/cpp relay.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: Long forgotton feature "dial the remote ip address" not available anymore.<!---->
Trap when handling presence subscription for VM object<!---->
''Status:''
pbx.cpp
      
      
=== PBX: Sending of multiple group indications after registration did not work ===
=== Allow dtmf features park/unpark for calls from voicemail object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49190 49190]
|[http://mantis.innovaphone.com/view.php?id=57582 57582]
|}
|}
problem: If a phone registers to the PBX, the PBX is sending group indications for all active calls. If more then one call was active not all group indications were sent successful. This also happened with the update of Boolean function keys.<br/><br/>solution: Sending of Group Indications fixed<br/><br/>files: pbx_gi.cpp, pbx_gi.h<br/><br/>products: all with PBX<br/><br/>risks: Risk of collateral damage in the area of Group Indications<!---->
Currently, calls from the voicemail object to the dtmf object were cancelled, as all calls from non user objects have been cancelled.<br/>Now, the features park and unpark are allowed.<!---->
      
      
=== ENUM/SIP interfaces shall accept incoming calls ===
=== SNMP, ifSpeed wrong ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49198 49198]
|[http://mantis.innovaphone.com/view.php?id=57610 57610]
|}
|}
Problem: Gateway interfaces of type ENUM/SIP did not accept incoming calls.<br/><br/>Solution: Make ENUM/SIP interfaces accept incoming calls.<br/><br/>Files: siptrans.cpp<br/><br/>Products affected: Gateway devices<br/><br/>Risk: No risk. <!---->
SNMP, ifSpeed wrong<!---->
      
      
=== PPP IP header compression traps ===
=== IP-DECT: MSF CLMS messages ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49221 49221]
|[http://mantis.innovaphone.com/view.php?id=57612 57612]
|}
|}
problem: PPP IP header compression traps because a word aligned buffer is addressed by a struct ip_hdr pointer and the GCC optimizer replaced a memcpy by inline long register assignments) <br/> <br/>solution: fix in code<br/> <br/>files: iphc.cpp, iphc.h<br/><br/>products: all<br/><br/>risks: None<br/><!---->
Now CLMS messages can be sent with the MSF module.<!---->
''Status:''
checked in to 9.00,8.00,09-80500
      
      
=== Trap when SIP closes unused transport connections ===
=== VM: trailing '#' in CDPN let's diverted call to VM fail ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49257 49257]
|[http://mantis.innovaphone.com/view.php?id=57649 57649]
|}
|}
Problem: Rare trap when SIP closes transport connections that failed to establish.<br/><br/>Solution: Fix cleanup of unused transport connections.<br/><br/>Files: siptrans.cpp<br/><br/>Products affected: SIP devices using SIP/TCP or SIP/TLS (not SIP/UDP)<br/><br/>Risk: No risk.<!---->
VM: trailing '#' in CDPN let's diverted call to VM fail<!---->
      
      
=== NTP Server must respond to SYNC clients even if the device has no correct time from an official server  ===
=== Filter did not work correctly with local objects and overlap sending ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49267 49267]
|[http://mantis.innovaphone.com/view.php?id=57652 57652]
|}
|}
problem: when the NTP server is used to syncronize devices (SYNC client) a correct time is not required but the server must respond. <br/><br/>solution: don't ask for correct time on a client request.<br/><br/>files: ntp.cpp<br/><br/>products: all<br/><br/>risks: None, responses with a time lower Y2K are ignored by the NTP client (but not by the SYNC client)<br/><!---->
For checking the filter in case of overlap sending, the number including the Node prefix was used regardless if the node prefix was dialed or not.<!---->
''Status:''
checked in to 9.00,8.00,09-80500
      
      
=== RAS registration over a PPTP connection fails - association of server-local addr to PPTP interface wrong ===
=== automatic or manual recording cannot be stopped if the recorded call is not the currently active call ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49308 49308]
|[http://mantis.innovaphone.com/view.php?id=57685 57685]
|}
|}
problem: RAS registration via a PPTP interface failed because a wrong rasAddress was returned in GatekeeperConfirm. Instead of the servers defaut IP address the remote endpoint address was associated to an interface without a configured server-local address.<br/><br/>solution: fix in code<br/><br/>files: ipproc.cpp<br/><br/>products: all<br/><br/>riscs: none<br/><!---->
Automatic or manual recording could not be stopped if the recorded call was not the currently active call.<br/>If the Redial-key is used to toggle recording this is intended behaviour because otherwise the Redial-key could not be used to transfer the non-recorded active call.<br/>If a 'Recording' function key is used to toggle recording there is no need for this restriction.<br/><br/>Now a 'Recording' function key stops automatically or manual started recording any case. <!---->
''Status:''
   
checked in to 9.00,8.00,09-80500
= V8 Hotfix 9 (80500.32)  =
Changes included in Version 8 hotfix9
[http://mantis.innovaphone.com/view.php?id=57750 Definition]
 
== New Features ==
 
 
      
      
=== Configurable distinction of internal and external call ringing on analogue port ===
=== SIP: Suppress Annex B of G.729 if "Silence Compression" is not enabled at the interface ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=49344 49344]
|[http://mantis.innovaphone.com/view.php?id=57540 57540]
|}
|}
For swiss users the internal call should be signaled with a ring sequence that is normally used for external calls and vice versa. Swiss seems to make an exception here. In addition it is now possible to configure 'always internal' and 'always external'.<!---->
Suppress Annex B of G.729 if "Silence Compression" is not enabled at the interface<!---->
      
      
=== wrong help url in ICP object ===
=== permit to send log messages, alarms and events via HTTPS with and without checking the server certificate ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49411 49411]
|[http://mantis.innovaphone.com/view.php?id=57785 57785]
|}
|}
problem: wrong help url for ICP object<br/><br/>solution: change url<br/><br/>files: pbx_edit_icp.xsl<br/><br/>products: pbx<br/><br/>riscs: absolutely none<!---->
Both for the log server and for the alarm/event forward server HTTPS can be configured now.<br/>But because distribution of certifcates a may be problematic if there is a big number of clients checking the server certificate can be supressed by<br/><br/>   config add LOG0 /tls-unchecked<br/><!---->
      
      
=== Basic authentication support in HTTP client ===
=== IP-DECT: OEM device GUI ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49450 49450]
|[http://mantis.innovaphone.com/view.php?id=57993 57993]
|}
|}
Problem: Currently the HTTP client does not support basic authentication but basic authentication is needed to access boxes that have Kerberos configured.<br/><br/>Solution: Integrate basic authentication into HTTP client.<br/><br/>Files: httpclient_i.h, httpclient_i.cpp<br/><br/>Risk: small<!---->
Some little changes for a DECT OEM device for the GUI.<!---->
      
      
=== PBX: After CFNR from Waiting with end of first announcement no MOH during call proceeding/alerting ===
=== IP-DECT: TONE interface ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=49458 49458]
|[http://mantis.innovaphone.com/view.php?id=58041 58041]
|}
|}
problem: If a CFNR is executed at the end of the first announcement of a Waiting object (no second announcement), MOH should be played during call proceeding/alerting of the forwarded call. This did not happen, because MOH was turned off by accident with clearing of the announcement call.<br/><br/>solution: Don't turn off MOH<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: litte risk of other media problems within PBX<!---->
The tone inferface is added to the IP1200.<!---->
      
      
=== SIP: 180/Ringing was not re-transmitted ===
=== product_id 153,154 added ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49461 49461]
|[http://mantis.innovaphone.com/view.php?id=58122 58122]
|}
|}
Problem: If 180/Ringing got lost and the caller re-transmitted INVITE, re-transmission of 180/Ringing was missing.<br/><br/>Solution: Re-transmit last sent provisional response.<br/><br/>Files: siptrans.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk.<!---->
these new IDs are needed for IP152 based phone versions <!---->
      
      
=== SIP: Incoming calls not accepted by PBX ===
=== PBX dtmf group feature marks dynamic in groups ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49473 49473]
|[http://mantis.innovaphone.com/view.php?id=58536 58536]
|}
|}
Problem: Incoming SIP calls are rejected with 407, if lookup of active registration fails due to display-name in Contact header of INVITE.<br/><br/>Solution: Skip display-name of Contact header when performing registration lookup for incoming call.<br/><br/>Files: siptrans.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: Low risk of collateral damage.<!---->
As the PBX dtmf group feature shows all dynamic in and out groups, the displayed name of dynamic in groups will be preceeded with '* ' now.<!---->
      
      
=== IP-DECT OEM compatibility with old MWI configuration ===
=== SIP: Mapping of "403 Forbidden" into "Q.931 Requested circuit/channel not available" ===


{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49518 49518]
|[http://mantis.innovaphone.com/view.php?id=58635 58635]
|}
|}
problem: The MWI configuration should be compatible with old configuration.<br/><br/>solution: Configuration added.<br/><br/>files: dectfty.h, dectfty.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
Previously mapped into "Q.931 Call rejected"<br/>Better mapped into "Q.931 Requested circuit/channel not available" in order to trigger re-routing at the Gateway<!---->
      
      
=== Trap when switching off SIP phone ===
=== SIP: Support of P-Called-Party-ID ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=49556 49556]
|[http://mantis.innovaphone.com/view.php?id=58748 58748]
|}
|}
Problem: Switching off causes unregistration. It traps when receiving REGISTER response.<br/><br/>Solution: Wait for response before deleting interface.<br/><br/>Files: sip.cpp/h<br/><br/>Products affected: SIP devices<br/><br/>Risk: Small risk of collateral damage.<!---->
Get CDPN of incoming SIP calls from P-Called-Party-ID if present.<br/><!---->
      
      
=== IP72 DSP acoustic web page not storing changes upon "OK" ===
=== 30s Timeout for dialing too short ===


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problem: IP72 DSP acoustic web page not storing changes upon "OK"<br/><br/>solution: fixed in code<br/><br/>files: box/omap/omap_code.cpp<br/><br/>products: IP72<br/><br/>risks: none<br/><br/><!---->
When putting someone on hold with 'R' there was a timeout of 30s until the consultation call was terminated. This could be too short to find the one to whom to transfer the call.<br/><br/>The protocol timeout in H.323 (TO302) was increased from 30s to 120s<!---->
''Status:''
h323sig.cpp
      
      
=== IP72: WLAN code upgrade ===
=== PBX: Don't apply Send Number to Recording calls ===


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problem: IP72: WLAN code upgrade to latst from Ascom (Meru fixes)<br/><br/>solution:  <br/><br/>files: ./WLAN/Supplicant/obj/libodSupp_O.a ./WLAN/esta_dk/obj/libestadrv.a ./WLAN/esta_dk/obj/firmware.o ./WLAN/esta_dk/inc/wspVer.h ./WLAN/esta_dk/inc/TI_IPC_Api.h ./WLAN/esta_dk/inc/paramOut.h ascom-drivers/WLAN_drv.cpp<br/><br/>products: IP72<br/><br/>risks: none<br/><!---->
For recording it is usually needed to know the real number<!---->
''Status:''
pbx.cpp
      
      
=== IP72 announcement calls should be routed to headset if plugged ===
=== MWI key with configurable DTMF signaling type for message center calls ===


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problem: IP72 announcement calls should be routed to headset if plugged. currently announced calls are always received with handset. <br/><br/>solution: fixed in code, has to be explicitly enabled in phone's web-ui: "Administration/Phone/Preferences/Route Automatically Connected Inbound Calls to Headset (if enabled)"<br/><br/>files: phone/sig/phonesig.* phone/user/phone_pref.xsl<br/><br/>products: all telephones<br/><br/>risks: none<br/><!---->
Some users must force inband DTMF for certain SIP providers but our Voice Mail requires out of band DTMF signaling.<br/>Now the type of DTMF signaling to be used for calls to the message center can be configured at the MWI key. <!---->
      
      
=== DTMF digits missing during DTMF generation  ===
=== phone: disable call intrusion via partner key when recording is active ===


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|[http://mantis.innovaphone.com/view.php?id=65918 65918]
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problem: Tones are not send out after channel init with a undefined coder <br/><br/>solution: fixed in code, ignore DSP status packets for timing calculation, DSP message trace function fixed<br/><br/>files: ac_dsp2.cpp Recordpck.h ac48xhi.c<br/><br/>products: ip6000/800/1200/1201/4001<br/>risks: low risk <!---->
Call intrusion cannot be performed while recording is active:<br/>- recording establishes a 3party conference between local party, remote party and recorder.<br/>- call intrusion establishes a 3party conference between local party and the two remote parties<br/>- recording and call intrusion at the same time would require a 4party conference which cannot be set up because the phone has only 2 DSP coder channels.<br/><br/>Now if any kind of recording is configured call intrusion is neither offered in 'recall' menu nor performed via partner key.<!---->
   
== Bug Fixes ==
 
 
      
      
=== PBX Waiting Queue did not provide diverting party display name ===
=== Disabling local authentication also turned off module authentication ===


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|[http://mantis.innovaphone.com/view.php?id=57863 57863]
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Problem: PBX Waiting Queue did not provide diverting party display name when forwarding/distributing calls.<br/><br/>Solution: Provide diverting party name.<br/><br/>Files: pbx_wait.cpp/h<br/><br/>Products affected: PBX<br/><br/>Risk: No risk.<!---->
When Kerberos was configured on a box and the local admin accounts were disabled, logging and PBX administration using PBX users did not work anymore.<!---->
''Status:''
files: command.cpp
      
      
=== headset mode must be kept when a knocking call is accepted via operator while a disconnected call is pending ===
=== SIP: Transfer handling at Gateway may cause on-way-audio ===


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|[http://mantis.innovaphone.com/view.php?id=57906 57906]
|}
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problem: a knocking call accepted via operator was routed to speaker instead to headset when the phone was in headset mode playing the busy tone for a call which was disconnected from remote.<br/><br/>solution: fix in code<br/><br/>files: app_ctl.cpp<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
I some scenarios where REFER is handled at the Gateway to transfer a local media call leg (e.g. ISDN) to any other call leg.<!---->
''Status:''
checked in to 9.00,8.00,09-80500 
      
      
=== IP800 conference ===
=== IP-DECT: no digits en-bloc timeout ===


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|[http://mantis.innovaphone.com/view.php?id=49800 49800]
|[http://mantis.innovaphone.com/view.php?id=57925 57925]
|}
|}
problem: Conference hardware initialization for channel ten does not work.<br/><br/>solution: Delay within initialization sequence inserted.<br/><br/>files: ipac_drv.cpp.<br/><br/>products affected: Devices with IPAC chip.<br/><br/>risk: No risk of collateral damage.<!---->
The timeout of the en-bloc timer is changed for the case that no digits are dialed. This fixes the Aastra PBX block bug.<!---->
      
      
=== H323 channel null pointer trap ===
=== Resuming TLS sessions did not work correctly ===


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problem: Trap caused by null pointer access.<br/><br/>solution: Null pointer check added.<br/><br/>files: h323_ch.cpp.<br/><br/>products affected: All devices with H323 protocol.<br/><br/>risk: No risk of collateral damage. <!---->
The server now ensures that session IDs are unique by adding a timestamp and a serial number. This increases the size of session IDs from 16 bytes to 24 bytes.<br/><br/>Also IP addresses were not handled correctly by the session cache.<!---->
''Status:''
tls.cpp
      
      
=== IP-DECT OEM system name update ===
=== QSIG Call Complettion to MD110 failed ===


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|[http://mantis.innovaphone.com/view.php?id=58372 58372]
|}
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problem: The OEM DECT needs update of the system name.<br/><br/>solution: Update added.<br/><br/>files: dectusers.cpp, dectlocalusers.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
QSIG Call Complettion to MD110 failed<!---->
      
      
=== IP-DECT unattended call transfer ===
=== phone directory collating sort  order unexpected ===


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|[http://mantis.innovaphone.com/view.php?id=58386 58386]
|}
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problem: It should not be possible to enter the unattended call transfer mode if the second call is in ring-back state.<br/><br/>solution: Condition added.<br/><br/>files: dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
The ordinal of the space character was higher than that of any alphameric character, thus for example "Smith Eric" was displayed behind "Smithson Eric".<br/>The ordinal of the space character is now 0. <!---->
      
      
=== SIP: INVITE after REFER for blind transfer missed Referred-By header ===
=== SIP: Don't send empty P-Asserted-Identity in provisional response ===


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|[http://mantis.innovaphone.com/view.php?id=58493 58493]
|}
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Problem: After receiving REFER for blind transfer a new INVITE is sent without Referred-By header.<br/><br/>Solution: Save Referred-By header of received REFER on existing call and send it in INVITE for new call.<br/><br/>Files: sip.cpp/h siptrans.cpp/h<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk.<!---->
SIP/2.0 183 Session Progress<br/>Via: SIP/2.0/TCP 10.64.32.2:14937;branch=z9hG4bK6728a259<br/>From: ""<sip:850@10exchange.wschneider.com;user=phone>;epid=123A3A4D16;tag=c755636afc<br/>To: <sip:00763773033@10.64.64.1;user=phone>;tag=3908677425<br/>Call-ID: d0248a8c-a324-454b-807a-923c30c1e24b<br/>CSeq: 34 INVITE<br/>Contact: <sip:00763773033@10.64.64.1:5060;user=phone;transport=TCP><br/>Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH<br/>Content-Length: 230<br/>Content-Type: application/sdp<br/>Server: (innovaphone IP800/8.00 dvl [tac-1.11108:/8050028/400])<br/>Supported: replaces,privacy,answermode,from-change,100rel,timer,histinfo<br/>P-Asserted-Identity: <br/>P-Sig-Options: Sending-Complete<br/><!---->
      
      
=== IP302/IP305: PCM connected channels disconnect other channels media ===
=== Invalid duplicate DTMF object caused the PBX to trap ===


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problem: disconnect is sent to wrong channel<br/><br/>solution: fixed in code<br/><br/>files: ac_dsp3.cpp <br/><br/>products: ip302 ip305 <br/><br/>risks: low risk <!---->
A false config with an invalid DTMF object (name like DTMF#pickup_group) caused the PBX to trap.<br/>Such an object will be ignored now.<!---->
      
      
=== IP2x IP30x: unreliable V.34 modem  ===
=== Pickup function key display discards leading letter on transferred call ===


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|[http://mantis.innovaphone.com/view.php?id=58520 58520]
|}
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problem: echo canceller needs to be off, DSP jitter buffer must be static, output volume must be reduced<br/><br/>solution: fixed in code. Use "disable echo canceller flag" to enable this features. Use http://addr/AC-DSP0/mod_cmd.xml?cmd=form&amp;xsl=dsp.xsl to tune the volume and disable modem-bypass.<br/><br/><br/>files: ac_dsp3.cpp ac_dsp.h <br/><br/>products: ip2x ip30x<br/><br/>risks: low risk <!---->
problem: Pickup function key display discards leading letter on transferred call, so the first letter or number of the calling party is always missing <br/><br/>solution: fixed in code<br/><br/>files: phone/app_disp.cpp<br/><br/>products: all telephones<br/><br/>risks: none<br/><br/><br/><!---->
      
      
=== IP800: V8 Firmware upload not possible after V7 licenseses are returned to myinnovaphone  ===
=== trap on late CHANNEL_INIT to relay_media_relay::serial_event() ===


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problem:  Missing label to identify new license scheme with certificates.<br/><br/>solution: fixed in makefile<br/><br/>files: ip800.mak<br/><br/>products: ip800<br/><br/>risks: low risk <!---->
A null pointer was referenced when a CHANNEL_INIT was passed to an object in closing state  <br/><br/><!---->
      
      
=== H.323, PROGRESS with cause treated as DISC causes problems ===
=== AD-replicator: xml-show-namingcontexts leaks memory ===


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problem: In H.323 no DISC message is defined. Because of that a PROGRESS message with Cause code was treated as a DISC message. This behaviour causes problems, because there is H.323 equipment sending PROGRESS with Cause even if no DISC is intended<br/><br/>solution: No special handling of PROGRESS with cause anymore<br/><br/>files: h323_tbl.h, h323sig.cpp<br/><br/>products: all<br/><br/>risks: old behaviour could be expected by other equipment<!---->
a memory leak occurred every time when clicked on Configuration/LDAP/Replicator(AD)/DN/"Show Options"<br/><!---->
      
      
=== Enblock flag not evaluated on Routes to MAP ===
=== Do not disconnect calls to directory search object from master/slave user ===


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problem: The enblock flag on routes to MAP could be set, but it did not do anything<br/><br/>solution: Evaluate enblock flag on routes to MAP<br/><br/>files: gk.cpp<br/><br/>products: All gateway products<br/><br/>risks: None, no change if enblock flag not set<!---->
Calls from a master/slave user where disconnected by the directory search object.<br/>These calls are allowed now.<!---->
      
      
=== Phone: Local coder config was not used on outgoing phone calls ===
=== Phone: Light up partner fkey even on active state ===


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Problem: Local coder config was not applied to outgoing phone calls, but is required when it comes to media re-negotiation.<br/><br/>Solution: Give local coder config to all kind of calls.<br/><br/>Files: phonesig.cpp/h<br/><br/>Products affected: All phones.<br/><br/>Risk: No risk.<!---->
While the phone itself is in active state (non-idle) a partner fkey lamp did not light up when partner's presence indicate 'on-the-phone' activity.<br/>Only in idle state the lamp indicated that partner is 'on-the-phone'.<!---->
      
      
=== One-way-voice after unpark/pickup together with SRTP, H323 and Registration with password ===
=== SIP: Dialog-Info did not show "confirmed" state ===


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problem: Within media renegotiation after unpark/pickup a wrong SRTP key was sent. This resulted in one-way media.<br/><br/>solution: Transmit correct SRTP key<br/><br/>files: h323sig.cpp<br/><br/>products: all<br/><br/>risk: Other media problems<!---->
"proceeding" was indicated instead.<br/>Caused Problems on snom phones.<!---->
      
      
=== PBX: No default device definition was added to new object ===
=== Soap::UserPickup() sometimes didn't work ===


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|[http://mantis.innovaphone.com/view.php?id=58665 58665]
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problem: If a new object was added to the PBX, with a Name, but without device hw-id, no default device definition was created containing name has hw-id. After an unknown enpoint was assigned to this user by dialing the number of the user a registration with name was not possible anymore.<br/><br/>solution: Create default device definition<br/><br/>files: pbx_admin.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Soap::UserPickup() sometimes didn't work<!---->
      
      
=== presence function key usage on phone traps with non-presence-available pbx ===
=== Call Intrusion across PBXs did not work (intrude call at slave from master) ===


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problem: presence function key usage traps with non-presence-available pbx<br/><br/>solution: fixed in code (check)<br/><br/>files: phone/app/app_disp.cpp<br/><br/>products: all telephones<br/><br/>risks: none<br/><!---->
There was a fix already for this, but this covered only intrude at master from slave.<!---->
''Status:''
pbx.cpp<br/>pbx.h
      
      
=== Media Negotiation between SIP and H.323 failed if Offer from both sides available ===
=== Gateway Routes with CDPN map to number containing '#' did not work ===


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problem: If a H.323 and a SIP call leg were to be connected and a media offer was available on both legs, nothing happend. The new offer should have been sent on the H.323 leg. This situation could happen in special cases with transfer and reverse media.<br/><br/>solution: Send offer on H.323 in this case<br/><br/>files: h323ch.cpp<br/><br/>products: all<br/><br/>risk: Small riks of collateral damage<!---->
The number starting with the '#' was omitted.<br/><br/>Collateral damage of fix: #56006: Gateway: Overlap Dialing routes did not work as expected<!---->
''Status:''
gk.cpp
      
      
=== PBX device definition with empty hw-id was generated for a user without name ===
=== PBX Trunk Object: Incomplete destination did not work for incoming incomplete enblock calls ===


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problem: For objects without device configurations, a default device is generated with the hw-id being the same as the Name of the object. This is for v7 compatibility. This was done even if there was no Name. But it was done only for a single object, because after that duplicate hw-id was detected. This caused registration with number being possible on this object even without device configuration.<br/><br/>solution: Check for empty name<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
collateral damage of fix: #54357: PBX Node 'incomplete Number' destination did not work for block dial calls<!---->
''Status:''
pbx.cpp
      
      
=== SIP: Incoming calls with anonymous From-URI were not tagged as CLIR ===
=== DRAM /Firmware upload stops sometimes ===


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Problem: Incoming SIP calls with anonymous From-URI were not tagged as CLIR.<br/><br/>Solution: Honour "anonymous" in From-URI and set Presentation Restricted flag in CGPN.<br/> <br/>Files: sip.cpp<br/><br/>Products affected: All SIP devices<br/><br/>Risk: No risk.<!---->
Depending on the timing the upload hangs.<br/>Seen with the innovaphone test program and minifirmware<!---->
''Status:''
servlet_post_file.cpp
      
      
=== IP-DECT old anonymous PPs ===
=== Gateway: Trap on early RELEASE from calling side ===


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problem: The old anonymous PPs saved in the system object in firmware version 6 should not longer be used.<br/><br/>solution: Anonymous PPs in the system object are automatically deleted.<br/><br/>files: dectusers.h, dectusers.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
If the caller stops calling at an early stage, a trap may occur:<br/><br/>0:0806:591:0 - LOG CALL 15 Alloc<br/>0:0806:591:3 - LOG CALL 15 A:Call    ->                        / PRI2::->*::<br/>0:0806:597:0 - LOG CALL 15 B:Call    100->226                  / PRI2:5336100:->RP2:226:<br/>0:0806:701:3 - LOG CALL 15 A:Rel    100->226                  / PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry<br/>0:0806:712:3 - LOG CALL 15 Media    100->226                  G711A,20(0,0,0)/G711A,20(0,0,0) PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry<br/>0:0806:713:7 - LOG CALL 15 B:Alert  100->226                  G711A,20(0,0,0)/G711A,20(0,0,0) PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry<br/>0:0806:714:0 - TRAP: 0x10<br/><!---->
      
      
=== PBX: 'Route Internal Calls to' only works for internal destinations being users or slaves ===
=== PBX: Name Identification was not forwarded with forked call ===


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problem: It was explicitly implemented that 'Route Internal Calls to' was only executed for Users or Slaves. This was does not seem to be a usefull restriction.<br/><br/>solution: Restriction removed<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: Could be that this restrictions turns out to be usefull<!---->
With call forking the original calling name id was not forwarded<!---->
''Status:''
pbx.cpp
      
      
=== GUI: Registration indicator not aligned ===
=== PBX: Trap if 'Escape dialtone from' is configured to a non-existent object ===


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Problem: The registration indicator (arrow) was not aligned on Gateway/GK page.<br/><br/>Solution: Make it aligned.<br/> <br/>Files: relay_ifs.xsl<br/><br/>Products affected: All gateways<br/><br/>Risk: No risk. <!---->
Check implemented to use internal TONE interface in this case<!---->
''Status:''
pbx.cpp
      
      
=== PBX: On CFB configured at Slave PBX executed on max_calls, additional digits were added to called number ===
=== SIP: re-INVITE without SDP offer was rejected with 504 Server Timeout in 'held' state ===


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|}
|}
problem: If a CFB on a Slave PBX was executed because max-calls, the original dialed digits should be added to the diverted to number. If the original dialed number did not exactly match a user in the slave, but additional digits were dialed, these digits were added twice.<br/><br/>solution: Add digits once only<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
re-INVITE without SDP offer was rejected with 504 Server Timeout if received on an inactive session.<br/><!---->
      
      
=== PBX: Presence subscription was rejected by object type 'Executive' ===
=== SIP: Handling of reject of re-INVITE without SDP offer was incomplete ===


{|
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|[http://mantis.innovaphone.com/view.php?id=50091 50091]
|[http://mantis.innovaphone.com/view.php?id=58824 58824]
|}
|}
Problem: Watching presence of an 'Executive' user was not possible. Subscription was rejected.<br/><br/>Solution: Accept presence subscription at 'Executive' user .<br/> <br/>Files: pbx.cpp<br/><br/>Products affected: All PBX devices<br/><br/>Risk: No risk.<!---->
Handling of reject of re-INVITE without SDP offer was incomplete.<br/>Need to generate dummy offer for app.<!---->
      
      
=== Gateway sends calls to wrong registered SIP endpoint ===
=== send PROGRESS after CALL-PROC to stop 10s T310 ===


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|[http://mantis.innovaphone.com/view.php?id=50102 50102]
|[http://mantis.innovaphone.com/view.php?id=58839 58839]
|}
|}
Problem: If the addressed endpoint is currently not registered at the registrar interface at the gateway, calls are delivered to another registered SIP endpoint.<br/><br/>Solution: Reject calls if addressed endpoint is not registered.<br/> <br/>Files: siptrans.cpp<br/><br/>Products affected: All gateways<br/><br/>Risk: No risk.<!---->
sometimes too short to forward a call<!---->
      
      
=== Trap if doing Pickup from analog interface with Feature Code ===
=== IP-DECT: potential trap ===


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|[http://mantis.innovaphone.com/view.php?id=50107 50107]
|[http://mantis.innovaphone.com/view.php?id=58920 58920]
|}
|}
problem: If a Pickup was performed from an anlog interface using Feature Codes, the gateway restarted. This was caused by an invalid cast.<br/><br/>solution: Cast fixed<br/><br/>files: relayfty.cpp, relay_api.h, relay.cpp<br/><br/>products: all gateway products with analog interfaces<br/><br/>risks: None<!---->
Potential trap in DECT devices fixed.<br/>Trap identification:<br/>XCPT: no 2 (TLB load)  pc 94273278  ra 94273254  va 0000000c<br/><!---->
      
      
=== Country settings in 'TELx/Physical' cannot select lines containing '+' character ===
=== Gateway: A call counter with name containing blank or other special character created problems ===


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|[http://mantis.innovaphone.com/view.php?id=58944 58944]
|}
|}
problem: lines with'+'character cannot be selcted.<br/><br/>solution: indroduce cmd-line parameters without '+' for si32xx_drv.cpp<br/><br/>files: si3210_drv.cpp, si3241_drv.cpp<br/><br/>products: ip22, ip24, ip28, ip302<!---->
It could be configured, but if another map was added to the same route the config was corrupted<!---->
''Status:''
gk.cpp
      
      
=== AD Replicator, Searches to Global Catalog Server weren't possible ===
=== Trap on CF remove while files are deleted ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=50129 50129]
|[http://mantis.innovaphone.com/view.php?id=58984 58984]
|}
|}
problem: Searches to Global Catalog Server weren't possible<br/><br/>solution: Fix configuration for LDAP port<br/><br/>files: ldaprep.cpp<br/><br/>products: all PBX-,Dect products<br/><br/>risks: None <!---->
When files are deleted from the CF card and the card is removed or has an error, the box could trap.<!---->
      
      
=== Control calls without facility elements were forwarded on ISDN ===
=== Potential trap if routes with DTMF output combined with pause chars (',') are used for calls without channel or out-of-channels ===


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|[http://mantis.innovaphone.com/view.php?id=50199 50199]
|[http://mantis.innovaphone.com/view.php?id=59012 59012]
|}
|}
Problem: Control calls (calls without media channel) without facility elements were forwarded on ISDN. Seems to causes trouble on some ISDN switches<br/><br/>Solution: Reject control calls without facility elements with "Invalid information element contents".<br/> <br/>Files: relay.cpp<br/><br/>Products affected: All gateways<br/><br/>Risk: No risk. <!---->
In this situation pause digits are passed to a channel, which does not exits. This causes the trap.<br/>Could also be dialed pause characters on a call-independent signaling.<!---->
''Status:''
relay.cpp
   
= V8 Hotfix10 (80500.33) =
Changes included in Version 8 hotfix10
[http://mantis.innovaphone.com/view.php?id=59505 Definition]
 
== New Features ==
 
 
      
      
=== fat32 check disc trap ===
=== Call Forwarding Function Key with "Apply 'Always' Setting Only" checkmark  (CFU  Only) ===


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|[http://mantis.innovaphone.com/view.php?id=50204 50204]
|[http://mantis.innovaphone.com/view.php?id=59077 59077]
|}
|}
Problem: After a firmware trap with a cf card, the afterwards check disk can cause a trap loop, if directory entries are corrupt because of the first trap.<br/><br/>Solution: Increment counter which caused the trap loop.<br/> <br/>Files: fat32.cpp<br/><br/>Products affected: All gateways with CF slot<br/><br/>Risk: minor risk<!---->
If "Apply 'Always' Setting Only" is checked the Function key toggles onls over the 'Always' (i.e. CFU) entries and keeps other existing diversions untouched.<br/>Thus CFB or CFNR diversions set at the phone or at the PBX are not changed when toggling this key.<!---->
      
      
=== Phone: Changes to option 'Proposed Registration Interval' were applied after reboot only ===
=== SIP: Registration lookup by attribute 'username' of Authorization header ===


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|[http://mantis.innovaphone.com/view.php?id=59078 59078]
|}
|}
Problem: Changes to option 'Proposed Registration Interval' had no effect until reboot. Demand for reboot was not indicated.<br/><br/>Solution: Apply configured registration interval at runtime. No reboot required.<br/> <br/>Files: phonesig.cpp<br/><br/>Products affected: All SIP phones<br/><br/>Risk: No risk.<!---->
Registration lookup by attribute 'username' of Authorization header (not only on anonymized calls)<!---->
      
      
=== Mobility: Reject of call to mobile endpoint did not work ===
=== x509: Support for DNS names in SubjectAltName extension of certificates ===


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|[http://mantis.innovaphone.com/view.php?id=50239 50239]
|[http://mantis.innovaphone.com/view.php?id=59171 59171]
|}
|}
problem: If a call on a mobile endpoint was rejected, on the calling side there was still ringback. Also a CFB was not executed in this case.<br/><br/>solution: Reject on mobile phone fixed<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h, pbx_mobility.cpp, pbx_mobility.h<br/><br/>products: all with PBX<br/><br/>risks: Risk of collateral damage with Mobility<!---->
Create self-signed certificates and certificate requests that contain a DNS name in the SubjectAltName extension. Display the DNS name in the certificate details.<!---->
''Status:''
Files: x509.cpp, x509.h, x509asn1.h, request.xsl, certificate_create.xsl, certificate.xsl, oids_asn1.h
      
      
=== Mobility: Call to obeject within other PBX not in root node failed ===
=== SIP: Support for another Contact-URI parameter in REGISTER ===


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|[http://mantis.innovaphone.com/view.php?id=50241 50241]
|[http://mantis.innovaphone.com/view.php?id=59174 59174]
|}
|}
problem: Routing of calls from mobile endpoint, did not work with nodes on other PBXs<br/><br/>solution: Routing fixed<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h, pbx_mobility.cpp, pbx_mobility.h<br/><br/>products: all with PBX<br/><br/>risks: Risk of collateral damage with Mobility<!---->
+u.sip!model.ccm.cisco.com<!---->
      
      
=== Mobility: Call forwarding on no response did not work for mobile endpoints if only mobile endpoint ===
=== SIP: Interop feature "X-cisco-srtp-fallback" ===


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|[http://mantis.innovaphone.com/view.php?id=59198 59198]
|}
|}
problem: A call forward on no response, either as CFNR or as no response destination at a trunk failed if only a mobile endpoint was present for a given object.<br/><br/>solution: Forwarding fixed<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h, pbx_mobility.cpp, pbx_mobility.h<br/><br/>products: all with PBX<br/><br/>risks: Risk of collateral damage with Mobility<!---->
Required for SRTP sessions<!---->
      
      
=== Trap if 'Escape Dialtone from' configured not being a User Object ===
=== H.323-Q.931-Interworking - display text provided in the Display Information Element of an ISDN Information Message on phone ===


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|[http://mantis.innovaphone.com/view.php?id=59506 59506]
|}
|}
problem: If a 'Escape Dialtone from' destination was configured, which was not a User object (e.g. a Gwateway) a trap happend when a escape dialtone was to be played.<br/><br/>solution: NULL pointer access fixed<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h<br/><br/>products: all with PBX<br/><br/>risks: Minimal<!---->
The text provided in the Display Information Element of an ISDN Information Message was silently discarded. Now it is displayed in the phone status line. <br/><!---->
      
      
=== Wrong presence status in PBX admin dialog ===
=== SIP: Interop feature "X-cisco-sis-3.0.0" ===


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|[http://mantis.innovaphone.com/view.php?id=59533 59533]
|}
|}
Problem: Presence status 'open' is displayed when no presence status is available.<br/><br/>Solution: Fix presence dialog.<br/> <br/>Files: pbx_edit_presence.xsl<br/><br/>Products affected: All PBXs<br/><br/>Risk: No risk.<!---->
Required for SRTP sessions<!---->
      
      
=== Wrong calling party info on CTI initiated calls from a phone to a Trunk Object with 'Set Calling=Diverting No' checked ===
=== Debug: Support to identify bad objects ===


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|[http://mantis.innovaphone.com/view.php?id=59714 59714]
|}
|}
problem: When a CTI application (TAPI or other SOAP based application) initiates a call from a phone to a Trunk Object with 'Set Calling=Diverting No' checked the the called party receives a wrong calling party info.<br/><br/>solution: when a CT-INITIATE is received on a RC-CONNECT call the cdpn in the CT-SETUP facility sent with the newly created outbound call is set to the phones own number (i.e. identical to the SETUP cgpn). <br/><br/>files: phonesig.cpp<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
Only mem-clients are allowed be deleted dynamically.<!---->
''Status:''
   
checked in to 8.00,9.00,09-80500
== Bug Fixes ==
 
 
      
      
=== In some countries the ring tone timing patterns for internal/external calls need to be swapped to meet country defaults ===
=== H.323 Remote address was not checked for calls coming in on special trunks with non-standard ports ===


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|[http://mantis.innovaphone.com/view.php?id=58958 58958]
|}
|}
problem: the builtin ring tone timing patterns for internal/external calls which are applied to the builtin ring melodies don't meet the country specific preferences for example in switzerland. swapping the patterns may help.<br/><br/>solution: "config add RING /swap-i-x" to swap patterns<br/><br/>files: ring.cpp, phone_pref.xsl<br/><br/>products: all phones<br/><br/>riscs: none<!---->
This is no problem which affects innovaphone standard products. It is only for H.323 trunks configured with fixed remote and local address and port.<!---->
''Status:''
''Status:''
checked in to 8.00,9.00,09-80500
h323sig.cpp
      
      
=== Phone: Stop trying to subscribe for own presence ===
=== Interworked Control-Calls without Facilities Shall Stop in Relay ===


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|[http://mantis.innovaphone.com/view.php?id=50346 50346]
|[http://mantis.innovaphone.com/view.php?id=59009 59009]
|}
|}
Problem: On PBX's not supporting presence subscription (v7 or earlier) the phone endlessly tries to subscribe for own presence.<br/><br/>Solution: Stop trying to subscribe for own presence.<br/> <br/>Files: phonesig.cpp<br/><br/>Products affected: All SIP phones<br/><br/>Risk: No risk.<!---->
Interworked Control-Calls without Facilities Shall Stop in Relay<!---->
      
      
=== Mobility: Send presence info of called user with ALERT at call to mobile endpoint ===
=== PBX Exec Object: A number Map object to be used to call exec directly ===


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|[http://mantis.innovaphone.com/view.php?id=59066 59066]
|}
|}
problem: When a mobile endpoint was called, the presence info of the mobile endpoint (typically there is no presence info available) was send to caller instead of the presence info of the related local user object.<br/><br/>solution: Send presence info of local user<br/><br/>files: pbx_mobility.cpp, pbx_api.h<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
A number map can be put in exec secretary or direct call groups to call the exec thru this Map Object directly. This did not work for calls from IP Phones, which sent a source name with the call.<!---->
''Status:''
pbx_exec.cpp
      
      
=== Phone: Fkey "Partner" should light up LED when partner's presence activity is "on-the-phone" ===
=== PBX: Trap if using SOAP Version Method if PBX not started ===


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|[http://mantis.innovaphone.com/view.php?id=59071 59071]
|}
|}
Problem: Fkey "Partner" does not light up LED when partner's presence activity is "on-the-phone".<br/><br/>Solution: Light up LED on partner key when partner's presence activity is "on-the-phone".<br/> <br/>Files: app_disp.cpp<br/><br/>Products affected: Phones with partner keys only<br/><br/>Risk: No risk. <!---->
null pointer access happens in this case<!---->
''Status:''
pbx_xml.cpp
      
      
=== SIP: Signaling not sent to non-standard port ===
=== DHCP client: "Wait for selected Server" timeout was not applied after a DHCP restart ===


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|[http://mantis.innovaphone.com/view.php?id=50394 50394]
|[http://mantis.innovaphone.com/view.php?id=59076 59076]
|}
|}
Problem: Even if non-standard remote port is configured, signaling is sent to 5060.<br/><br/>Solution: Apply configured remote port.<br/><br/>Files: sip.cpp<br/><br/>Products affected: SIP devices<br/><br/>Risk: No risk.<!---->
When the DHCP client receives a DHCP restart request a timer is setup to trigger the restart. The failure happens when an offer arrives before this timer fires.<!---->
      
      
=== PBX object device config lost, if invalid info added somewhere else (e.g. duplicate number) ===
=== SIP: Media negotiation problem when processing INVITE without SDP ===


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|[http://mantis.innovaphone.com/view.php?id=50415 50415]
|[http://mantis.innovaphone.com/view.php?id=59082 59082]
|}
|}
problem: If a PBX object editor is opened and invalid information is added, then after Apply or OK the error message is displayed and the devices list was empty. After correcting the error and Apply or OK again the object is saved without the device list.<br/><br/>solution: Fill in device list on error as well<br/><br/>files: pbx_admin.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Media negotiation problem when processing INVITE without SDP<!---->
      
      
=== SIP: Problems parsing exotic SIP URIs ===
=== H.323: Don't send a call-independent-signaling call without facilities ===


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|[http://mantis.innovaphone.com/view.php?id=50421 50421]
|[http://mantis.innovaphone.com/view.php?id=59088 59088]
|}
|}
Problem: Failed to decode destination port from a redirect URI like this: <sip:2204;phone-context=cdp.udp@livio.nl:16618;maddr=10.2.10.3;transport=udp;x-nt-redirect=redirect-server><br/><br/>Solution: Fix URI parsing.<br/><br/>Files: sipmsg.cpp<br/><br/>Products affected: All SIP devices<br/><br/>Risk: No risk<!---->
This could happen if a QSIG call was interworked, with facilities we do not support<!---->
''Status:''
h323_tbl.tbl<br/>h323sig.cpp<br/>h323sig.h<br/>phonesig.cpp
      
      
=== Gateway: Cannot use SIP interfaces without having "Media-Relay" and "Exclusive Coder" enabled ===
=== send PROGRESS after CALL-PROC to stop 10s T310 - in ISDN Stack not PBX ===


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|[http://mantis.innovaphone.com/view.php?id=59195 59195]
|}
|}
Problem: Cannot use SIP interfaces without having "Media-Relay" and "Exclusive Coder" enabled. Installations with many SIP interfaces and heavy load will suffer from RTP traffic.<br/><br/>Solution: Do not enforce "Media-Relay" and "Exclusive Coder" in SIP interfaces.<br/><br/>Files: relay_edit_sip.xsl<br/><br/>Products affected: Gateways with SIP interface<br/><br/>Risk: No risk.<!---->
sending PROGRESS in the PBX could have some unwanted side effects, like a Cisco Callmanager believing that there is actual in-band media available<!---->
''Status:''
pbx.cpp<br/>q931.cpp<br/>q931.h<br/>te_tbl.tbl<br/>nt_tbl.tbl<br/>isdn_interop.xsl
      
      
=== INCA phones - monitoring a headset conversation via handset (headset-spy) did not work ===
=== H.323 slowstart avoid sending duplicate TerminalCapabilitySet messages ===


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|}
|}
problem: the /headset-spy option was skipped because of a bug in the driver option handler<br/> <br/>solution: fix in code<br/> <br/>files: inca_dsp.cpp<br/><br/>products: all phones<br/><br/>risks: None<br/><!---->
If a media re-negotiation happened on a remote system at a time the local H.245 channel was not even established, it could happen that a sequence of TCS, TCS0 and TCS again was sent to a calling system. This irritated especially a Cisco Call Manager.<br/><br/>This happened for example, if a call was received from the call manager on one PBX, which was routed to another PBX on which a CFNR was configured.<!---->
''Status:''
''Status:''
checked in to 9.00,8.00,09-80500
h323ch.cpp
      
      
=== IP-DECT Radio call statistics ===
=== H.323 Slowstart media renegotiation did not work if TCS was not yet received ===


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|}
|}
problem: Radio call statistics like call or handover counter are missed in the master radios overview GUI for DECT deployment.<br/><br/>solution: Radio call statistics added.<br/><br/>files: dectmaster.h, dectmaster.cpp, dectmaster_radios.xsl (OEM), dectradio.h, dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
This caused a CFNR not being executed (call was cleared on the original called endpoint, but was not sent to new destination) for calls from Cisco Call Manager<!---->
''Status:''
h323ch.cpp
      
      
=== IP-DECT Handset's product number and software version ===
=== PBX Mobility: Filters were not evaluated for mobility calls ===


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|[http://mantis.innovaphone.com/view.php?id=50738 50738]
|[http://mantis.innovaphone.com/view.php?id=59398 59398]
|}
|}
problem: The DECT handset's product number and software version are not shown in the user list in the DECT master.<br/><br/>solution: Information is shown if available.<br/><br/>files: dect_users_right.xsl (OEM).<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
Calls from mobile phones thru the mobility object were not affected by filter configurations for the user<!---->
''Status:''
pbx_mobility.cpp
      
      
=== in  any phone recording mode the recorder gets number and/or h323id of the  currently connected remote party ===
=== SNMP, ifSpeed wrong ===


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|}
|}
problem: without this infomation the identification of the remote party in an recorded call requires syncronisation with log files not directly available in the recorder.<br/> <br/>solution: a CT-COMPLETE with number/h323id of the remote party is sent to the recorder whenever the remote party changes because of a consultation call or a call transfer.<br/> <br/>files: appp_form.cpp, app_fkey.cpp, app_disp.cpp, app_ctl.cpp, app_ctl.h, app_call.cpp <br/><br/>products: all phones<br/><br/>risks: Minimal for recorders not able to deal with CT-COMPLETE info<br/><!---->
SNMP, ifSpeed wrong<!---->
''Status:''
checked in to 8.00,9.00,09-80500
      
      
=== Phone: Fkey "Partner" should try to subscribe for Presence only if checkmark set ===
=== SIP: Media negotiation problem in some early media scenarios ===


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|}
|}
Problem: Fkey "Partner" tries to subscribe for partner's presence. Even on PBXs not supporting Presence.<br/><br/>Solution: Added checkmark to Partner fkey config.<br/> <br/>Files: phone_config.h/cpp phone_edit.cpp app_fkey.cpp fkey_edit_partner.xsl <br/><br/>Products affected: Phones with partner keys only<br/><br/>Risk: No risk.<!---->
SIP/H323 interworking problem.<br/>Call was terminated with "504 Server Time-out" and "Recovery on timer expiry (102)"<!---->
''Status:''
sip.cpp
      
      
=== LDAP Searches for unsupported DNs disconnected all LDAP connections ===
=== Phone IP150 - dialling numbers containing asterisks '*' does not work ===


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|}
|}
problem: LDAP Searches for unsupported Distinguished Names (DN) disconnected all LDAP connections<br/><br/>solution: Remove (meanwhile surplus) v7 code<br/><br/>files: ldapsrv.cpp<br/><br/>products: all PBX products<br/><br/>risks: None<!---->
if in offhook mode the asterisk key is pressed for a short time the key is ignored, if it is pressed longer it is evaluated as mute key.<!---->
      
      
=== Better norwegian translation for telephone text entries ===
=== SIP: Registration refresh interval not parsed from REGISTER response if behind NAT ===


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|Id
|[http://mantis.innovaphone.com/view.php?id=50941 50941]
|[http://mantis.innovaphone.com/view.php?id=59826 59826]
|}
|}
problem: Better norwegian translation for telephone text entries<br/><br/>solution: Changed translation file<br/><br/>files: phone/txt/phonetxt.base<br/><br/>products: All telephones<br/><br/>risks: none<br/><br/><!---->
Registration refresh interval not parsed from REGISTER response if behind NAT.<br/>Wrong handling of 'received' and 'rport' parameters in Via header (RFC-3581).<!---->
''Status:''
REGISTER sip:talk.arcstel.netpbx5.net SIP/2.0<br/>Proxy-Authorization: Digest username="1295_1",realm="talk.arcstel.netpbx5.net",nonce="12935856813:4d1dfa2cd75027df50e51d433f90d3a6",response="09e7e72f21d1772b12b73dffb5b51e3c",uri="sip:talk.arcstel.netpbx5.net",qop=auth,cnonce="b35c9f24e909d311",nc=00000001,algorithm=MD5<br/>Via: SIP/2.0/UDP 192.168.0.34:2057;branch=z9hG4bK-E9764661;rport<br/>From: <sip:1295_1@talk.arcstel.netpbx5.net>;epid=00013e01b12b;tag=847121008<br/>To: <sip:1295_1@talk.arcstel.netpbx5.net><br/>Call-ID: fc72cde0e909d3119b2500013e01b12b@192.168.0.34<br/>CSeq: 1001 REGISTER<br/>Contact: <sip:1295_1@192.168.0.34:2057;transport=UDP>;expires=3600<br/>Content-Length: 0<br/>Expires: 3600<br/>Max-Forwards: 70<br/>User-Agent: (Ascom IP-DECT Base Station/ [4.1.24/4.1.24/IPBS1-A3/4C])<br/>Allow-Events: reg,dialog,message-summary,presence<br/><br/>SIP/2.0 200 OK<br/>Via: SIP/2.0/UDP 192.168.0.34:2057;received=89.233.254.81;branch=z9hG4bK-E9764661;rport=58537<br/>From: <sip:1295_1@talk.arcstel.netpbx5.net>;epid=00013e01b12b;tag=847121008<br/>To: <sip:1295_1@talk.arcstel.netpbx5.net>;tag=5439c50a<br/>Call-ID: fc72cde0e909d3119b2500013e01b12b@192.168.0.34<br/>CSeq: 1001 REGISTER<br/>Contact: <sip:1295_1@192.168.0.34:2057;transport=UDP>;expires=54<br/>User-Agent: Advoco/5.0.3046<br/>Content-Length: 0
      
      
=== Manufacturer URL is needed in static HTML pages  ===
=== PBX: Call was possible from registration as standby PBX ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=50950 50950]
|[http://mantis.innovaphone.com/view.php?id=59844 59844]
|}
|}
problem: Manufacturer URL is needed in static HTML pages <br/><br/>solution: added %U option to servlet_vars.cpp<br/><br/>files: servlet_vars.cpp<br/>products: all<br/><br/>risks: low risk <!---->
A standby PBX registers at the active PBX to check if it is alive. This registration could be misused for calls. It could be done with H.323 and SIP. This fix prohibits calls from this registration and allows registration with H.323 only<!---->
''Status:''
pbx.cpp
      
      
=== Phone: Fkey "Partner" show presence activity even if partners presence status is "closed" ===
=== Phone - switch off microphone while sending DTMF as voice data, increase volume of DTMF tones sent as voice data  ===


{|
{|
Line 2,181: Line 2,323:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=50952 50952]
|[http://mantis.innovaphone.com/view.php?id=59846 59846]
|}
|}
Problem: Fkey "Partner" does not show presence activity if partners presence status is "closed" (not registered).<br/><br/>Solution: Show presence activity regardless of the status.<br/> <br/>Files: app_disp.cpp<br/><br/>Products affected: Phones with partner keys only<br/><br/>Risk: No risk.<!---->
When "Registration x/General/No DTMF Detection" is checked DTMF tones are sent as voice data. Detection of such tones at the receiving side may fail when mixed with microphone input. <!---->
      
      
=== Ringing style upon incoming message is not configurable via web - ui ===
=== PBX CDR records with a size  near 1kB or larger were garbled when sent via HTTP ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=50963 50963]
|[http://mantis.innovaphone.com/view.php?id=59966 59966]
|}
|}
problem: Ringing style upon incoming message is not configurable via web - ui <br/><br/>solution: fixed in xsl<br/><br/>files: reg_edit_general.xsl<br/><br/>products: all telephones<br/><br/>risks: none<br/><br/><!---->
PBX CDR records with a size near 1kB or larger were garbled when sent via HTTP because of an encoding bug. Locally logging worked correct.    <!---->
      
      
=== Flashdir: Comparison for 'guid' could fail ===
=== Phone: Translation for presence activities ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=50965 50965]
|[http://mantis.innovaphone.com/view.php?id=60119 60119]
|}
|}
problem: Comparison for 'guid' could fail<br/><br/>solution: apply binary comparison (was case insensitive)<br/><br/>files: flashdir.cpp<br/><br/>products: all PBX products<br/><br/>risks: None<br/><!---->
  Abwesend, Beschäftigt, Mittagessen, Besprechung, Urlaub<br/> Away, Busy, Lunch, Meeting, Vacation<br/> Parti, Occupé, Déjeuner, Réunion, Vacances<br/> Assente, Occupato, Pranzo, Riunione, Ferie<br/> Ausente, Ocupado, Comida, Reunión, Vacaciones <br/> Fravær, Opptatt, Lunsj, Møte, Ferie<br/><!---->
      
      
=== Phone: access to PBX directories failed if the PBX System Name contained non ascii characters (&gt;= 128) ===
=== PBX: Trap if a call from mobile endpoint was diverted to a waiting queue, with altert Timeout ===


{|
{|
Line 2,214: Line 2,356:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=50974 50974]
|[http://mantis.innovaphone.com/view.php?id=60161 60161]
|}
|}
problem: the LDAP API expects latin1 but the name was utf8 encoded<br/><br/>solution: convert name to latin1 before passing to API<br/><br/>files: phone_dir.cpp<br/><br/>products: all phones<br/><br/>risks: none<br/><!---->
A NULL pointer access happend in this case while sending the ALERT message<!---->
''Status:''
''Status:''
checked in to 8.00,9.00,09-80500
pbx_wait.cpp
      
      
=== IP-DECT memory leak ===
= V8 Hotfix11 (80500.34) =
Changes included in Version 8 hotfix11
[http://mantis.innovaphone.com/view.php?id=60189 Definition]
 
== New Features ==
 
 
   
=== SIP: Interop flag for Avaya: /no-t38-in-initial-offer ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=50975 50975]
|[http://mantis.innovaphone.com/view.php?id=59176 59176]
|}
|}
problem: There are memory leaks with update event of uninitialized radio registrations.<br/><br/>solution: Cleanup added.<br/><br/>files: dectmaster.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
config change SIP /no-t38-in-initial-offer<br/>Can be used to suppress T.38 capability indication in initial SDP offer.<br/>A switch to T.38 fax mode may follow, if T.38 is enabled at the interface.<!---->
      
      
=== Ringing tone used for incoming message can not be reconfigured permanently ===
=== SIP: Add PAI/PPI header to 200/Ok for INVITE ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=50976 50976]
|[http://mantis.innovaphone.com/view.php?id=60249 60249]
|}
|}
problem: Ringing tone used for incoming message can not be reconfigured permanently. It switches back to default after ringing once without user interaction.<br/><br/>solution: fixed in code<br/><br/>files: phone/app/app_ctl.cpp<br/><br/>products: all telephones<br/><br/>risks: none<br/><br/><br/><br/><!---->
Some SIP servers wants us to send P-Asserted-Identity/P-Preferred-Identity header in final INVITE response.<!---->
      
      
=== Potential trap with Mobility ===
=== IP-DECT: number map for incoming calls (OEM) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=50989 50989]
|[http://mantis.innovaphone.com/view.php?id=60294 60294]
|}
|}
problem: A trap could happen with a collision of CFNR Timeout and call disconnect, when calling a mobile endpoint, because of NULL pointer access.<br/><br/>solution: Check for NULL pointer added<br/><br/>files: pbx_mobility.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Number map for incoming calls added for OEM devices.<!---->
      
      
=== IP-DECT hanging calls ===
=== SIP: Add PAI/PPI header to 181 response for INVITE ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51007 51007]
|[http://mantis.innovaphone.com/view.php?id=60438 60438]
|}
|}
problem: Sometime there are hanging calls in the radio.<br/><br/>solution: New timer added to check for hanging calls.<br/><br/>files: dectradio.h, dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
To get full identity information of the new remote partner<!---->
      
      
=== IP-DECT OEM module MSF trap ===
=== SIP: Module option /share-local-port ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51008 51008]
|[http://mantis.innovaphone.com/view.php?id=60542 60542]
|}
|}
problem: Traps occur after using of the MSF module.<br/><br/>solution: Pointer cleanup added.<br/><br/>files: dectmsf.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
This option forces outbound TCP signaling connection to be bound to the same local port as the signaling interface is listening on.<br/>(In order to make the remote peer do connection reuse)<!---->
      
      
=== Mobility Object returns busy if called from a unknown mobile phone ===
== Bug Fixes ==
 
 
   
=== SIP: Handling of re-INVITE w/o SDP offer while in held (inactive) state ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51010 51010]
|[http://mantis.innovaphone.com/view.php?id=60296 60296]
|}
|}
problem: The mobility object answers calls only if called by a mobile phone which is configured as forking destination. Calls from other mobile phones are rejected. The cause "user busy" was used in this case, which was misleading.<br/><br/>solution: Use cause "Service unavailable, unspecified" instead.<br/><br/>files: pbx_mobility.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
A re-INVITE w/o SDP offer while in held (inactive) state must be answered with 200/Ok containing an sendrecv offer (not inactive).<!---->
      
      
=== Potential trap when disconnecting call, WEBMEDIA-CH.5 default(82c09798): serial_event(814) ===
=== SIP: SRTP re-negotiation failed sometimes ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51132 51132]
|[http://mantis.innovaphone.com/view.php?id=60387 60387]
|}
|}
problem: Under special timing conditions a trap could happen during call disconnect. This only happened if the call terminated at a physical interface on the given box.<br/><br/>solution: Cleaning up of media channel fixed<br/><br/>files: media.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
After switching to non-encrypted media (MOH) the re-negotiation for encrypted media failed (on CCM).<!---->
      
      
=== Trap in SRTP socket ===
=== PBX: Slave license update period too short ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51146 51146]
|[http://mantis.innovaphone.com/view.php?id=60390 60390]
|}
|}
Problem: Under special conditions SRTP sockets send events to serials that are already deleted.<br/><br/>Solution: Check if the destination does still exist before sending the event.<br/><br/>Files: srtp_cipher.cpp<br/><br/>Risk: Small risk of damaging SRTP encryption on IP6000/IP2000<br/><br/><!---->
was 100s (v8) or 10s (v7) should be 10min<!---->
''Status:''
pbx.h
      
      
=== One way media after SRTP renegotiation on IP6000 ===
=== Gateway: Trap on early RELEASE from calling side ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51148 51148]
|[http://mantis.innovaphone.com/view.php?id=60400 60400]
|}
|}
Problem: On the IP6000 platform the SRTP ROC was not reset on media renegotiation.<br/><br/>Solution: Reset SRTP ROC when rtp_channel::set_media_config is called<br/><br/>Files: srtp_cipher.cpp<br/><br/>Risk: no risk known<br/><br/><br/><br/><!---->
Trap when Notification Indicator is received with ALERT while peer call is released already.<!---->
      
      
=== SIP: Remote number update after pick-up does not work ===
=== IP-DECT: potential trap ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51268 51268]
|[http://mantis.innovaphone.com/view.php?id=60406 60406]
|}
|}
Problem: PBX send UPDATE with changed From URI (rfc4916) too early (interfering with ongoing INVITE transaction). UPDATE is rejected by picking party.<br/><br/>Solution: Postpone UPDATE(from-change) until INVITE transaction is complete.<br/> <br/>Files: sip.cpp/h<br/><br/>Products affected: PBXs with SIP endpoints doing call pick-up<br/><br/>Risk: No risk.<!---->
Some pointer checks are added to prevent traps.<!---->
      
      
=== IP-DECT OEM multi-cast module support ===
=== PBX Waiting object: Problem with announcements from Boolean Object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51269 51269]
|[http://mantis.innovaphone.com/view.php?id=60421 60421]
|}
|}
problem: Some new functions are needed for the oem multi-cast module support.<br/><br/>solution: Function added.<br/><br/>files: signal.h, signal.cpp, dectusers_if.h, dectusers.h, dectusers.cpp, dectlocalusers.h, dectlocalusers.cpp, dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
The announcement worked, but if DTMF dialing to another Waiting object was done, DTMF dialing on this second Waiting object did not work anymore.<!---->
''Status:''
pbx.cpp<br/>pbx_api.h<br/>pbx_wait.cpp
      
      
=== wrong calculations causing check disc to damage data ===
=== PBX CF Loop detection indicated loop with CFNR even if there was no loop ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51305 51305]
|[http://mantis.innovaphone.com/view.php?id=60427 60427]
|}
|}
Problem: check disc could produce damaged data in certain situations<br/><br/>Solution: correctly calculate partial records and clusters for next run. Also check if clusters are used multiple times.<br/><br/>Files: fat32.cpp, fat32.h, fat32.xsl<br/><br/>Risk: minor risk<!---->
A CFNR loop is only detected if the CFNRs are executed because of no registration. The loop was detected with a single Object without registration instead of only detecting the loop if all objects are without registration<!---->
''Status:''
pbx.cpp
      
      
=== IP-DECT OEM protocol display update ===
=== H.323: If INFO was sent with cdpn and kp it could happen that it was forwarded with cdpn in SETUP and kp in INFO ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51354 51354]
|[http://mantis.innovaphone.com/view.php?id=60443 60443]
|}
|}
problem: If a OEM protocol is used the display update wrongly inserts the last pre-dialed digit as post-dialed keypad info.<br/><br/>solution: Function fixed.<br/><br/>files: dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
If a call was established by the application (or incoming signaling) without dialing information and then before the TCP connection was established a INFO message was sent with keypad and called-party-number, the call (SETUP) was sent with the called-party-number followed by an INFO with keypad.<br/><br/>This could result in a duplication of the dialed digits.<br/><br/>Only in special OEM scenarios, because keypad is usually not used.<!---->
''Status:''
h323_tbl.h
      
      
=== When upgrading a phone to V8 directories having been disabled in V7 may come up enabled in V8 ===
=== editing function keys via WEB interface broken after invalid characters have been entered in an e164 number field ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51381 51381]
|[http://mantis.innovaphone.com/view.php?id=60468 60468]
|}
|}
problem: to save space in flash the default V8 directory configuration is not stored in xml-config. When a V7 config is merged to a default V8 config a default enable='1' may override an enable='0' from V7 (V7 does not write bools with a value '0' to xml config) <br/><br/>solution: fix wrong overrides by checking for V7 specific config patterns<br/><br/>files: phone_user.cpp<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
xml syntax characters like < > &amp; entered in a number field were not encoded on output and thus garbled the xml structure    <!---->
''Status:''
checked in to 8.00,9.00,09-80500
      
      
=== PBX-CDR: Local Time wrong (same as UTC) ===
=== Memory leak when configuring H.323 NAT ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51415 51415]
|[http://mantis.innovaphone.com/view.php?id=60474 60474]
|}
|}
problem: In the CDRs from the PBX the local time was always set to UTC<br/><br/>solution: Use correct time<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Memory leak when configuring H.323 NAT<!---->
      
      
=== IP-DECT wrong forward of internal information event ===
=== possible trap with enabled trace flag on CF checkdisc ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51418 51418]
|[http://mantis.innovaphone.com/view.php?id=60513 60513]
|}
|}
problem: A internal information event is wrongly forwarded to the PBX.<br/><br/>solution: Forward of this event is avoided.<br/><br/>files: dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
The box could trap while checking the card, if the trace flag for CF0 was enabled.<!---->
      
      
=== PBX: No Inband Disconnect for Gateway Object ===
=== PBX/SOAP: Potential trap when disconnecting a mobility call ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51421 51421]
|[http://mantis.innovaphone.com/view.php?id=60538 60538]
|}
|}
problem: 'No Inband Disconnect' was not configurable for Gateway objects<br/><br/>solution: Configuration added<br/><br/>files: pbx_edit_gw.xsl<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
If a SOAP application (e.g. TAPI) disconnects a call to/from a mobile user, a trap could happen<!---->
''Status:''
pbx_xml.cpp
      
      
=== PBX CGPN missing with call to mobile endpoint, if not supplied by calling endpoint ===
=== PBX DECT System object: DECT parameters got lost, when changing critical flag ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51424 51424]
|[http://mantis.innovaphone.com/view.php?id=60565 60565]
|}
|}
problem: If a calling endpoint registered to the PBX, did not supply the calling number, the PBX did not set it, when calling a mobile endpoint<br/><br/>solution: PBX sets calling number<br/><br/>files: pbx_mobility.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
The object was written back to flash without the parameters stored by the DECT system<!---->
''Status:''
pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_dect.cpp<br/>pbx_dect.h
      
      
=== PBX BC Conference member type restriction / call information ===
=== PBX SOAP Admin: Critical flag could not be set in object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51429 51429]
|[http://mantis.innovaphone.com/view.php?id=60568 60568]
|}
|}
problem: Some other PBX objects can not be called as conference members. Conference object call target is not shown correctly in the PBX call list. This information is also used as calling party number for the other conference member calls, useful for recording with the VM object and several conference objects.<br/><br/>solution: PBX object type restriction removed and remote endpoint information (cgpn) fixed.<br/><br/>files: pbx_bc_conf.cpp.<br/><br/>products affected: All devices with PBX.<br/><br/>risk: Minimal risk of collateral damage. <!---->
The attribute "critical" was not allowed<!---->
''Status:''
pbx.cpp
      
      
=== no RTP-DTMF after rerouting ===
=== Ldap Replication, Problems with Percent-Char in Password ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51431 51431]
|[http://mantis.innovaphone.com/view.php?id=60611 60611]
|}
|}
problem: If rerouting happened from one media endpoint to another, for example if a TONE interface is used for a dialtone after one digit dialed there is a rerouting to another interface, RTP-DTMF does not work.<br/><br/>solution: Media renegotiation fixed for this case<br/><br/>files: ac_dsp.h, ac_dsp2.h, ac_dsp3.h<br/><br/>risks: none<!---->
Ldap Replication, Problems with Percent-Char in Password<!---->
      
      
=== PBX BC conference object TAPI feature clear call ===
=== Optional display of text provided in the Display Information Element of an ISDN Information Message ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51434 51434]
|[http://mantis.innovaphone.com/view.php?id=60612 60612]
|}
|}
problem: The TAPI connection of the broadcast conference object does not support clearing calls.<br/><br/>solution: Feature added.<br/><br/>files: pbx_bc_conf.h, pbx_bc_conf.cpp.<br/><br/>products affected: All devices with PBX.<br/><br/>risk: Minimal risk of collateral damage. <!---->
The text provided in the Display Information Element of an ISDN Information Message is displayed at the phone status line.<br/>This may be supressed now by checking "Phone/Preferences/Hide Display Info from ISDN Providers" <br/><!---->
      
      
=== dyn PBX General configuration page changes did not work sometime ===
=== SIP: Authentication issue (AVAYA-SM interworking) ===


{|
{|
Line 2,460: Line 2,626:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51436 51436]
|[http://mantis.innovaphone.com/view.php?id=60712 60712]
|}
|}
problem: Sometimes strange behaviour, when removing config like 'Route Master calls if no Master to' or 'Max Calls to Master'<br/><br/>solution: editor fixed<br/><br/>files: pbx_admin.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Another re-try with authentication required.<!---->
      
      
=== SIP: NOTIFY(message-summary) not handled by PBX ===
=== Group Indication with a diverting number of zero length caused a encoding error ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51443 51443]
|[http://mantis.innovaphone.com/view.php?id=60715 60715]
|}
|}
Problem: NOTIFY(message-summary) was not handled by PBX (server side). Only by phones (client side)<br/><br/>Solution: Implement handling of unsolicited NOTIFY(message-summary) at server side.<br/> <br/>Files: sip.cpp/h<br/><br/>Products affected: PBX with SIP clients<br/><br/>Risk: No risk. <!---->
The number should not be sent at all. This happend if a group indication was to be sent from a call which was diverted by an object without number<!---->
''Status:''
h450.cpp
      
      
=== PBX: Retrieve was not sent in case of chained Waiting Queues ===
=== PBX Waiting: Don't forward DTMF to announcement source ===


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|[http://mantis.innovaphone.com/view.php?id=60838 60838]
|}
|}
problem: When using DTMF destinations with Waiting Queues, the waiting queue is sending a Hold Notific when DTMF map destination is alerting. A Retrieve Notific must be sent when the destination connects. This was missing if the destination was another Waiting Queue.<br/><br/>solution: Send missing Retrieve<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Announcement source could be a boolean object and DTMF could change the state of the boolean<!---->
''Status:''
pbx_wait.cpp
      
      
=== PBX: Busy Name was not sent if busy because of 'Busy on ... calls' ===
=== IP-DECT: cause code changed ===


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|[http://mantis.innovaphone.com/view.php?id=60958 60958]
|}
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problem: No Name Identification Facility was sent if call was busy because of 'Busy on ... calls'<br/><br/>solution: Send Name Id<br/><br/>files: pbx.cpp, pbx_api.h<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
The cause code is changed to "cause unassigned number" if the call is released because no radios are available.<!---->
      
      
=== IP72: beacon recv time now configurable through command line ===
=== Fix for SIP requests with 10+ header instances ===


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|[http://mantis.innovaphone.com/view.php?id=61014 61014]
|}
|}
problem: beacon recv time now configurable through command line. This is required for a special Meru Networks interop. (config change WLAN0 /beacon-recv-time 10)<br/><br/>solution: fixed in code<br/><br/>files: ascom-drivers/WLAN_drv.cpp<br/><br/>products: IP72<br/><br/>risks: none<br/><!---->
Response to following INVITE request did not returned all Via headers:<br/><br/>INVITE sip:229@192.168.193.181:2058;transport=UDP SIP/2.0<br/>Record-Route: <sip:145bf82@192.168.193.210;transport=udp;lr><br/>Record-Route: <sip:192.168.193.219:15060;lr;sap=433098584*1*016asm-callprocessing.sar-624908352~1296718381566~-535462628~1><br/>From: "H323-2" ;tag=8084387dbc40e01d7f4d42da8200<br/>To: <sip:229@localdomain.com><br/>Call-ID: 8084387dbc40e01d8f4d42da8200<br/>CSeq: 1 INVITE<br/>Via: SIP/2.0/UDP 192.168.193.210;rport;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099903-AP;ft=192.168.193.210~13c4<br/>Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099903<br/>Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199901<br/>Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199900<br/>Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bK8084387dbc40e01d7f4d42da8200-AP;ft=6565<br/>Via: SIP/2.0/TLS 192.168.193.104;branch=z9hG4bK8084387dbc40e01d7f4d42da8200;avaya-cm-term-reaction=shortcut<br/>Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099899-AP;ft=7355<br/>Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099899<br/>Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199897<br/>Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199896<br/>Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bK8084387dbc40e01d9f4d42da8200-AP;ft=6565<br/>Via: SIP/2.0/TLS 192.168.193.104;branch=z9hG4bK8084387dbc40e01d9f4d42da8200<br/>Supported: 100rel,histinfo,join,replaces,sdp-anat,timer<br/>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH<br/>User-Agent: Avaya CM/R016x.00.1.510.1 AVAYA-SM-6.1.0.0.610012<br/>Contact: "H323-2" <sip:201@192.168.193.104:5061;transport=tls><br/>Accept-Language: en<br/>Accept-Contact: *;+avaya-cm-line=1<br/>Alert-Info: <cid:internal@localdomain.com>;avaya-cm-alert-type=internal<br/>History-Info: <sip:229@localdomain.com>;index=1<br/>History-Info: "229" <sip:229@localdomain.com>;index=1.1<br/>Min-SE: 1200<br/>P-Asserted-Identity: "H323-2" <sip:201@localdomain.com><br/>Record-Route: <sip:145bf82@192.168.193.210;transport=tls;lr><br/>Record-Route: <sip:192.168.193.219:15061;transport=tls;lr;sap=433098584*1*016asm-callprocessing.sar-624908352~1296718381477~-535462632~1><br/>Record-Route: <sip:145bf82@192.168.193.210;transport=tls;lr><br/>Record-Route: <sip:192.168.193.104:5061;transport=tls;lr><br/>Session-Expires: 1200;refresher=uac<br/>Content-Type: application/sdp<br/>Content-Length: 178<br/>P-Location: SM;origlocname="Interoplab";termlocname="Interoplab"<br/>Max-Forwards: 63<br/><br/>v=0<br/>o=- 1296719515 1 IN IP4 192.168.193.104<br/>s=-<br/>c=IN IP4 192.168.193.105<br/>b=AS:64<br/>t=0 0<br/>m=audio 2564 RTP/AVP 8 18 96<br/>a=fmtp:18 annexb=no<br/>a=rtpmap:96 telephone-event/8000<br/><!---->
      
      
=== Make LCD dump to be displayed in browser ===
=== SIP: Do not send INFO(dtmf) before call is connected ===


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|[http://mantis.innovaphone.com/view.php?id=61025 61025]
|}
|}
Problem: LCD dump was displayed by external program.<br/><br/>Solution: Fix Content-Type of lcd_dump.bmp to make browsers display it.<br/> <br/>Files: http.cpp<br/><br/>Products affected: All phones<br/><br/>Risk: No risk.<!---->
Do not send INFO(dtmf) before dialog is in confirmed state.<!---->
      
      
=== Config Wizard Update ===
= V8 Hotfix12 (80500.36) =
Changes included in Version 8 hotfix12
[http://mantis.innovaphone.com/view.php?id=60894 Definition]
 
== New Features ==
 


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=51501 51501]
|}
problem: Several issues with config wizard: CLIP no screening mappings for international calls wrong, CLIP no screening mappings did not handle internal numbers matching Trunk/National/International Prefix, switchboard waiting object was not configured, extern only needed for insert mode<br/><br/>solution: config wizard fixed<br/><br/>files: setup.cpp, ip800/config_wizard.txt, ip6000/config_wizard.txt, ip6010/config_wizard.txt, ip24/config_wizard.txt<br/><br/>products: IP30x, IP800, IP6000, IP6010<!---->
      
      
=== function keys defined in a config template could not be overloaded by an associated user object ===
=== Phone: New config option "Proxy" for SIP registrations ===


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|[http://mantis.innovaphone.com/view.php?id=59396 59396]
|}
|}
problem: a function key defined in a template could not be overridden by an associated user object, the key in the template did always win. <br/><br/>solution: the changed function key must be kept in user object<br/><br/>files: phone_config.cpp<br/><br/>products: all PBX and phones<br/><br/>riscs: none<br/><!---->
Now DNS names can be specified.<br/>Replaces config option "Primary Server Address".<!---->
''Status:''
checked in to 8.00,9.00,09-80500
      
      
=== IP72: Upgrade WLAN subsystem to Ascom 1.7.10 ===
=== phone: " reject if busy" option for incoming announcement calls ===


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|[http://mantis.innovaphone.com/view.php?id=61412 61412]
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problem: Upgrade WLAN subsystem to Ascom 1.7.10<br/>Ascom i75 v1.7.10 release.<br/>- Beacon reception time can be changed.<br/>- Scan interval can be changed.<br/>- Null data keep alive period can be changed.<br/>- Two different site filters can be chosen.<br/>- RSSI filter parameters is changed: 25% new value + 75% old value (previous releases use 10% + 90%).<br/>- Roaming threshold is changed to -67 dBm (from -70).<br/>- Authentication timeout changed to 100 ms (from 500 ms).<br/><br/>solution: upgraded shared code<br/><br/>files: WLAN/* ascom-drivers/WLAN_drv.*<br/><br/>products: IP72<br/><br/>risks: none known<br/><!---->
In some scenarios it's required that announcement calls are not accepted when the phone is busy.<!---->
      
      
=== Phone: Presence subscription of partner fkey not created sometimes ===
=== v8 Firmware for IP6010, IP3010, IP1060, IP0010 ===


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|[http://mantis.innovaphone.com/view.php?id=61522 61522]
|}
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Problem: In some cases the presence subscription of the partner fkey was not established.<br/><br/>Solution: Fix lookup of existing presence subscription.<br/> <br/>Files: phonesig.cpp<br/><br/>Products affected: Phones with partner fkeys<br/><br/>Risk: No risk.<!---->
Version 8 Firmware will be released for the new IP6010 Gateway familiy as part of a hotfix release.<!---->
      
      
=== Problems with Mobility and Nodes ===
=== IP-DECT: Abnormal call release error event ===


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|[http://mantis.innovaphone.com/view.php?id=61705 61705]
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problem: When calling from a mobile handset belonging to a user which is configured in a node a CLI without the node prefix was sent. Using a user configured in a node was not possible to use as mobile endpoint.<br/><br/>solution: handle node prefixes<br/><br/>files: pbx_mobility.cpp<br/><br/>products: all with PBX<br/><br/>risks: little risk of collateral damage with mobility<!---->
Now the DECT Master sends an error event to the event logger every time if an abnormal call release occurs.<!---->
      
      
=== SIP: Send Call-Info header with "answer-after=0" for auto answer signaling ===
=== new: phonesig api method to restart registration process without deregistration ===


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|[http://mantis.innovaphone.com/view.php?id=62165 62165]
|}
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Problem: Snom phones (and others) do not support Answer-Mode header (RFC-5373). But they honour "anser-after" parameter in Call-Info header.<br/><br/>Solution: Send Call-Info header with "answer-after" header.<br/> <br/>Files: sipmsg.cpp/h siptrans.cpp<br/><br/>Products affected: All SIP PBXs<br/><br/>Risk: No risk.<!---->
WLAN phones we need a way to restart a RAS registration when coming back from a out-of-coverage condition to syncronize the handsets and PBX's registration state.<!---->
      
      
=== IP-DECT trap during call release and information message ===
== Bug Fixes ==
 
 
   
=== IP2x/30x: T.38: Option for  high speed data redundancy  ===


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problem: A trap occurs if the DECT handset sends a information message and the call is release by the PBX. Only the channel is released, but not yet the call.<br/><br/>solution: Null pointer check added.<br/><br/>files: dectmaster.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
to configure this option use <br/>http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl<!---->
      
      
=== Phone: Trap when re-creating presence call ===
=== IP2x/30x: T.38: Calling tone (CNG) detect didnt work ===


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|[http://mantis.innovaphone.com/view.php?id=60879 60879]
|}
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Problem: Trap when re-creating presence call, because facility entity did not exist.<br/><br/>Solution: Re-create facility entity when re-creating call object.<br/> <br/>Files: phonesig.cpp<br/><br/>Products affected: All phones<br/><br/>Risk: No risk.<!---->
to configure this option use <br/>http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl<!---->
      
      
=== IP-DECT configuration option 'Redirection with GK ID' ===
=== IP3xx: Trap if switching a PBX from Standy to Off ===


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problem: Configuration option needed to append the GK ID if the registration is redirected by the PBX.<br/><br/>solution: Configuration option added.<br/><br/>files: dectmaster.h, dectmaster.cpp, dectmaster.xsl.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
This happens because we try to unregister from a CONF interface, which does not exist on the IP3xx platform<!---->
''Status:''
pbx.cpp
      
      
=== PBX: Pickup call did not show original called/parked endpoint ===
=== SIP: Trap when receicing provisional response for obsolete INVITE ===


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|}
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problem: When doing pickup, the to be picked up call did not show what endpoint was called. This is especially a problem if group pickup is used with a function key without display<br/><br/>solution: Add ct_setup/leg2 info to pickup call<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
In overlap dialing scenarios overlapping INVITE client transactions are used.<br/>Same Call-ID, different CSeq and different To-URI.<!---->
      
      
=== compatibility issue with PBX Waiting queue sending ct-complete before connect ===
=== SIP: Read PAI/PPI header when receiving MESSAGE request ===


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|[http://mantis.innovaphone.com/view.php?id=61086 61086]
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problem: If a call alerting at a PBX waiting queue is connected by a operator, the PBX is sending out a ct-complete message to indicate to the caller, which operator connected. This was sent right before the connect, but ct-complete is allowed by the standard only after connect. This created an interworking issue with when this was sent out to a QSIG PBX<br/><br/>solution: send ct-complete after connect<br/><br/>files: pbx_wait.cpp<br/><br/>products: all with PBX<br/>risks: none<!---->
Read PAI/PPI header when receiving MESSAGE request in order to get calling party identity<!---->
      
      
=== PBX: SOAP initiated calls were sent with CT-SETUP ===
=== Phone: Memory leak when deleting SIP registration ===


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problem: If a call was initiated by SOAP to the PBX, the outgoing call contained a CT-SETUP facility. This way the call was displayed as transfered call by the destination, but it should be displayed just the same as a call initiated on the phone itself.<br/><br/>solution: remove CT-SETUP from outgoing call<br/><br/>files: pbx_xml.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Failed to delete registration, but only if trying to delete during state "rgistration failed due to no response from server".<!---->
      
      
=== PBX send call to mobile phone as diverted call ===
=== H.450 encoding problem with call-transfer and diverting facilities, if length of number was 0 ===


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problem: Billing applications need to associate a call to a mobile phone to the respective user. This can be done with the diverting leg info.<br/><br/>solution: Add diverting leg info 2 to call to mobile phone<br/><br/>files: pbx_mobility.cpp, pbx_gi.cpp<br/><br/>products: all with PBX<br/><br/>risks: none<!---->
A zero lenght number cannot be encoded, it must be omited from the message<!---->
''Status:''
h450.cpp
      
      
=== IP72: ring though handset ===
=== SIP: Bug in handling of re-direct responses ===


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problem: IP72 feature: ring through handset if configured so and handset plugged<br/><br/>solution: fixed in code<br/><br/>files: box/omap/omap_dsp.* box/omap/omap_codec.cpp<br/><br/>products: IP72<br/><br/>risks: none<br/><br/><!---->
New remote port was not respected when maddr parameter was present in redirection URI.<br/>E.g.<br/><br/> sip:662@10.0.77.46:4432;user=phone;transport=Tcp;maddr=10.0.77.46;x-mss-call-id=a515c882e909d311874700903306177f%4010.0.77.70 <!---->
      
      
=== Possible trap when removing a cf card without previous unmount ===
=== IP2x/IP30x: T38: Missing "no signal indications" on remote initiated T.38 session ===


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Problem: events where queued to not exisiting serial<br/><br/>Solution: cf driver shouldn't answer outstanding events after removing the cf card, as the fat32 module won't wait for any events after receiving status removed event<br/><br/>Files: cf_drv.cpp<br/><br/>Risk: minor risk <!---->
This solves a problem with SIP-Provider behing a NAT router on outgoing fax calls.<br/><!---->
      
      
=== IP6000 LE newer kernel support ===
=== Critical Flag at DECT System Object disappears ===


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|[http://mantis.innovaphone.com/view.php?id=61318 61318]
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problem: Newer Linux kernel included in Debian does not work.<br/><br/>solution: Support for linux kuser helper functions added.<br/><br/>files: startup_littleendian.S.<br/><br/>products affected: Only IP6000 little-endian firmware.<br/><br/>risk: No risk of collateral damage. <!---->
If the DECT system is replicated from the PBX and systems settings are changed on the DECT system, the critical flag on the DECT System object in the PBX is lost<!---->
''Status:''
dectusers.cpp<br/>dectusers.h
      
      
=== HTTP client header access ===
=== Calls redialled from call list were not set up with CLIR although CLIR was active for the original call ===


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|[http://mantis.innovaphone.com/view.php?id=61321 61321]
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problem: access to received  httpclient headers needed in some applications,OEM Manufacturer  in httpclients user agent header needed<br/><br/>solution: added virtual function to access received http headers, user agent header uses OEM struct manufacturer as user agent string<br/>files: httpclient_i.cpp httpclient_i.h httpclient.h<br/>products: all<br/><br/>risks: low risk <!---->
The CLIR setting of the original call was saved in the call list but not applied when the call was redialled from list.<!---->
      
      
=== LDAP/Replicator-Status  "There is no replicator active" ===
=== CFNR at PBX object, was executed on call to busy endpoint ===


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Problem: When no replicator is enabled the replicator status window is showing<br/>an empty drop-down list. It should be a message shown indicating that<br/>no replicator is enabled.<br/><br/>Solution: Display "There is no replicator active"<br/><br/>Files: ldaprep_status.xsl<br/><br/>Risk: none<br/><!---->
should only be executed registration down or no respone at all<!---->
''Status:''
pbx.cpp
      
      
=== ISDN: Sending of CEI facilities as Point to Multipoint endpoint did not work ===
=== phone: function key Boolean Object with 'Toggle State' checked did not display the correct state sometimes ===


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Problem: Call independent signaling did not work on Point to Multipoint interfaces. Required for Call Completion.<br/><br/>Solution: Fixed.<br/> <br/>Files: q931.cpp<br/><br/>Products affected: BRI Gateways<br/><br/>Risk: Small risk of collateral damage.<!---->
This happened when the state of the boolean object was toggled from 'manual-on' to 'automatic-off' state at the PBX or by another phone with such a key. It did not happen when with a key where the 'Toggle State' checkmark was not set.   <!---->
      
      
=== Assertion to verify that access to license data structures is correct ===
=== SIP: No overlap sending if 'sending complete' was declared ===


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problem: There is a hint, that access to license data structures could corrupt memory.<br/><br/>solution: Verify that access to license data structures is correct with a assertion which results in a restart if this does happen.<br/><br/>files: inno_lic.cpp, pbx.cpp<br/><br/>products: all except phones<br/><br/>risks: Additional restarts could happen, but only in cases memory would be corrupt otherwise, so restart is the better choice<!---->
Do not start overlapping INVITE transaction for new dialing digit if 'sending complete' was indicated for the call.<!---->
      
      
=== IP-DECT logging release code ===
=== PBX phone config templates could overrun when a big number of function keys was configured ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=51759 51759]
|[http://mantis.innovaphone.com/view.php?id=61476 61476]
|}
|}
problem: The release code is not correctly shown in logging events.<br/><br/>solution: Fixed.<br/><br/>files: dectmaster.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
There was a general 4kB size limitation for attributes read from LDAP directory which was too small for the 'phone' attribute of a config template.   <!---->
      
      
=== Trap if some but not all interfaces of a given type are unlicensed ===
=== Webdav: Bad encoding of special characters in XML properties ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=61505 61505]
|}
|}
problem: If some but not all interfaces are unlicensed (e.g. IP6000 with 4 PRI interfaces is licensed for 2 PRI interfaces) a trap could happen any time after the Gateway config was updated.<br/><br/>solution: Access to license structure fixed<br/><br/>files: inno_lic.cpp, gk.cpp<br/><br/>products: all gateway products<br/><br/>risks: None<!---->
Bad encoding of file/folder names containing special characters.<!---->
      
      
=== IP72: function keys only available in idle mode ===
=== do not open multiple HTTP sessions when forwarding a big number of alarms in a short time ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=61527 61527]
|}
|}
problem: IP72: function keys only available in idle mode<br/><br/>solution: option /softkey-mode now defines whether a predefined function key overlays a softkey in idle mode, in active mode, or not at all <br/><br/>files: phone/forms/forms*<br/><br/>products: IP72<br/><br/>risks: none<br/><br/><!---->
when alerm forwarding is active the fault handler passed new alarms immediately to the forwarding httpclient and httpclient opens a new session when there is no idle session.<!---->
      
      
=== DSP debug ===
=== PBX: Boolean Function Key was not updated when joining group ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=52124 52124]
|[http://mantis.innovaphone.com/view.php?id=61590 61590]
|}
|}
problem: sporadic trap in ac-dsp, defect return address<br/><br/>solution: debug added to show packets sent to DSP. Enable on IP6000 with config+change+AC_DSP11+/dtrace  config+write and restart. Use not with SRTP, since CPU load with 60channel RTP is increased on the IP6000 from 66% to 77% <br/><br/>files: ac_48xhi.c<br/><br/>products: ip800 ip6000 ip1200 ip1201 ip4001<br/><br/>risks: low risk <!---->
For the Boolean function key it is required to receive Group Indications from the Boolean object, which does not happen if the phone is not member of the group (dynamic out). When joining the group an update should be sent to the phone.<!---->
''Status:''
pbx.cpp<br/>pbx.h<br/>pbx_gi.cpp<br/>pbx_gi.h<br/>pbx_bool.cpp
      
      
= V8 Hotfix  3 (80500.09, withdrawn) =
=== Possible to configure use of Feature Codes on Basic Rate ISDN ===
Changes included in Version 8 hotfix3
[http://mantis.innovaphone.com/view.php?id=51967 Definition]
 
== New Features ==
 
 
   
=== Allow to limit license usage of slave PBX ===


{|
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|[http://mantis.innovaphone.com/view.php?id=61620 61620]
|}
|}
problem: In some configuration it is desireable that it can be ensured, that a slave PBX cannot use up too many licenses from the master. This is esspecially the case with dynamic slave PBXs managed by the customer in a hosted environment.<br/><br/>solution: Limits configurable by admin login to the Host box only<br/><br/>files: pbx.cpp, pbx.h, pbx_general.xsl, config_options.cpp, config_options.h<br/><br/>products: all with PBX<br/><br/>risks: Risk of collateral damage, no trivial fix.<!---->
This configuration option is not useful on ISDN BRIs. In fact it usually results in unexpected behaviour.<br/><br/>This option is removed from the user interface.<!---->
''Status:''
ip800/platform/config.h<br/>ip24/platform/config.h
      
      
=== mem info for TLS socket ===
=== IP-DECT: OEM module update function ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=61671 61671]
|}
|}
description: Implement mem info for tls_socket objects for debugging purposes.<br/><br/>products: all<br/><br/>files: tls.h, tls.cpp<br/><br/>risk: no risk<!---->
The update function for an OEM module was changed.<!---->
      
      
=== IP-DECT interface functions for OEM modules ===
=== IP-DECT: trap with call transfer ===


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|}
|}
problem: New interface functions for OEM modules needed.<br/><br/>solution: Interface functions added.<br/><br/>files: dectusers_if.h, dectusers.h, decctmaster_if.h.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
Null pointer trap with call transfer and release event from the DECT side.<br/>Trap identification, IP1200, V8 Hotfix 10:<br/>XCPT: no 2 (TLB load)  pc 943fd6d4  ra 94278e9c  va 0000000c<br/><!---->
      
      
=== distictive ringing support for SIP registrations ===
=== PBX-SOAP: Admin function removed password if Object Long Name (cn) was changed ===


{|
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|[http://mantis.innovaphone.com/view.php?id=61725 61725]
|}
|}
problem: distictive ringing was not supported for SIP calls at all. Some SIP registrars use the 'alert-info' to identify external/internal calls. As far as our SIP stack knows the magic words the numbering plan of the called party number identifies external/internal calls.<br/><br/>solution: evaluate numbering plan for SIP calls<br/><br/>files: app_call.cpp<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
If the cn is changed the object must be identified by guid an the password of this old object is to be used<!---->
''Status:''
''Status:''
checked in to 8.00,9.00
pbx.cpp
      
      
=== Option to map a calling Name to a Number in trunk object for Outgoing calls ===
=== PBX-SOAP: Admin function could not be used to configure "phone-config" ===


{|
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|}
|}
This allows objects to be defined which are represented to the public network with a number defined as part of the name, which is different from the number used internally. For calls from the public network a called party number can be mapped back to a name depending on a configurable prefix. This allow these endpoints to be called from the outside by the same number.<!---->
"phone-config" was missing in the list of allowed attributes<!---->
''Status:''
''Status:''
pbx.cpp, pbx_admin.cpp
pbx.cpp
      
      
=== GSM License algorithm did not work for GSM version &gt;1 ===
=== SNMP, If Index sometimes missing in interfaces walk ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=53045 53045]
|[http://mantis.innovaphone.com/view.php?id=61985 61985]
|}
|}
The license checking algorithm did only work for GSM-1 license. A client version 2 and higher would not have been accepted even with correct license installed<!---->
SNMP, If Index sometimes missing in interfaces walk<!---->
''Status:''
pbx_mobility.cpp
      
      
=== Alarm on certificates that will expire soon ===
=== SIP: Very large SIP request headers were rejected with 414 Request-URI Too Long ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=53071 53071]
|[http://mantis.innovaphone.com/view.php?id=62033 62033]
|}
|}
Boxes shall throw an alarm if the box certificate or certificates in the trust list have expired or will expire during the next 30 days. To trigger the alarm the certificates are checked once an hour.<br/><br/>Files: x509.h, x509.cpp<!---->
SIP request headers larger than 2000 bytes were rejected with 414 Request-URI Too Long<!---->
      
      
=== IP-DECT user import content checks ===
=== ISDN, QSIG, NT, Invalid Progress message was sent ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=53078 53078]
|[http://mantis.innovaphone.com/view.php?id=62190 62190]
|}
|}
New file content checks for the user import function added: maximum item lengths, unsupported xls file format.<!---->
The mandatory Progress Indicator was missing in Progress message when rejecting a call. This could cause that the inband busy tone could not be sent.<!---->
''Status:''
''Status:''
dectuser_if.h, dectusers.h, dectusers.cpp
nt_tbl.h
      
      
=== IP-DECT OEM protocol release reasons ===
= V8 Hotfix13 (80500.37) =
Changes included in Version 8 hotfix13
[http://mantis.innovaphone.com/view.php?id=63025 Definition]
 
== New Features ==
 
 
   
=== CAS E1 3bit pulse dialing ===


{|
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|[http://mantis.innovaphone.com/view.php?id=62191 62191]
|}
|}
IP-DECT Release reasons for OEM protocol are changed.<!---->
Support for CAS E1 3bit pulse dialing, which is sometimes used instead of DTMF addressing.<!---->
''Status:''
dectmaster.cpp
      
      
=== IP-DECT submodule gui information interface ===
=== RPCAP uses system time instead of uptime now ===


{|
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|[http://mantis.innovaphone.com/view.php?id=53143 53143]
|[http://mantis.innovaphone.com/view.php?id=62745 62745]
|}
|}
Interface added for IP-DECT submodules to provide appending additional gui informations to the main IP-DECT page.<!---->
A wireshark capture with RPCAP will now receive packet timestamps with the system time and not the uptime anymore.<!---->
''Status:''
dectuser_if.h, dectusers.h, dectusers.cpp
      
      
=== PBX SOAP: Allow CLIR calls to be made ===
=== Gateway Routing: Support of '?' wildcards in CGPN and CDPN output ===


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|[http://mantis.innovaphone.com/view.php?id=62809 62809]
|}
|}
If the srce164 argument of UserCall starts with 'r' or 'R', the call is sent with CLIR (calling line identification restricted)<!---->
In the routing table digits received at places marked with '?' are forwarded to the respective '?' in the output number. This works for CDPN and CGPN maps in routes. It does not work in interface maps<!---->
''Status:''
''Status:''
pbx_xml.cpp
gk.cpp<br/>gk.h
   
== Bug Fixes ==
 
 
      
      
=== IP-DECT OEM module GUI menu configuration ===
=== SIP: INVITE after redirect must not contain the old remote tag ===


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|[http://mantis.innovaphone.com/view.php?id=62263 62263]
|}
|}
Mode string depends on an OEM module changed for the DECT menu configuration.<!---->
INVITE after redirect did contain the old remote tag.<br/>Now it is cleared before new INVITE is sent to new destination.<!---->
''Status:''
dectusers.cpp
      
      
=== Change packet creator information for debugging ===
=== SIP: Expect early inband information if 180 with SDP answer is received ===


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|Id
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|}
|}
Packet's creator is changed to destination module by the OS if trace is enabled for source or destination module. For debugging.<!---->
Expect early inband information if 180 with SDP answer is received<!---->
''Status:''
debug.cpp
      
      
=== IP-DECT configurable endpoint response timeout ===
=== PBX Quickdial: Transferscenario leaves orphaned call ===


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|[http://mantis.innovaphone.com/view.php?id=62311 62311]
|}
|}
New IP-DECT Master configuration option: response timeout. If the timeout is configured and the handset does not answer, the call is released with cause 'No user responding'.<!---->
PBX Quickdial: Transferscerio leaves orphaned call<br/>The orphaned call remains under PBX/Calls and cannot be cleared.<!---->
''Status:''
dectmaster.h, dectmaster.cpp, dectmaster.xsl
      
      
=== permit to adjust ring tone volume on phone while phone is ringing ===
=== License: License upload shows error "No licenses available" ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=53220 53220]
|[http://mantis.innovaphone.com/view.php?id=62318 62318]
|}
|}
While the phone is ringing the volume of the current ring tone can be changed by pressing the +/- or the left/right key now. the ring tone is restarted on each keystroke and a slider is displayed to indicate the volume level.<br/>The changed volume is saved in user config if the current ring tone is one of the tones configured under "Configuration/Registration x/Preferences/Ring Tones" or "Configuration/Registration x/Preferences/Ring Filter". The volume of ring tones assigned to a telephone directory entry is not saved. <br/><!---->
"No licenses available" when uploading license XML.<!---->
      
      
=== Stuttering sound when WLAN handset is held when using U-APSD mode ===
=== Do SRTP Re-keying when doing media renegotiation ===


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|}
|}
Problem: When WLAN handset is held (receives "sendonly" from PBX) the handset stops sending RTP. This makes MOH sound bad.<br/><br/>Solution: Now config option /no-recvonly (don't stop sending RTP, even in recvonly mode)<br/> <br/>Files: sip.cpp/h<br/><br/>Products affected: All SIP devices<br/><br/>Risk: No risk.<!---->
Using the same SRTP key could be a security issue. When after a transfer the same SRTP keys are used, in theory the party doing the transfer could still decrypt the SRTP even if not in this call anymore<!---->
''Status:''
h323ch.cpp<br/>media.cpp<br/>channel.cpp<br/>channel.h
      
      
=== phone: config option to supress the dial tone ===
=== phone: a call unparked by a phone with recording active was released instead of reconnected  ===


{|
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|}
|}
Specially when using a headset the dial tone may be annoying when for example the active call is put on hold with the R-key and the user needs some time before dialling the consultation call. This dial tone can be supressed now by setting the checkmark "Administration/Phone/Preferences/No Local Dial Tone" <!---->
When the phone receices the SETUP indicating the unparked call the call should be automatically connected and become the active call. This failed because the currently active call was not put on hold before and thus there was no free DSP cannel to connect the unparked call. <!---->
      
      
=== IP-DECT OEM option type changed ===
=== Polish Language could not be configured in the PBX Phone Config ===


{|
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|}
|}
DECT module interface option type changed for DECT OEM support.<!---->
The table entry for polish language was missing<!---->
''Status:''
signal.h, signal.cpp, dectmaster_if.h, dectmaster.h, dectmaster.cpp, dectradio.h, dectradio.cpp
      
      
=== 16 SIP Interfaces configurable on IP6000 ===
=== General btree library problem: Potential Trap if many outgoing registrations need to be retried ===


{|
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|}
|}
To allow up to 16 dynamic PBX on an IP6000 each with its own SIP trunk<!---->
Actually the problem is in the commonly used btree library, but there are not that many cases in which the libray is used in a way that create the problem<!---->
''Status:''
''Status:''
relay.cpp, relay.h, relay_api.h, gk.cpp, gk.h, gk_if.h, relay_ifs.xsl, relay_edit_route.xsl, lib.cpp, xml.cpp, xml.h, latin1.cpp, latin1.h
btree.cpp
      
      
=== phone: configurable audible signal for automatically connected inbound calls (announcement calls) ===
=== PBX Waiting: Limited DTMF targets could be added using Internet Exporer ===


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|}
|}
Announcement calls were indicated with a short tone sequence. Melody, volume and duration could not be configured.<br/><br/>Now one of the usual ring melodies and its duration may be configured under<br/>  "Configuration/Registration <n>/Preferences/Ring Tones/Announcement Calls"<br/>  "Configuration/Registration <n>/Preferences/Ring Tones/Multicast Announcement"<br/>The default 'melody' is short single tone repeated for 1,5 seconds. <br/><br/>To keep existing installations running playing of this melodies must be enabled explicitely by checking<br/> "Administration/Phone//Preferences/Play Configured Ring Melody before<br/>Automatically Connecting an Announcement Call"<br/><br/>Signaling of Announcement calls can be switched off separately per registration by checking<br/>  "Configuration/Registration <n>/Preferences/Announcement Calls/No Audible Signal"<!---->
URL size limitiation of IE -> use POST instead<!---->
''Status:''
pbx_edit_waiting.xsl
      
      
=== Direct Dial timeout configurable on analog Interfaces ===
=== PBX Waiting: Connected Number handling different from normal Connected Number Handling ===


{|
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|[http://mantis.innovaphone.com/view.php?id=62437 62437]
|}
|}
The timeout was fix 4s, but in some applications a smaller timeout (e.g. 0s) is desired<!---->
This caused different behaviour whether the operator answered the call on a SIP or H.323 phone. In case of SIP the Connected Number was sent, in case of H.323 not<!---->
''Status:''
pbx.cpp<br/>pbx.h
      
      
== Bug Fixes ==
=== SIP: Media negotiation failed when interworking with H.323 ===
 
 
   
=== innovaphone endpoints which do not support group indications can turn off sending of group indications in PBX ===


{|
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problem: High load can be generated by group indications on endpoints which do not support group indications (e.g. IP-DECT).<br/><br/>solution: These endpoints can turn off sending of group indications in the PBX<br/><br/>files: gk.cpp, pbx_api.h, pbx.cpp, pbx_gi.cpp, h323sig.cpp, h323ras.cpp, h323.h, voip.h, dectmaster.cpp<br/><br/>products: all<br/><br/>risks: Small risk of collateral damage<!---->
When calling from H323 to a user with multiple registrations<br/>and the called user accepts on one of its (SIP type) secondary registration,<br/>the media negotiation can fail.<!---->
      
      
=== PBX dtmf object selects wrong mobility user for feature codes ===
=== PBX: Progress Indicator in Alert not forwarded by PBX ===


{|
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|}
|}
Problem: e.g. calling cfu activate activates cfu on any user with a mobility fork and not just on the mobility user who is calling<br/><br/>Solution: use correct number to determine mobility user<br/><br/>Files: pbx_dtmf.cpp<br/><br/>Risk: no risk<!---->
This could result in in-band info not played at receiving phone in case no progress incator was sent in previous message of same call<!---->
''Status:''
pbx.cpp
      
      
=== PBX dtmf/icp object couldn't assign e164 without node/pbx if another object already has this e164 ===
=== Call Completion to MD110 didn't work ===


{|
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|}
|}
Problem: setting an e164 for features in an object without node/pbx didn't work if another object already had this number, but with configured node/pbx<br/><br/>Solution: use a new method for determining existing e164<br/><br/>Files: pbx_dtmf.cpp, pbx_icp.cpp, pbx.cpp, pbx.h, pbx_api.h<br/><br/>Risk: no risk <!---->
Call Completion to MD110 didn't work<!---->
      
      
=== Phone: Mis-configuration may cause phone to try presence subscription for nobody ===
=== VM, Smtp authentication sometimes in-place, although not required ===


{|
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|}
|}
Problem: Configuring a partner fkey without specifying partner's name or number causes the phone to subscribe for presence without name or number.<br/><br/>Solution: Never try to subscribe for presence without name or number.<br/> <br/>Files: phonesig.cpp<br/><br/>Products affected: All phones<br/><br/>Risk: No risk.<!---->
VM, Smtp authentication sometimes in-place, although not required<br/><!---->
      
      
=== Ignore calls from gateway objects to dtmf object if no mobility user is found for incoming number ===
=== SIP: Media negotiation issue ===


{|
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|}
|}
Problem: If one would call the dtmf object over a gateway object and no mobility user is found for the caller, the feature codes would be applied to the gateway object.<br/><br/>Solution: Reject calls over gateway objects without mobility user for incoming number.<br/><br/>Files: pbx_dtmf.cpp<br/><br/>Risk: no risk <!---->
Handling of re-INVITE w/o SDP offer in 'held' state requires change.<br/><!---->
      
      
=== Trap of PBX when relasing webmedia call ===
=== PBX: Blind transfer with consultation to mobile endpoint -&gt; Retrieve missing ===


{|
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|[http://mantis.innovaphone.com/view.php?id=62638 62638]
|}
|}
Problem: Trap when releasing a webmedia call (MOH, WQ announcement, Voicemail, etc).<br/><br/>Solution: Don't give events to channel object after CHANNEL_DISCONNECT.<br/> <br/>Files: dummysig.cpp<br/><br/>Products affected: All PBX devices<br/><br/>Risk: No risk. <!---->
The caller is put on hold for the consultation, but is not retrieved when the transfer happens. If the caller is SIP, this results in no media sent.<!---->
''Status:''
pbx.cpp
      
      
=== Potential trap when disconnecting call, WEBMEDIA-CH.5 default(82c09798): serial_event(814) ===
=== Possible trap on certain compact flash operations ===


{|
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|}
|}
problem: Under special timing conditions a trap could happen during call disconnect. This only happened if the call terminated at a physical interface on the given box.<br/><br/>solution: Cleaning up of media channel fixed<br/><br/>files: media.cpp, media.h, medialib.h<br/><br/>products: all<br/><br/>risks: Other problems with media negotiation<!---->
There has been the possibility of a trap on certain compact flash file operations.<br/>This trap has been fixed.<!---->
      
      
=== PBX: Filter needed for Gateway or Trunk objects ===
=== DHCP client: timeout for response to a REQUEST too small in some case ===


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|[http://mantis.innovaphone.com/view.php?id=62709 62709]
|}
|}
problem: Filter configuration was removed from the user interface of Trunk and Gateway objects, but there are applications for which Filters are needed for these objects.<br/><br/>solution: Filter configuration added for Trunk and Gateway<br/><br/>files: pbx_edit_trunk.xsl, pbx_edit_gw.xsl<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
When the DHCP client REQUESTs an OFFERed address a variable timeout (min 2 seconds) is set up. In the case in question the server always responds to DISCOVERs and REQUESTs with a delay of a little bit more than 2 seconds and thus a new DISCOVER was triggered a short time before the ACK arrived.<br/>To overcome this problem the minimum timeout is changed to 5 seconds which should be enough for any server.   <!---->
      
      
=== HTTP client head request waits for data ===
=== ADSP driver: initialization changed ===


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|[http://mantis.innovaphone.com/view.php?id=51978 51978]
|[http://mantis.innovaphone.com/view.php?id=62869 62869]
|}
|}
problem: HTTP client head request waits for data since a content_len is set in the HTTP header. That data is not sent during a head request<br/><br/>solution: ignore content-len header on a head request<br/><br/>files: httpclient_i.cpp <br/><br/>products: all<br/><br/>risks: low risk <!---->
The ADSP2191 initialization is changed. This fixes some missed voice channels in conference calls.<!---->
      
      
=== IP-DECT Master potential trap ===
=== Diagnostic/Tracing on IP6000: Trace flag on TEL could not be cleared ===


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|[http://mantis.innovaphone.com/view.php?id=51983 51983]
|[http://mantis.innovaphone.com/view.php?id=62914 62914]
|}
|}
problem: There is a potential trap if the IP-DECT Master is used in the IP6000.<br/><br/>solution: Condition added.<br/><br/>files: dectmaster.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage.<!---->
once set, it could only be cleared with a !config change command<!---->
''Status:''
tracing.xsl
      
      
=== ISDN: Call completion could not be activated at point-to-point interfaces ===
= V8 Hotfix14 (80500.47) =
Changes included in Version 8 hotfix14
[http://mantis.innovaphone.com/view.php?id=63026 Definition]
 
== New Features ==
 
 
   
=== New flash S29GL256P90/S29GL128P90 on IP1200 ===


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|[http://mantis.innovaphone.com/view.php?id=51990 51990]
|[http://mantis.innovaphone.com/view.php?id=58643 58643]
|}
|}
Problem: Call completion could not be activated at point-to-point interfaces. ccbs_T_Request was badly encoded.<br/><br/>Solution: Fixed encoding ccbs_T_Request and added handling of ccbs_T_RequestResult.<br/><br/>Files: q932asn1.cpp relay.cpp q950.cpp/h fty.h<br/><br/>Products affected: BRI Gateways<br/><br/>Risk: No risk.<!---->
This flash is used on new IP1200 devices.<br/>Bootcode downgrade to older bootcode is disabled.<br/>If the bootcode is downgraded the bootcode version is shown as 1013.<!---->
      
      
=== SIP: UPDATE with SDP with "sendrecv" wasn't handled ===
=== SNMP, innoColdStart Trap to be sent only after sw failure or button reset ===


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|[http://mantis.innovaphone.com/view.php?id=63160 63160]
|}
|}
Problem: UPDATE(sendrecv) wasn't handled after UPDATE(sendonly/inactive).<br/><br/>Solution: Fixed handling of UPDATE with SDP.<br/><br/>Files: sip.cpp<br/><br/>Products affected: All SIP devices<br/><br/>Risk: No risk.<!---->
Settlement of a feature request to have the innoColdStart SNMP trap indicate severe reboot reasons only.<br/><!---->
      
      
=== H323 potential trap during signaling cleanup ===
=== DECT: GUI password input limit info ===


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problem: A trap can occur if a call is still active in accept state and its signaling is cleaned-up.<br/><br/>solution: Call membership fixed.<br/><br/>files: h323sig.cpp.<br/><br/>products affected: All devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
The user password is truncated to 15 signs. Now the input field is limited and an info is shown.<!---->
      
      
=== Phone: Presence fkey disappears and cannot be configured ===
=== support for external ringer unit ===


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|[http://mantis.innovaphone.com/view.php?id=63358 63358]
|}
|}
Problem: An already configured fkey "Presence" disappears and cannot be configured after uploading hotfix2.<br/><br/>Solution: Fixed presence fkey.<br/><br/>Files: phone_config.cpp phone_edit.cpp<br/><br/>Products affected: All phones<br/><br/>Risk: No risk.<!---->
some special purpose phones may be equipped with an external ringer unit. the information controlling the internal ringer is now passed to the module controlling the external ringer unit.<!---->
      
      
=== PBX: Successive diversion activate/deactivate and dynamic group status sometimes failed ===
== Bug Fixes ==
 
 
   
=== H.323: Don't send a call-independent-signaling call without facilities and user-user information ===


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problem: If a diversion activate/deactivate from an endpoint arrived before the last was written to flashdir, update was performed based on information in dram not information being written into flash<br/><br/>solution: Update information in dram before writing to flashdir<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h, pbx_dtmf.cpp<br/><br/>products: all with PBX<br/><br/>risks: Collateral damage with diversion/dynamic group updates<!---->
This fix is related to the fix #59088.<br/>A call-independent-signaling call without facilities should not be sent, but if it has got a user-user information, it should be sent.<br/>This fixes the DECT messaging problem on the IP1200.<!---->
''Status:''
h323sig.cpp
      
      
=== DHCP client must check if an address provided by the server or a kept/reused address  is not already in use ===
=== TCP: Ack was not sent under special conditions with re-transmissions ===


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|[http://mantis.innovaphone.com/view.php?id=62965 62965]
|}
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problem: some DHCP servers may deliver addresses which are already used by another device in the network. This may happen if the server does not check the addresses before delivery and either the server crashed and forgot previous assignments or there is a statically configured device with this address in the network<br/><br/>solution: if there is ARP reply for the provided address send a DHCPDECLINE to the server and restart discovery<br/><br/>files: dhcp.cpp, dhcp.h, arp_p.cpp<br/><br/>products: all<br/><br/>riscs: none <!---->
This could cause the breaking of a TCP connection in case of packet loss, even if the packet loss was not too bad<!---->
''Status:''
''Status:''
checked in to 8.00
tcp.cpp
      
      
=== Control calls without facility elements were forwarded on ISDN ===
=== Trap when processing webdav requests ===


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|[http://mantis.innovaphone.com/view.php?id=62980 62980]
|}
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Problem: Control calls (calls without media channel) without facility elements were forwarded on ISDN. Seems to causes trouble on some ISDN switches<br/><br/>Solution: Reject control calls without facility elements with cause "Invalid information element contents".<br/> <br/>Files: q931.cpp/h nt_tbl.tbl te_tbl.tbl<br/><br/>Products affected: All gateways<br/><br/>Risk: No risk.<!---->
Trap when webdav request session were terminated irregularly.<!---->
      
      
=== H.323 RAS Registration thru NAT to PBX does not work with password ===
=== SIP: Bad encoding of To-URI in INVITE when handling REFER with special chars in user part of Refer-To URI ===


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|[http://mantis.innovaphone.com/view.php?id=63030 63030]
|}
|}
problem: If registration with password is done, the client is sending a GatekeeperRequest as first message. Response to this message is sent to the (private) address contained in the message itself and not to source address<br/><br/>solution: Send response back to source<br/><br/>files: h323ras.cpp, h323.h<br/><br/>products: all<br/><br/>risks: Minimal<!---->
Refer-To: <sip:+49231395710880_(399)@172.20.173.104><br/>received with REFER was mangled into<br/>To: <sip:%2049231395710880_(399)@172.20.173.104><br/>and send in INVITE<!---->
      
      
=== IP-DECT OEM location recovery ===
=== HTTP-Server: Closing connection after transaction causes trouble with Webdav client ===


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|[http://mantis.innovaphone.com/view.php?id=52146 52146]
|[http://mantis.innovaphone.com/view.php?id=63045 63045]
|}
|}
problem: No location cancel acknowledge response message is sent back if the endpoint is unknown. Needed in OEM system.<br/><br/>solution: Location cancel acknowledge response message added.<br/><br/>files: dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. OEM devices only.<!---->
NetDrive client fails when uploading files<!---->
''Status:''
http.cpp
      
      
=== IP-DECT trap during debugging ===
=== Webdav: Bug when handling GET with Range header ===


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|[http://mantis.innovaphone.com/view.php?id=63131 63131]
|}
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problem: Trap occurs if endpoints are listed for debugging in DECT radio.<br/><br/>solution: Pointer check added.<br/><br/>files: dectlocalusers.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage.<!---->
When applied on a zero length file this response was returned:<br/><br/>\tHTTP/1.1 206 Partial Content<br/>\tDate: Tue, 12 Apr 2011 14:52:23 GMT<br/>\tServer: innovaphone Virtual Appliance / 9.00 dvl [xxx/1000/0]<br/>\tAccept-Ranges: bytes<br/>\tContent-Type: application/octet-stream<br/>\tContent-Length: 0<br/>\tContent-Range: bytes 0-4294967295/0<br/><br/>Error response "416 Requested Range Not Satisfiable" must be returned instead.<!---->
      
      
=== SIP: SRTP key changes right after connect ===
=== Webdav: Don't keep zero-length files open on server side ===


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|[http://mantis.innovaphone.com/view.php?id=63133 63133]
|}
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Problem: During call establishment with SRTP a re-INVITE is initiated right after connect providing a new SRTP key. It's unnecessary and some equpiment fail to handle change of SRTP key.<br/><br/>Solution: Avoid change of SRTP during call.<br/><br/>Files: sip.cpp/h<br/><br/>Products affected: SIP Gateways<br/><br/>Risk: No risk.<!---->
In case of large files, NetDrive performes GET operation between PUT0 and PUT.<br/>The actual PUT was rejected with 500 error resonse then.<!---->
      
      
=== SIP: Anonymize remote-party info when sending dialog-info if remote-party calls with CLIR ===
=== 62879: ISDN, QSIG, NT: No Disc Option can be used to send PROGRESS instead of DISC - fix for this fix ===


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|[http://mantis.innovaphone.com/view.php?id=63209 63209]
|}
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Problem: Calling party is seen in dialog-info even if CLIR was set.<br/><br/>Solution: Hide remote party in dialog-info if remote-party calls with CLIR.<br/><br/>Files: sipmsg.cpp<br/><br/>Products affected: PBX serving SIP endpoints<br/><br/>Risk: No risk.<!---->
This fix from hotfix13 did only for calls on which a CALL-PROC was sent as well. For calls still in overlap dialing (only SETUP-ACK sent) it did not work<!---->
''Status:''
nt_tbl.tbl
      
      
=== SIP: Trap when using TCP as transport ===
=== SIP: Fix for dialog-info notification ===


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Problem: Trap when trying to cleanup idle/unused TCP connections.<br/><br/>Solution: Check connection state before initiating connection shutdown.<br/> <br/>Files: siptrans.cpp<br/><br/>Products affected: SIP devices doing SIP over TCP<br/><br/>Risk: No risk.<!---->
NOTIFY for dialog state 'terminated' was missing sometimes.<!---->
      
      
=== Gateway: Potential Trap with collision of transfer an call clearing ===
=== SIP: Trap when session timer is used ===


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|[http://mantis.innovaphone.com/view.php?id=63271 63271]
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problem: If a transfer is attempted to a call to the gateway at the same time this call is release an NULL pointer access could happen<br/><br/>solution: Check for this situation added<br/><br/>files: gk.cpp<br/><br/>products: all with gateway<br/><br/>risks: Nonen<!---->
Trap on collision of session timer and call release<!---->
      
      
=== innovaphone parameters were not sent with H.323 ras registration confirm, if the confirm was lost the first time ===
=== SIP: Authentication passwords were truncated ===


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|[http://mantis.innovaphone.com/view.php?id=63321 63321]
|}
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problem: If a RasRegistrationRequest needed to be retransmitted, because the RegistrationConfirm was lost, the second RegistrationConfirm did not contain innovaphone parameters and the client did not detect it was connected to a innovaphone PBX/Gatekeeper.<br/><br/>solution: Send innovaphone parameters in this case also<br/><br/>files: h323.h, h323ras.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
Authentication failed because password was truncated.<!---->
      
      
=== SIP Entity URI in "application/dialog-info+xml" and "application/pidf+xml" was wrong ===
=== SIP: Not accepting calls from alternative proxy ===


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|[http://mantis.innovaphone.com/view.php?id=63327 63327]
|}
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Problem: The SIP URI in the "entity" attribute was wrong in presence and dialog XML.<br/><br/>Solution: Fix SIP URI in the "entity" attribute.<br/> <br/>Files: sip.cpp/h sip_dialog_info.cpp/h<br/><br/>Products affected: PBXs serving SIP endpoints<br/><br/>Risk: No risk.<!---->
When being registered at a proxy with 2 ip addresses the gateway does not accept calls from the alternative ip address.<!---->
   
= V8 Hotfix15 (80500.49) =
Changes included in Version 8 hotfix15
[http://mantis.innovaphone.com/view.php?id=63485 Definition]
 
== New Features ==
 
 
      
      
=== DSP fix sporadic trap  ===
=== DECT: Radio firmware for new handsets ===


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problem: sporadic trap in ac-dsp, defect return address<br/><br/>solution: fax buffer size increased, buffer check added, dsp receive packet relase done later <br/><br/>files: ac_fax2.cpp ac_dsp2.cpp<br/><br/>products: ip800 ip6000 ip1200 ip1201 ip4001<br/><br/>risks: low risk <!---->
The new radio firmware PCS05Ah accepts new handsets with the new IPEI number range.<!---->
      
      
=== SIP: Bad REGISTER request was not rejected ===
=== phone: improved czech display texts ===


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|}
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Problem: If an incoming REGISTER request does not contain userpart in To-URI, no response was sent to client.<br/><br/>Solution: Reject with "400 Bad Request".<br/> <br/>Files: sip.cpp sipmsg.h<br/><br/>Products affected: PBXs serving SIP endpoints<br/><br/>Risk: No risk.<!---->
now all texts are translated to czech, previous errors were fixed (translations provided by zakharova@annexnet.cz)<!---->
      
      
=== Potential hanging h323 signaling (Mem Leak) on collision of removing a signaling entity with outg. call ===
== Bug Fixes ==
 
 
   
=== PBX: License mechanism changed to allow easy migration to new version ===


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problem: If a signaling entity is being removed (e.g. by configuration change) and at the same time an outgoing call is attempted at this interface, it could happen that the remove of the signaling interface failed.<br/><br/>solution: handle this collision<br/><br/>files: h323sig.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
- licences of different versions may be installed<br/>- check for min version<br/>- v8 master can act as license master for v9 licenses<br/>- applications may run on older version<!---->
''Status:''
inno_lic.cpp<br/>inno_lic.h<br/>pbx.cpp<br/>pbx_api.h<br/>pbx_general.xsl<br/>pbx_edit_loc.xsl<br/>
      
      
=== Avoid reboot when reading traces, if trap happens after firmware update ===
=== PBX: Trunk - don't retry call to next gateway if wrong number ===


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|[http://mantis.innovaphone.com/view.php?id=63386 63386]
|}
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problem: If a firmware update is done and before a regular restart a trap happens, the next reading of the trace buffer could generate another trap, because firmware dependent content is accessed in trace buffer.<br/><br/>solution: Avoid trap, by reading save info only from trace buffer in this case<br/><br/>files: debug.cpp, debug.h, arm.cpp, mips.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
all gateways registered to a trunk are by definition to the same network, so a rerouting is useless, if the cause indicates that the dialed number was wrong<!---->
''Status:''
q931lib.h
      
      
=== PBX: Potential Trap if changing groups ===
=== Command traps in minifirmware on joining or leaving Kerberos realms ===


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|}
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problem: If group memberships are changed esspecially at Waiting Queues or Waiting Queue operators a trap could happen, because PBX internal information could get inconsistent.<br/><br/>solution: Fixed update of internal information<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: Other traps, complex operations, so it is possibly still wrong<!---->
Because command does not check if kerberos_client_provider::provider is null.<br/><br/>Files: command.cpp<!---->
      
      
=== socket bind/connect sometimes failed because of  duplicate assignment of local wildcard port ===
=== TEL and PRI1-4 not contained in 'PPP connection port' dropdown menu on ip6010, ip3010 and ip1060 ===


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|[http://mantis.innovaphone.com/view.php?id=63419 63419]
|}
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problem: a local wildcard port could be assigned twice in case a socket using this port did exist over a period where all port numbers above and below this number had been assigned once.<br/><br/>solution: fix in code<br/><br/>files: tcp.cpp, tcp.h<br/><br/>products: all<br/><br/>riscs: none<br/><!---->
'PPP connection port' dropdown should contain TEL and PRI1-4<!---->
''Status:''
''Status:''
checked in to 9.00,8.00
ip_config.cpp
      
      
=== IP-DECT call counter ===
=== ip0010 wizard configures PRI1, gateway/interfaces shows PRI1 ===


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|[http://mantis.innovaphone.com/view.php?id=63430 63430]
|}
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problem: The call counter for maximum cpu load for SRTP should count radio calls and handover-ins together.<br/><br/>solution: Call counter changed. Handover-in calls aren't counted separately.<br/><br/>files: dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
PRI1-L1 must be renamed into PRI1-CLK<!---->
''Status:''
config.h, ip6010.cpp
      
      
=== SIP: Trap when performing call transfer on ARM based hardware ===
=== HTTP-Client: Bad encoding of uri parameter in digest authentication ===


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|}
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Problem: Trap due to alignment error.<br/><br/>Solution: Fix alignment of data.<br/> <br/>Files: sip.cpp<br/><br/>Products affected: ARM based devices talking SIP<br/><br/>Risk: No risk.<!---->
Uri parameter in digest authentication was not URL encoded<!---->
      
      
=== IP-DECT call counter busy state ===
=== Gateway: Outgoing Call Completion did not work when outgoing call was routed through TONE interface ===


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problem: The last possible call from master is not accepted by DECT because the DECT is switched to busy state before sending this call setup.<br/><br/>solution: Sequence of signaling setup and busy message changed.<br/><br/>files: dectradio.h, dectradio.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
Outgoing CC request did not went out to ISDN interface.<!---->
      
      
=== Kerberos trap when turning off standby PBX with replication ===
=== SIP: Message buffer too small for REGISTER request for re-try with authentication ===


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problem: The box traps when a standby PBX with LDAP replication is turned off. <br/><br/>solution: Fix event flow in kerberos_ldap_realm_tree.<br/><br/>products: all with PBX<br/><br/>files: kerberos_ldap.cpp, pbx.cpp<br/><br/>risk: no risk known<!---->
On some installations a change-of-nonce at server side may cause volatile "Registration down error" on client side.<!---->
      
      
=== PBX: Unknown Registrations did not display name _UNKNOWN_ on the phone anymore ===
=== certain non latin-1 characters entered via WEB interface or provided by an external LDAP Server cause display errors ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=52417 52417]
|[http://mantis.innovaphone.com/view.php?id=63591 63591]
|}
|}
problem: By accident the name _UNKNOWN_ was removed for unknown registration<br/><br/>solution: Name _UNKNOWN_ added<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
entering such characters via copy/paste as when editing a PBX object may result in an xml-error when showing PBX objects.<br/>when such characters are provided by an external LDAP Server to a phone the display may get cleared.<br/>Now such characters are transcribed to a single latin1 character or replaced by a '-' if no transscription is available. <!---->
      
      
=== IP-DECT OEM protocol memory leaks ===
=== Web-UI: PBX password length is limited to 15 chars ===


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|[http://mantis.innovaphone.com/view.php?id=52466 52466]
|[http://mantis.innovaphone.com/view.php?id=63640 63640]
|}
|}
problem: There are memory leaks if Skinny protocol is used.<br/><br/>solution: Cleanup fixed.<br/><br/>files: skinny.h, skinny.cpp, skinny_signaling.cpp, skinny_translation.cpp.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
Added tooltip and fixed maxlength attribute on input elements.<!---->
      
      
=== IP-DECT IP6000  DECT module ===
=== License: Character encoding problem ===


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|[http://mantis.innovaphone.com/view.php?id=52486 52486]
|[http://mantis.innovaphone.com/view.php?id=63645 63645]
|}
|}
problem: Unused OEM DECT module should not be available in IP6000.<br/><br/>solution: Configuration changed.<br/><br/>files: config.h, changed OEM files, removed files.<br/><br/>products affected: All DECT devices.<br/><br/>risk: Minimal risk of collateral damage. <!---->
Character encoding problem<!---->
      
      
=== PBX SOAP FindUser did not work correctly with users not in root node ===
=== config download may trap when malformed LDAP config data  has been uploaded ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=52498 52498]
|[http://mantis.innovaphone.com/view.php?id=63678 63678]
|}
|}
problem: A library function adding prefixes to a number did not work correctly. This was used within the FindUser SOAP function of the PBX<br/><br/>solution: library function fixed<br/><br/>files: q931lib.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
a buffer overrun happens on config download when a "mod cmd FLASHDIR0 add-view nnn cn=..." line with a length > 63 characters has been uploaded. <!---->
      
      
=== H.323 Signal IE when sent once during a call, was then sent with each subsequent message ===
=== Presence functionality is not available when registered via H323 at a non-innovaphone PBX ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=52517 52517]
|[http://mantis.innovaphone.com/view.php?id=63745 63745]
|}
|}
problem: A Signal IE sent with one message was repeated with all the messages. This could cause the a Signal IE indicating Ringback sent with Alert was repeated with Disc, so that no busy tone was played with Disc, but Ringback<br/><br/>solution: Send Signal IE once only<br/><br/>files: h323sig.cpp<br/><br/>products: all<br/><br/>risks: Minimal risk of collateral damage<!---->
Presence operations via H323 are encoded in private facility elements which are unknown to a non-innovaphone PBX. Presence control calls sent to such a PBX may be misunderstood and routed back as normal voice call to the sending phone.<br/>Thus no presence control calls must be sent to such a PBX.<!---->
      
      
=== IP-DECT BMC trace off command ===
=== Trap when starting from flash_stick ===


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|[http://mantis.innovaphone.com/view.php?id=63752 63752]
|}
|}
problem: The BMC should be able to stop the trace for debugging.<br/><br/>solution: BMC message and handler added.<br/><br/>files: dect.h, dect.cpp.<br/><br/>products affected: All OEM DECT devices.<br/><br/>risk: No risk of collateral damage. <!---->
and flash memory not yet programmed with bootcode<!---->
''Status:''
ip6010.cpp
      
      
=== Problems with TLS event handling ===
=== SIP: Allocated message size to small for INVITE redirect response (Avaya) ===


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|[http://mantis.innovaphone.com/view.php?id=52537 52537]
|[http://mantis.innovaphone.com/view.php?id=63829 63829]
|}
|}
problem: Two problems with the TLS state machine were found: <br/>A) SOCKET_SHUTDOWN from application is ignored if it is received between SOCKET_CONNECT from application and SOCKET_CONNECT_COMPLETE from TCP.<br/>b) When the TCP connection is closed during TLS handshake, TLS tries to send a SOCKET_RECV_RESULT(fin) to the application instead of SOCKET_SHUTDOWN.<br/><br/>solution: fix state machine<br/><br/>products: all<br/><br/>files: tls.cpp, tls.h<br/><br/>risk: risk of damaging applications using TLS<!---->
Memory allocation is a bit to tight to fit the message due to many Via headers.<br/><br/>INVITE sip:3003@192.168.150.140:2059;transport=UDP SIP/2.0<br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=udp;lr><br/>Record-Route: <sip:192.168.150.114:15060;lr;sap=315810451*1*016asm-callprocessing.sar1905633216~1304428214402~-1054885358~1><br/>Via: SIP/2.0/UDP 192.168.150.115;rport;branch=z9hG4bKC0A896726E7526620194612-AP;ft=192.168.150.115~13c4<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526620194612<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194610<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194609<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bK0e2106b7388e016424db9a29200-AP;ft=11786<br/>Via: SIP/2.0/TCP 192.168.150.118;branch=z9hG4bK0e2106b7388e016424db9a29200;avaya-cm-term-reaction=shortcut<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bKC0A896726E7526620194608-AP;ft=12651<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526620194608<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194606<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194605<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bK0e2106b7388e018424db9a29200-AP;ft=11786<br/>Via: SIP/2.0/TCP 192.168.150.118;branch=z9hG4bK0e2106b7388e018424db9a29200<br/>Via: SIP/2.0/TCP 192.168.150.84;branch=z9hG4bK200_f1774512c29cc2e5cd78966_I2371<br/>User-Agent: Avaya one-X Deskphone AVAYA-SM-6.1.1.0.611023 Avaya CM/R016x.00.1.510.1<br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=tcp;lr><br/>Record-Route: <sip:192.168.150.114:15060;transport=tcp;lr;sap=315810451*1*016asm-callprocessing.sar1905633216~1304428214355~-1054885362~1><br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=tcp;lr><br/>Record-Route: <sip:192.168.150.118;transport=tcp;lr><br/>Session-Expires: 1800;refresher=uac<br/>Content-Type: application/sdp<br/>Content-Length: 215<br/>...<!---->
      
      
=== local call forwarding on busy for already alerting calls ===
=== IP152: Flash access not working with version 8050047 ===


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|[http://mantis.innovaphone.com/view.php?id=52541 52541]
|[http://mantis.innovaphone.com/view.php?id=64009 64009]
|}
|}
problem: local call forwarding on busy was supported only for calls rejected with cause busy before entering alerting state, i.e. when call waiting was disabled and the phone was busy with another call. there seems to be a need to forward already alerting calls too when the disconnect button is  pressed to get rid of the call. <br/><br/>solution: implement in code<br/><br/>files: phonesig.cpp<br/><br/>products: all phones<br/><br/>riscs: none, only used when local call forwarding is enabled<!---->
With fix #58643 16 bit access to spansion flash doesnt work<!---->
''Status:''
''Status:''
checked in to 9.00,8.00
boot_coldfire.mak common.mak flash_coldfire.c
      
      
=== phone: DHCP configuration of a non-automatic primary registration fails when the registration is created before DHCP completion ===
=== No received cause code should be treated as 'normal clearing' ===


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|[http://mantis.innovaphone.com/view.php?id=64043 64043]
|}
|}
problem: when DHCP completed after creation of a non-automatic primary registration some parameters provided in phonesig_if::create_phone_reg() were overriden (non-automatic primary registration: a primary registration not created automatically by phonesig.cpp with the parameters given on the "config change PHONE SIG ..." command line but by an application).  <br/><br/>solution: fix in code<br/><br/>files: phonesig.cpp<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
Was sometimes treated as cause code to do re-routing. This happened esspecially with multiple registrations to v8 gateway object. A call sent successfully to the gateway on the first regsitration was sent again on the second registration after call clearing.<!---->
''Status:''
''Status:''
checked in to 9.00,8.00
q931lib.cpp<br/>relay.cpp
      
      
=== Call Intrusion across PBXs did not work ===
=== missing response 'reset required' when changing PRIx-Lx config options ===


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|[http://mantis.innovaphone.com/view.php?id=64055 64055]
|}
|}
problem: If a call was to be intruded with the destination of the intrusion (which is one of the endpoints of the call to be intruded call) on a different PBX the intrusion failed in a strange way, because the intrusion facilities were not correctly forwarded between the PBXs<br/><br/>solution: Fix forwarding of intrusion facilities<br/><br/>files: pbx.cpp, pbx.h, pbx_api.h<br/><br/>products: all with PBX<br/><br/>risks: Small risk of collateral damage with intrusion<!---->
changing i.e. the ,NT-Mode' config option didn't show the 'reset required' link button after pressing 'OK'.<!---->
''Status:''
falc56_drv.cpp, config.h ipac_drv.cpp V9:falc56_drv.xsl
      
      
=== PBX Trunk automatic disconnect did not work if user was monitored by SOAP ===
=== PBX: Transfer Recall timer was not started if destination was ringing after blind transfer ===


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|[http://mantis.innovaphone.com/view.php?id=64064 64064]
|}
|}
problem: If a user was monitored by SOAP a automatic hangup was not sent to the endpoint<br/><br/>solution: Send automatic hangup<br/><br/>files: pbx.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
After a blind transfer without consultation to a busy destination the recall timer should be started as soon as the destination is not busy anymore and the call is delivered<!---->
''Status:''
pbx.cpp
      
      
=== Potential trap if HTTP session is closed while command is still pending ===
=== Gateway: Allow interface maps for analog interfaces as well ===


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|[http://mantis.innovaphone.com/view.php?id=64068 64068]
|}
|}
problem: The command processor was deleted in this case without waiting for result of the command<br/><br/>solution: Wait for result until deleting command processor<br/><br/>files: command.cpp, command.h<br/><br/>products: all<br/><br/>risks: Hanging command processors if check wrong<!---->
Was prohibited in the past, but there are uses for this.<!---->
''Status:''
ip24/config.h
      
      
=== IP-DECT/Analog: CC was lost if initiating endpoint busy ===
=== Conference on IP6000 Hardware 200 and lower not working with v8hf14 and v9 ===


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|[http://mantis.innovaphone.com/view.php?id=64132 64132]
|}
|}
problem: If with an anlog phone connected to a IP24/28/... gateway or an IP-DECT phone initiated a call-completion but was busy itself at the time the call-completion could be executed, the call-completion was silently discarded<br/><br/>solution: delay call-completion until not busy<br/><br/>files: dtmffty.cpp, dtmffty.h<br/><br/>products: DECT and Analog gateways<br/><br/>risks: Minimal risk of collateral damage<!---->
The ADSP serial port has been changed from SPORT1 to SPORT0 for the IP6010.<br/>Old IP6000 hardware has the SPORT0 not connected, so now SPORT1 is again used on IP6000.<br/><!---->
      
      
=== Switch-PCM together with Media-Relay caused disconnect of call ===
=== PBX: Potential Trap on calls to exec, map or waiting object ===


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|[http://mantis.innovaphone.com/view.php?id=64135 64135]
|}
|}
problem: If a call from one ISDN interface to another ISDN interface on the same box with Switch-PCM enabled (so that normaly the PCM Switch should be used), was routed thru a PBX with Media-Relay, the call was disconnected.<br/><br/>solution: In case of Media-Relay the PCM-Switch should not be used<br/><br/>files: h323ch.cpp<br/><br/>products: all<br/><br/>risks: None<!---->
under some rare circimstances, which are unfortunatly not known, there could be a NULL pointer access<!---->
''Status:''
pbx_exec.cpp<br/>pbx_wait.cpp<br/>pbx_map.cpp
   
= V8 Hotfix17 (09-80500.55) =
Changes included in Version 8 hotfix17
[http://mantis.innovaphone.com/view.php?id=65485 Definition]
 
== New Features ==
 
 
      
      
=== Registration with Name or Number was only possible if Device with hw-id identical to name was configured ===
=== QSIG: Avaya expect Progress Indicator with external calls ===


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|}
|}
problem: In v8 devices were introduced into PBX configuration, which allow much better control of SOAP applications on which devices are used and also a mechanism to prohibt registration by Name or Number was built in. Even if this is useful in some cases, this mechanism was so obscur that it generated problems.<br/><br/>solution: Allow registrtation with name or number again independent of configured devices. If no device is configured with hw-id matching the name of the object, such a registration is associated to the first configured device.<br/><br/>files: pbx.cpp, pbx_admin.cpp<br/><br/>products: all with PBX<br/><br/>risks: In some installation the new feature may have been used already.<!---->
Avaya uses the Progress indicator 'Interworking with a public network' to identify a call as external. This Progress Indicator is now added for calls from a Number NOT with private numbering plan (which is our way to identify internal calls)<!---->
''Status:''
q931.cpp
      
      
=== phone: names containing non-ascii characters entered in the registration menu of secondary registrations are garbled ===
=== ISDN: New interop flag to forward network provided or checked cli only ===


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|[http://mantis.innovaphone.com/view.php?id=66183 66183]
|}
|}
problem: if any name containing non-ascii characters was entered in the registration menu of a secondary registration (Registration 2...6/Registration) the text was stored in wrong encoding. in this case it was not not possible to register via 'Name' or to access a gatekeeper via 'Gatekeeper Identifier'.<br/><br/>solution: fix in code<br/><br/>files: phonesig_if.cpp, phonesig_if.h, phone_edit.cpp, phonesig.cpp, phonesig.h, app_fkey.cpp<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
Useful if the real calling number is needed and not a number provided by CLIP no screening<!---->
''Status:''
''Status:''
checked in to 9.00,8.00
q931.cpp<br/>q931.h<br/>isdn_interop.xsl
      
      
=== Noise after transfering a waiting queue connection ===
== Bug Fixes ==
 
 
   
=== SIP: Session refresh was taken as session modification ===


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|}
|}
Problem: Being connected to a waiting queue announcement; Transfering this call to another endpoint; Transfer destination will hear noise instead of waiting queue announcement (in case of code change only)<br/><br/>Solution: Re-start announcement in matching coder.<br/> <br/>Files: webmedia.cpp/h<br/><br/>Products affected: All PBXs<br/><br/>Risk: No risk.<!---->
Local SRTP key was re-calculated after re-INVITE for session refreh was received.<br/>Causes SRTP decode error at remote side.<br/>CUCM scenario<!---->
      
      
=== PBX Mobility: Unexpected restart if 3 or more mobility destinations configured at a user ===
=== IP6010, IP6000: Use optimized memcpy ===


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|[http://mantis.innovaphone.com/view.php?id=64587 64587]
|}
|}
problem: The binary tree used to keep track of the mobility calls at a user got corrupted because of a typo in the Mobility object code. For some strange reasons this only happened with 3 ot more mobile destinations<br/><br/>solution: typo corrected<br/><br/>files: pbx_mobility.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
Use of load/store multiple and shifts for 32 bit alignment speeds up memcpy by a factor of approx 2<br/><br/>Orginal memcpy<br/><info product="IP6010" mips="800Mips"><br/><memcpy bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><read bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><write bytes="1000000" time="2ms" speed="470.588Mbyte/s"/><br/><stack_memcpy bytes="1000000" time="7ms" speed="133.333Mbyte/s"/><br/><uncached_memcpy bytes="1000000" time="41ms" speed="24.169Mbyte/s"/><br/><aes bytes="1000000" time="135ms" speed="7.373Mbyte/s"/><br/><sha bytes="1000000" time="70ms" speed="14.260Mbyte/s"/><br/></info><br/> <br/>Optimized memcpy:<br/><info product="IP6010" mips="800Mips"><br/><memcpy bytes="1000000" time="1ms" speed="888.888Mbyte/s"/><br/><read bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><write bytes="1000000" time="2ms" speed="421.052Mbyte/s"/><br/><stack_memcpy bytes="1000000" time="7ms" speed="142.857Mbyte/s"/><br/><uncached_memcpy bytes="1000000" time="15ms" speed="64.000Mbyte/s"/><br/><aes bytes="1000000" time="138ms" speed="7.200Mbyte/s"/><br/><sha bytes="1000000" time="70ms" speed="14.285Mbyte/s"/><br/></info><br/><br/>CPU load with the test test/9.00/box/dsp/ip6010 shows approx 1% lower CPU load.<br/>Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate, compared to 9708Kbyte/s with the old memcpy.<br/><br/>With ECC enabled the CPU load was 19% / 21% without SRTP and 31% / 33% with SRTP<br/>With ECC Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate10309<!---->
''Status:''
ip6010.mak ip6000.mak arm.mak box/arm/memcpy.S<br/><br/>v8: ip6010.mak, box/box.mak, box/memcpy.S
      
      
=== PBX Mobility: No media if 3 or more forking/mobility destinations without delay configured ===
=== Incorrect rpcap timestamp after TRACE LOST messages ===


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|[http://mantis.innovaphone.com/view.php?id=64915 64915]
|}
|}
problem: The media path was not switched correctly when calling mobile endpoints. More or less by accident it worked anyway for the first and the last forking destination<br/><br/>solution: media path switrching fixed<br/><br/>files: pbx_mobility.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
The RPCAP timestamp (Wireshark) after a TRACE LOST message was incorrect, as the TRACE LOST message contained an incorrect timestamp.<!---->
      
      
=== phone: passwords containing non-ascii characters did not work for the primary registration ===
=== VM, Project script didn't run for endpoints having "Send Number" configured ===


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|}
|}
problem: if a password containing non-ascii characters was entered in the registration menu of a primary registration (Registration 1) the password was stored in wrong encoding and thus did not match the password configured at PBX. <br/><br/>solution: fix in code<br/><br/>files: phonesig_if.cpp, phonesig_if.h, phone_edit.cpp, phonesig.cpp, phonesig.h, app_fkey.cpp<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
VM, Project script didn't run for endpoints having "Send Number" configured<!---->
''Status:''
checked in to 9.00,8.00
      
      
=== PBX Mobility: Trap if consultation call is cleared by remote side and user attempts to switch back to first call ===
=== Kerberos: Do not allow registration of multiple databases for one realm name ===


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|[http://mantis.innovaphone.com/view.php?id=65589 65589]
|}
|}
problem: If a consultation call is cleared by remote side an attempt to switch back to the first call (by sending R-Key pattern) leads to a trap<br/><br/>solution: Handle switch back to first call better<br/><br/>files: pbx_mobility.cpp<br/><br/>products: all with PBX<br/><br/>risks: None<!---->
This happened when a box hosted multiple PBXes with the same system name.<br/><br/>files: <br/>kerberos_if.cpp<br/>kerberos_kdc.h (v9 only)<br/>kerberos_kdc.cpp<br/>kerberos_db.cpp<!---->
      
      
=== directory entries displayed duplicate when using delayed input with slow LDAP-servers ===
=== DECT: Trap during registration up handling ===


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|[http://mantis.innovaphone.com/view.php?id=52877 52877]
|[http://mantis.innovaphone.com/view.php?id=65698 65698]
|}
|}
problem: with delayed keyboard input the number of queries sent to the LDAP server when typing a name is reduced. The first query is sent after the number of characters configured by 'delay-count' or the timeout configured by 'delay-ticks'. If another character was typed before the response arrived a new search was started but the previous search results were not cleared.<br/><br/>solution: fix in code<br/><br/>files: phone_dir_ui.cpp<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
Trap in DECT Master fixed. It occurs if the master endpoint is in delete state and a RAS registration up event is received.<!---->
''Status:''
checked in to 7.00,8.00,9.00
      
      
=== Option added to configure DTMF detection sensitivy ===
=== MWI does not work in various Node/Pbx combination ===


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|[http://mantis.innovaphone.com/view.php?id=52879 52879]
|[http://mantis.innovaphone.com/view.php?id=65750 65750]
|}
|}
problem: sporadic false DTMF detect<br/><br/>solution: Option added to configure DTMF detection sensitivy. Use <br/> config change AC-DSP0 /dtmf-threshold <val><br/> config write<br/> reset <br/>to change the sensitivity. <br/>0 selects -38dBm ( default), <br/>1 selects -28dBm, <br/>2 selects -33dBm,<br/>3 selects -43dBm, <br/>4 selects -48dBm<br/>During boot a non-default sensitivity is shown in the trace<br/><br/>files: ac_dsp2.cpp/h ac48xlo.c ac48xdef.h <br/><br/>products: ip800 ip6000<br/><br/><!---->
MWI does not work in various Node/Pbx combination<!---->
      
      
=== report result of call triggered by a Dial function key with  "Send as Control Call\t" checked ===
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===


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|[http://mantis.innovaphone.com/view.php?id=65798 65798]
|}
|}
problem: feedback for the keystroke was missing<br/><br/>solution: report result via a popup message on display<br/><br/>files: phonesig_if.h, phonesig.cpp, phonesig.h, app_reg.cpp, app_fkey.cpp, app_ctl.h<br/><br/>products: all phones<br/><br/>riscs: none<br/><!---->
Occurred on IPBL[4.1.22]<!---->
''Status:''
checked in to 7.00,8.00,9.00
      
      
=== Replication truncated attributes greater 1024 bytes ===
=== SIP: Fix for SDP answer to SDP offer with "a:inactive" ===


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|}
|}
Replication truncated attributes greater 1024 bytes. <br/>asn.1 encoding fixed.<br/>All PBX devices were affected.<br/><!---->
Interop with CUCM.<br/>Should return RTP/AVP(inactive) if offer was RTP/AVP(inactive).<br/>Not not RTP/SAVP(inactive).<!---->
      
      
=== RSA encryption problems with some compilers ===
=== Message Waiting Interrogation: Result message coding wrong ===


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|}
|}
The code was based on invalid assumtions about the evaluation order of function parameters. Hence the code did not work with some compilers.<br/><br/>Files: rsa.cpp<!---->
a malformed message was displayed in wireshark<!---->
''Status:''
h450.cpp<br/>h450asn1.h
      
      
=== Transferring VM calls could trap ===
=== SIP: Set CLIR if display string of From-URI contains "Anonymous" ===


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|}
|}
problem: Transferring VM calls could trap<br/><br/>solution: Add NULL pointer access check<br/><br/>files: pbx_vm.cpp<br/><br/>products: all with PBX<br/><br/>risks: none<br/><!---->
Not only if userpart of From-URI contains "anonymous".<!---->
      
      
=== Trap of PBX when relasing webmedia call ===
=== ip6010 - same MAC address was assigned to ETH0 and ETH1 ===


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|}
|}
Problem: Trap when releasing a webmedia call (MOH, WQ announcement, Voicemail, etc).<br/><br/>Solution: Wait for CHANNEL_CONTROL_ACK from channel before releasing call object.<br/><br/>Files: dummysig.cpp/h<br/><br/>Products affected: All gateways<br/><br/>Risk: Small risk of collateral damage.<!---->
this results in problems when both interfaces are connected to the same LAN segment <!---->
      
      
=== Reject TLS sessions if the server uses an expired certificate ===
=== PBX-SOAP: Don't provide caller number if CLIR was used on call to monitored endpoint ===


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|}
|}
TLS connections were not closed on the client side if the server used an expired certificate. This was treated as a warning but not as a fatal error.<br/><br/>Files: tls.cpp<!---->
If this was an internal call, the PBX knows the calling number anyway, but it should not be sent on SOAP<!---->
''Status:''
pbx_xml.cpp
      
      
=== unset "don't fragment bit" in dummy ip header for rpcap ===
=== PBX-SOAP: UserDTMF did not send DTMF to Voicemail or Waiting Objects ===


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|}
|}
The "Don't fragment bit" of the dummy IP header for rpcap was set, which can cause some confusion, if one does not know, that this header is just a dummy header when reading the innovaphone wireshark trace.<br/>This flag is now off.<!---->
It only sent DTMFs to a VOIP connection<!---->
''Status:''
''Status:''
debug.cpp
pbx_xml.cpp
      
      
=== PBX Waiting: 'CFU disables operator' was not taken into account for 'Max Call/Operator(%)' ===
=== Gateway SIP Interfaces: Could not configure internal registration for a disabled interface ===


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|}
|}
If the CFU disables operator option is used, an operator with a configured CFU should be treated as if it was not in the operator group. It was still counted as operator for the Max Call/Operator feature.<!---->
and if a interface was disabled afterwards, the config for the internal registration was lost<!---->
''Status:''
''Status:''
pbx_wait.cpp<br/><br/>v7 merge fehlt
gk.cpp
      
      
=== Call Completion to Gateway object with prefix option failed ===
=== SIP: Trap when receicing provisional response with RSeq header ===


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|[http://mantis.innovaphone.com/view.php?id=65986 65986]
|}
|}
The destination number of the call completion needs to be adjusted: The number of the Gateway object removed.<!---->
Trap when trying to send PRACK<!---->
''Status:''
pbx_gw.cpp
      
      
=== Send Number/URL not configurable at Waiting and Broadcast ===
=== ip6010 - frame loss on ethernet ports running in a VLAN ===


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|}
|}
These parameters are needed for some applications<!---->
receiving of VLAN tagged frames did not work stable, when running ping -t over a longer time a frame loss from 5 to 10 percent was reported <!---->
''Status:''
pbx_wait.h, pbx_api.h, pbx.cpp, pbx_bc.h, pbx_edit_object.xsl, pbx_admin.cpp
      
      
=== DECT System Object could not be configured as critical ===
=== PBX Broadcast: CFNR was executed only after No Response Timeout even if no member ===


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|[http://mantis.innovaphone.com/view.php?id=66032 66032]
|}
|}
This could be definitly a critical object since operation of a DECT system could depend on it<!---->
If there is no member in the broadcast group, a CFNR configured at the Broadcast object should be executet immediatelly.<br/><br/>This was a collateral damage from hotfix<br/><br/>65261: PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions" <!---->
''Status:''
''Status:''
pbx_edit_dect.xsl
pbx_bc.cpp
      
      
=== PBX: Clear Slave license if Slave deregisters ===
=== IP3010/6010: fax problems ===


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|[http://mantis.innovaphone.com/view.php?id=66110 66110]
|}
|}
If a Slave was deregistered the licenses consumed by the slave were not freed on the master. The number of licenses used on the slave was only corrected when the slave registered again or when the PBX object was deleted or on reboot of the master.<!---->
* CED is not transfered <br/>* Wrong T38 encoding in V8 <!---->
''Status:''
''Status:''
pbx.cpp, pbx.h
ac_dsp3.cpp ( AC491 doesnt want the V21/V22... relay bits set )<br/>config.h ( config.h, X missing, on V9 this parameter is not needed )
      
      
=== H.323 RAS registration more robust ===
=== PBX: Missing Group Indications when SIP phone is monitoring ===


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|}
|}
On wireless or congested networks registrations were lost easily if only a few packets were lost. Esspecially if a call signal failed the registration was assumed lost right away. Now this triggers only a keep-alive cycle.<!---->
If a SIP phone is monitored by another SIP phone,<br/>there are GI's missing if the monitored SIP phone is calling.<!---->
''Status:''
h323.h, h323ras.cpp, h323sig.cpp<br/><br/>v7 merge fehlt
      
      
=== Minor User Interface implementation change ===
=== DECT: Delete duplicate LDAP 'pbx' &lt;gw&gt; items ===


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|}
|}
Needed to support special OEM Features. (Use CMD0 for xml-modes exclusively and not sometimes CPU)<!---->
Now duplicate LDAP 'pbx' <gw> items are deleted by the DECT users module.<!---->
''Status:''
box.mak, administrator.htm, left.xsl, up_dram.xsl, update_hdr.xml, xml_modes.xml
      
      
=== H.323 coding fixed, Wireshark indicated error ===
=== PBX Trunk: Prefix was added to connected number even if no connected number present ===


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|}
|}
The mandatory field 'maintainConnection' was missing in the Setup message. This usually does not create problems, because it is an extension, so that the message can be decoded even if the field is missing.<br/><br/>NULL element 'symmetricOperationRequired' was coded with length 0. According to the standard it should be a byte with all 0 bits and length 1.<!---->
The PBX then displayed just the Trunk prefix as remote number on the calls page when the call was connected.<!---->
''Status:''
''Status:''
h323sig.cpp, asn1_per.cpp
pbx_trunk.cpp
      
      
=== Mobility together with PCM calls did not work ===
=== PBX-SOAP: FindUser should not show hidden objects ===


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|}
|}
For calls from an ISDN interface with 'Enable PCM' set, which were sent out again on the same or another ISDN interface on the same box, also with 'Enable PCM' set, sending of additional dialing digist did not work as soon as the PCM connection was switched. For calls thru mobility PCM switching must be disabled.<!---->
Could be confusing<!---->
''Status:''
''Status:''
h323ch.cpp
pbx_xml.cpp
      
      
=== PBX-SOAP: UserClear could not be used to cancel UserCall ===
=== IP6010-CF: Kingston compact flash was not recognized ===


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|[http://mantis.innovaphone.com/view.php?id=66269 66269]
|}
|}
If the local phone was still ringing because of a UserCall, the UserClear could not be used to cancel this call.<!---->
the card was not recognized because a register was wrongly initialized.<!---->
''Status:''
pbx_xml.cpp
      
      
=== log records to a SYSLOG (UDP) server were sent delayed  ===
=== SIP: Bug in SDP handling ===


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|}
|}
log records passed to the logging module before the transport layer is up are saved in a queue. in case of SYSLOG (UDP) this queue was not flushed correctly and sending of a record was triggered only by the following record with the effect that one or more of the latest records were always pending in queue.<!---->
If value of the session id and version in the o line are zero.<!---->
      
      
=== dyn. PBX: Replication started only after reset ===
=== phone: Hexadecimal values instead of descriptive texts were displayed for some rare disconnect causes ===


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|}
|}
After a dynamic PBX with replication was configured a restart was needed to get the replication going.<!---->
"0x57 - unknow cause" was displayed instead of "user not a CUG member". Mainly german descriptive texts were missing.<!---->
''Status:''
   
pbx.cpp, pbx.h, pbx_admin.cpp
= V8 Hotfix18 (80500.57) =
Changes included in Version 8 hotfix18
[http://mantis.innovaphone.com/view.php?id=66417 Definition]
 
== New Features ==
 
 
      
      
=== PBX Mobility: Calls from mobile endpoints with presentation restricted were accepted ===
=== X.509: Add key usage to certificate requests ===


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|}
|}
calls from mobile endpoints with clir were accepted and associated to any configured mobile number. Even two-stage calls were possible<!---->
The Microsoft CA (standard) does not write the key usage into the certificate if it is not specified in the request.<!---->
''Status:''
pbx.cpp
      
      
=== Stopping/Starting of dyn PBX did not work correctly ===
=== DHCP-client  monitors ethernet link down/up events and revalidates current lease after link up ===


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|}
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Several Problems:<br/>- Trap could happen<br/>- Slave was not unregistered/re-registered<br/>- Registrations did not work after restart<br/>- ...<!---->
This prevents problems when a device is hot plugged to another network.<br/>Further this helps to overcvome a problem with certain cable modems.     <!---->
''Status:''
   
pbx.cpp, pbx.h, pbx_general.xsl, pbx_password.xsl
== Bug Fixes ==
 
 
      
      
=== Kerberos server with no realm configured stops listening when processing requests ===
=== SOAP, Send leg2Info.originalCalled Info ===


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|}
|}
If a Kerberos server with no realm configured received a Kerberos request it stopped listening. After that the server was not reachable any more.<br/><br/>Files: kerberos_kdc.cpp<!---->
As CallInfo.No with type="leg2orig"<!---->
''Status:''
pbx_xml.cpp
      
      
=== Potential trap with Mobility ===
=== PBX CF Filter for external calls did not work as expected in case of chained CFs ===


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|[http://mantis.innovaphone.com/view.php?id=66599 66599]
|}
|}
A trap could happen if a no response timer for a waiting call expired right after a recall for the waiting call was attempted<!---->
A filter for external calls did not match if the external call was forwarded already by an internal user<!---->
''Status:''
''Status:''
pbx_mobility.cpp
pbx.cpp
      
      
=== GSM License alorithm did not work with Feature Codes ===
=== Gateway: Trap in case of collision of hold and clearing from remote ===


{|
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|}
|}
There are features, which are transmitted as A<feature>B DTMF sequences. For example the calling Id can be sent this way. This did not work together with the license check.<!---->
This could happen on gateways with analog interfaces if the R-Key was pressed right when the other side hung up<!---->
''Status:''
pbx_mobility.cpp
      
      
=== Leak in TLS socket ===
=== H.323 potential trap if AlertingNumber is received ===


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|}
|}
The TLS socket did not delete the data from SOCKET_SEND if it was disconnected before.<!---->
is no problem with existing equipment, because we don't know of any sending an AkertingNumber. Could become an problem if we do this sometimes in the future<!---->
''Status:''
tls.cpp
      
      
=== HTTP to HTTPS redirect ===
=== H.323 Coding error, when forwarding tunneled SDP in some cases ===


{|
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|}
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Devices trap or redirect does not work if force https is enabled and some http pages are requested.<!---->
This could happen if during call setup a media negotiation happened on a call with a SIP and a H.323 leg.<br/><br/>This happened for example if a call was received from a SIP Trunk to a Quickdial object in the PBX. The outgoing call from Quickdial could fail because of this.<!---->
''Status:''
http.h, http.cpp
      
      
=== formatting of small cf cards didn't work ===
=== Release not forwarded in quick dial object ===


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The formatting of small cf cards (<512 MB) didn't work, as the cluster size was calculated too high.<br/>Smaller cluster sizes are used now.<br/><!---->
If the called party released the call, the remote party didn't get the release.<!---->
      
      
=== formatting of an unknown first partition broke this partition ===
=== possible noise in PRI connections with ip6010 ip3010 ip1060 ===


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|}
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If a cf card has partitions and the first partition wasn't recognized, the formatting of this partition broke the first partition, as the wrong boot sector was used.<br/><!---->
some few gateways may produce noise when using the PRI ports. This can be fixed with a new CPLD code contained in future firmware.<!---->
''Status:''
cpld.h
      
      
=== XML attribute "href" in PROPFIND response was not URL encoded ===
= V8 Hotfix19 (80500.58) =
Changes included in Version 8 hotfix19
[http://mantis.innovaphone.com/view.php?id=67521 Definition]
 
== New Features ==
 
 
   
=== ip200a/230/240:  handset conversations can be monitored in a directly connected  headset ===


{|
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Problem: XML attribute "href" in PROPFIND response was not URL encoded.<br/><br/>Solution: URL-encode XML attribute "href" in PROPFIND response.<br/> <br/>Files: servlet_webdav.cpp<br/><br/>Products affected: All gateways with CF card<br/><br/>Risk: No risk.<!---->
This feature is required for a special application and is supported only for ip200a/230/240 phones with a directly connected headset (non DHSG).<br/>It is enabled via<br/>   config add INCA_DSP /handset-spy <volume><br/>whith <volume> in the range from 1..8  <!---->
   
== Bug Fixes ==
 
 
      
      
=== Trap if Waiting Queue Announcement reaches end while doing DTMF two stage dialing ===
=== IPxx10: error handling in sata driver ===


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The call to the waiting queue is terminated if the first announcement reaches its end. If then still DTMF two stage dialing was pending a trap happened<!---->
Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.<!---->
''Status:''
pbx_wait.cpp, pbx.cpp, pbx_api.h
      
      
=== PBX Mobility: Handling of disconnect from mobile phone improved ===
=== DECT: IP6000/IP6010/... default config Master mode off ===


{|
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|}
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If the DISC from the mobile phone was received from ISDN with in-band information it could take 30s after DISC until Mobile Phone could be called again. If the User with Mobility was monitored by SOAP (or TAPI), the call could hang until the SOAP/TAPI Application was terminated<!---->
Now the Dect Master is in mode off by default for the IP6000/IP6010/...<!---->
''Status:''
pbx_mobility.cpp
      
      
=== PPTP connection failed because of packet reordering by certain DSL providers/equipment ===
=== VM: Trap while processing self-forwarded call ===


{|
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|[http://mantis.innovaphone.com/view.php?id=67570 67570]
|}
|}
dial out PPTP connections to a central innovaphone IPxxx failed when tried from certain remote locations but succeeded from other locations.<br/>The reason for the failure was that packets sent by the central IPxxx were reordered by some network equipment and this case was not handled correctly in the connection setup phase.<!---->
VM: Trap while processing self-forwarded call<!---->
      
      
=== IP-DECT configuration reset state ===
=== SIP: Uninitialized data in SDP offer/answer ===


{|
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|}
|}
Reset is needed if the primary IP address is changed in standby mode.<!---->
Applies to G.726 exclusive calls only.<!---->
''Status:''
dectmaster.cpp
      
      
=== Waiting queue announcement does not stop ===
=== SIP: Interoperability with Lync and media-bypass ===


{|
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|}
|}
Problem: If WQ is configured w/o explicit announcement, the built-in MOH pattern is played to caller. When DTMF dialing starts, announcement is not stopped.<br/><br/>Solution: Keep built-in MOH pattern from being re-started.<br/> <br/>Files: webmedia.cpp<br/><br/>Products affected: All gateways<br/><br/>Risk: No risk.<!---->
Ack contained wrong To-Tag when calling a lync client in media-bypass scenario.<br/>Results into call drop after 30 seconds.<!---->
      
      
=== Trap on media recording ===
=== PBX: Don't forward original diverting_leg2 info if divertion is executed ===


{|
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|[http://mantis.innovaphone.com/view.php?id=67686 67686]
|}
|}
Problem: Media recording may cause a trap when destination HTTP server does not support PUT.<br/><br/>Solution: Fix error handling.<br/> <br/>Files: webdav_client.cpp<br/><br/>Products affected: All gateways<br/><br/>Risk: Small risk of collateral damage.<!---->
The leg2 information which is generated when executing an diversion already contains theoriginal called number from previous diversions, so the old leg2 info is not needed anymore. In fact it is harmfull if the call is received by an application only looking at the first leg2 info (e.g. Voxtron)<!---->
      
      
=== phone: hot desking with phone config stored on PBX did not work ===
=== PBX: License accounting in centralized licensing scenario wrong if master not available ===


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|}
|}
A PBX user has 'Store Phone Config' checked in the user object. If such a user registered via the 'Hot Desking' key a 'Delete Registration' key was added to the user confguration overriding another key on this position (if any).<br/><br/>This can be supressed now by configuring the 'Hot Desking' key with 'User Config Stored at PBX' checked.<br/>With this flag set the 'Hot Desking' key is functional only in the primary registration and creates the new registration without a 'Delete Registration' key. If required a 'Delete Registration' key must be provided in the stored config. If the stored config contains itself a 'Hot Desking' key with 'User Config Stored at PBX' checked this key works as a 'Delete Registration' key in a hot desking registration.<!---->
When the master is available the slave stores the licenses from the master including the usage. This stored usage included the licenses used by the slave itself, so if after a reset the master was not available the local usage just added to this.<br/><br/>Now from the stored usage the local usage is subtracted.<!---->
      
      
=== PBX: More then one registration was accepted for a Slave PBX. Caused problems with Standby switchover ===
=== PBX Trunk: Problem with Forking to trunk if multiple GWs are registered to Trunk ===


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|[http://mantis.innovaphone.com/view.php?id=67720 67720]
|}
|}
The normal rules were applied for registrations as Slave PBX. This meant if authentication was used multiple registrations were accepted. This caused the address to which registrations should be redirected to be set wrong. After a switchover to a standby slave and a switchback it could happen that on the Master registrations were not correctly redirected to the slave<!---->
If one of the gateways rejected the call (no channel, not connected, ...), the original call from which was forked was disconnected<!---->
''Status:''
pbx.cpp
      
      
=== UserRc did not work for some calls connected to special objects (such as e.g. waiting queue) ===
=== SIP: Fix for early media from Waitng Queue ===


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|[http://mantis.innovaphone.com/view.php?id=67775 67775]
|}
|}
Some PBX objects (such as the waiting queue) do not support sending of facilities to intercepted calls. Sending innovaphone remote control facility via SOAP UserRc on such calls did not work thus. <!---->
PROGRESS after ALERT was not handled by SIP stack.<br/>Now 183 Session Progress with SDP is send after 180 Ringing w/o SDP.<!---->
''Status:''
pbx_xml.cpp
      
      
=== PBX: Obsolete config from v7 created problems ===
=== H.323: A name_id of length 0 resulted in invalid H.450 coding ===


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|[http://mantis.innovaphone.com/view.php?id=67796 67796]
|}
|}
In PBX version7 it was possible to configure some parameter (e.g. Group Indications) at objects were it did not work correctly. In version 8 this is prohibited, but old config from v7 was still evaluated and was not easily removed.<br/><br/>Obsolete config is not evaluated anymore.<!---->
An empty name identification received was forwarded in H.323 as invalid H.450. Such a name is now forwarded as 'name not available'.<!---->
      
      
=== PBX Broadcast: Duplicate display of diverted Broadcast number if 'Execute Member Diversions' set ===
=== H.323 Malformed packet ===


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|}
|}
original called and diverting number was set to the Broadcast object<!---->
The ASN.1 encoder had a bug under one special condition: For a constrained character string with a maximum length of more or equal to 16bits, with an effective length of zero, the padding for octett alignment was missing for the zero length bitfield containing the string.<br/><br/>In H.323 this only happens for the CallIdentity used for H.450 call transfer message in case of blind transfer without consultation.<br/><br/>This fix breaks compatibility with earlier versions, for this reason this fix is available for version 9,8,7 and 6.<br/><br/>If phones and PBX with versions containing and not containing this fix are mixed  the following problems will occur:<br/>- A blind transfer without consultation (initiated with the redial key) is not possible<br/>- A call which was transfered without consultation is not displayed at the transfered-to phone as transfered<br/><!---->
''Status:''
pbx_bc.cpp
      
      
=== Trap when disabling or deleting SIP gateway interfaces ===
=== SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg ===


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|[http://mantis.innovaphone.com/view.php?id=67819 67819]
|}
|}
Problem: Trap when disabling or deleting a SIP gateway interface while interface is in DNS resolving state.<br/><br/>Solution: Wait for DNS query completion before deleting interface.<br/> <br/>Files: sip.cpp<br/><br/>Products affected: All gateways using SIP<br/><br/>Risk: No risk.<!---->
If in incoming SIP was routed to multiple destinations<br/>the final session could be media-relay although not configured.<!---->
      
      
= V8 Hotfix  4 (80500.11) =
=== SIP: DNS problem when SRV response provides no additional records ===
Changes included in Version 8 hotfix4
[http://mantis.innovaphone.com/view.php?id=53772 Definition]
 
== New Features ==
 


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67907 67907]
|}
If 2-step resolving is required (SRV and A) the service port<br/>of the SRV response got lost and default SI Pport 5060 was used.<!---->
      
      
== Bug Fixes ==
=== SIP: Trap when configuring STUN server on a SIP/TCP or SIP/TLS interface ===
 
 
   
=== Trap after re-configuring a SIP gateway interface ===


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|[http://mantis.innovaphone.com/view.php?id=67923 67923]
|}
|}
Problem: Reconfiguring a SIP registrar interface may leave the system in inconstent state. System trapts on next in coming call.<br/><br/>Solution: Fixed handling of interface re-configuration<br/><br/>Files: sip.cpp/h<br/><br/>Products Affected: Gateways with SIP registrar interfaces<br/><br/>Risk: No risk<!---->
STUN is for SIP/UDP only.<!---->
      
      
=== send disengageRequest also if call is cleared by TCP disconnect, respond to disengageRequest even if there is no call  ===
=== PBX: Master/Slave compatibility problem with version 9 and version 8 and non-ascii characters in PBX name ===


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|[http://mantis.innovaphone.com/view.php?id=67956 67956]
|}
|}
On an Avaya PBX it happened that the signalling TCP session was closed by the PBX just after a SETUP had been received. 1,5 seconds later a disengageRequest for this call was received but because the call did not exist anymore no disengageConfirm was sent. Because of the missing confirmation the PBX gets hanging.<br/>Now we send disengageRequest also if call is cleared by TCP disconnect and respond to disengageRequest even if there is no call  <!---->
In version 8 only latin1 characters were allowed, which means in unicode the high byte was always 0. So it could be ignored and when sending location information between master and slave sometimes the high byte contained 0xff.<br/><br/>In version 9 this non-ascii location information was not correct unicode at all.<br/><br/>The problem happened only if non-ascii characters were used when naming a PBX.<!---->
      
      
=== Hotdesking does not work when the primary registration is down ===
=== PBX: End of call intrusion was not signaled to the phone ===


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|[http://mantis.innovaphone.com/view.php?id=68007 68007]
|}
|}
A hotdesking registration inherits the gatekeeper configuration of the primary registration (primary/alternate gk address, gk ID).<br/>The hotdesking registration tries to register to the active gatekeeper of the primary registration.<br/>If the primary registration is down the configured primary gatekeeper is tried but in case of failure the alternate gatekeeper was not tried.<br/>Now the alternate gatekeeper is tried when the primary cannot be reached. <br/><br/> <!---->
The call intrusion tone was generated even if the intrusion was terminated<!---->
      
      
=== Wrong crypto tag in SDP answer ===
=== phone_inca:  "ETH0/Isolate PC Link" checkmark could not be cleared via WEB UI once set ===


{|
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|[http://mantis.innovaphone.com/view.php?id=53834 53834]
|[http://mantis.innovaphone.com/view.php?id=68098 68098]
|}
|}
Problem: On SIP/H323 interworking scenarios, the crypto tag in the SDP answer was wring. did not match the offer's tag.<br/><br/>Solution: Fix crypto tag in SDP answer.<br/><br/>Files: sip.cpp<br/><br/>Products Affected: SIP devices<br/><br/>Risk: No risk<!---->
Only a WEB UI problem, a "config rem ETH0 /isolate-pc" did help.<!---->
      
      
=== PBX SOAP Function LocationURL returns wrong URL if dynamic PBX ===
=== SIP: Interoperability with LinkSys SPA3102 ===


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|[http://mantis.innovaphone.com/view.php?id=68174 68174]
|}
|}
The URL does not contain the correct module name (PBX0-<id>)<!---->
LinkSys SPA3102 gives "g729a" as RTP payload type mapping:<br/><br/>   v=0<br/>    o=- 510843041 510843041 IN IP4 192.168.10.20<br/>    s=-<br/>    c=IN IP4 192.168.10.20<br/>    t=0 0<br/>    m=audio 16404 RTP/AVP 18 100 101<br/>    a=rtpmap:18 G729a/8000<br/>   a=fmtp:18 annexb=no<br/>    ...<br/><br/>Needs to be handled.<!---->
''Status:''
pbx.cpp, pbx.h, pbx_api.h, pbx_xml.cpp, h323_sig.cpp
      
      
=== Trap on call independent (CEI) signaling ===
=== Gerneral/Admin page was broken if too many authentication servers were configured ===


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|[http://mantis.innovaphone.com/view.php?id=68231 68231]
|}
|}
Problem: Trap when call independent (CEI) signaling is used on ISDN interface (e.g. Call Completion)<br/><br/>Solution: Fix handling of call independent (CEI) signaling<br/><br/>Files: gk_if.h gk.cpp/h relay.cpp<br/><br/>Products: ISDN Gateways<br/><br/>Risk: No risk<!---->
The number of authentication servers is now restricted to 10.<!---->
      
      
=== IP-DECT trap with unbound call objects ===
=== phone: intrusion call started in handset mode is not terminated when going on hook when TAPI or operator run on PBX ===


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|[http://mantis.innovaphone.com/view.php?id=68249 68249]
|}
|}
It is required to send empty events to call objects for binding.<!---->
With TAPI or operator running on the PBX the the signaling of a busy condition is changed such that a disconnect instead of a release is sent. The disconnect was not handled correctly, the hookswitch state was lost and the next on-hook signal was ignored. TThus teh call could be terminated with the disc-key only.<br/><!---->
''Status:''
signal.h, signal.cpp, dectmaster.cpp
      
      
=== HTTP client: Put  does not to work with digest authentication if the HTTP session is already authenticated ===
=== IP-DECT: Adding OEM radios to Kerberos realm did not work with passwords containing special characters ===


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|}
|}
problem: Put  does not to work with digest authentication if the HTTP session is already authenticated. The client tries to reauthenticate with by putting a file with 0 length.<br/><br/>solution: Just send the PUT request if the http session is already established<br/><br/>files: httpclient_i.cpp <br/><br/>products: all<br/><br/>risks: low risk <!---->
The password was not URL-decoded when reading it from the UI.<!---->
      
      
=== With hf3 fax 14440 and 1200 send didnt work on ip6000 and ip800 ===
=== DTMF user configuration with invalid checkbox check for presence setting ===


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|}
|}
With change 52247 the dsp packet size was 0 for the higher speeds, so the send failed<!---->
The check of the checkmark of the presence setting was wrong.<!---->
      
      
= V8 Hotfix  5 (80500.12) =
=== X509: Fix for reading innovaphone info from flash ===
Changes included in Version 8 hotfix4
[http://mantis.innovaphone.com/view.php?id=53960 Definition]
 
== New Features ==
 
 
   
=== IP-DECT Master call clear and list information ===


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|}
|}
Now it is possible to clear a DECT call and there are new informations about the master calls in the call list:<br/>- Uptime<br/>- Media<br/>- Encrypted call<br/><!---->
Parsing the innovaphone info text was incorrect<!---->
''Status:''
signal.h, dectmaster.h, dectmaster.cpp, dectmaster_calls.xsl, dectmaster_call.xsl, dectmaster.mak
      
      
=== IP-DECT OEM monitor function for location change ===
=== License: Be safe against factory reset during license invalidation ===


{|
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|}
|}
For a OEM module a new endpoint monitor function is added to notify about an endpoint location change.<!---->
If factory reset is done before license invalidation procedure is complete,<br/>will keep you from completing the license invalidation.<br/>Now the procedure can be completed even after factory reset.<!---->
''Status:''
Changed files: dectmaster.h, dectmaster.cpp
      
      
=== PBX Multicast object can optionally call endpoints without automatic connect ===
=== phone: DHSG headset not reset to idle after a hookswitch signal in idle state ===


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|}
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So that the phone rings<!---->
most DHSG headsets generate a hookswich signal and enter voice mode when taken out of basestation. This hookswitch signal was simply ignored.<br/>Now the voice mode is cleared after one second if there is no other DHSG event before.<!---->
''Status:''
pbx_mc.cpp, pbx_mc.h, pbx_edit_multicast.xsl
      
      
=== PBX Node 'incomplete Number' destination did not work for block dial calls ===
= V8 Hotfix20 (80500.59) =
Changes included in Version 8 hotfix20
[http://mantis.innovaphone.com/view.php?id=69989 Definition]
 
== New Features ==
 
 
   
=== ISDN interop issue with SecuGATE LI 30 from Sirrix ===


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|}
|}
A block dial call to an incomplete number was not sent to the configured destination for incomplete numbers, but failed.<!---->
The SecuGATE LI30 is sending/receiving ISDN INFO messages in Call Proceeding State (State 3 and state 9), which was not supported<!---->
''Status:''
pbx.cpp
      
      
=== IP-DECT consultation calls by TAPI connections ===
=== Allow multiple HTTP IP address filters (allowed stations) ===


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|}
|}
The DECT base station supports consultation calls initiated by TAPI connections (e.g. Servonic CtiServer with IXI-Call).<!---->
synced from V9<!---->
''Status:''
''Status:''
dtmffty.cpp, dectradio.h, dectradio.cpp
http.cpp<br/>http.h<br/>http.xsl
      
      
=== Max number of PBX call filters increased to 32 ===
== Bug Fixes ==
 
 
   
=== Gateway: Allow configuration of username and password for ENUM/SIP interfaces ===


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|}
The limit was 16 and this seems to be too small for some special applications<!---->
For rare where remote destination server asks for authentication.<br/>(And all remote destination servers ask for same auth or remote destination server s always the same.)<!---->
''Status:''
pbx_admin.cpp, pbx_global.xsl
      
      
=== IP800: Number of SIP interfaces and GWs increased to 16 ===
=== SIP/TCP: Transport error when connection is closed by client ===


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|}
|}
Same as already on IP6000<!---->
If transaction client closes connection before final response has been sent,<br/>the server tries to open a new connection toward ephemeral port of closed connection.<!---->
''Status:''
config.h
      
      
=== Improve mem/cpu Statistic Layout ===
=== SIP: Fix for Dialog-Info notification ===


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|}
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So it can be imported to spreadsheet applications more easily (see http://wiki.innovaphone.com/index.php?title=Howto:Device_Health_Check).<!---->
Send an empty dialig-info XML after inbound subscription.<br/>Required for interop with Grandstream GXP2010.<!---->
''Status:''
os.cpp
      
      
=== quick dial keys (keys not assigned to display line) of IP230/IP240 cannot be controlled via soap ===
=== SIP: Problem decoding INFO(application/dtmf-relay) ===


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|}
|}
key codes > 256 are used to control function keys (257..356 for F1...F100) but this did work only for F1 to F7. Now the keys can either be addressed using the codes 257 .. 356 or per key block using codes > 1000:<br/>1000 + key number ==> keys assigned to display<br/>2000 + key number ==> quick dial keys on the phone<br/>3000 + key number ==> quick dial keys on 1st extension keybank<br/>4000 + key number ==> quick dial keys on 2nd extension keybank<br/>5000 + key number ==> quick dial keys on 3rd extension keybank<br/>The key numbers start with 1<br/><!---->
DTMF digit was not decoded from message body if whitespace between EQUAL and DIGIT.<br/>E.g. Signal= 5<!---->
      
      
== Bug Fixes ==
=== Phone: Changing config option /sip-hold does not call for reset ===
 
 
   
=== IP-DECT DRAM Upload Link ===


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|}
|}
The DRAM Upload Link is fixed.<!---->
Reset is required and 'reset required" must be displayed.<!---->
''Status:''
update_hdr_1200.xml
      
      
=== Trap if sending DTMF R-Key (**) while sending busy to mobile phone ===
=== Kerberos: Protect against ping pong attacks ===


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|[http://mantis.innovaphone.com/view.php?id=68822 68822]
|}
|}
If inband busy is played to mobile phone (e.g. after dialing wrong number) a DTMF R-key caused a trap.<br/><br/>This also happened after blind transfer with R-4 in this case the error was sending the inband busy in the first place.<!---->
Do not answer with an error message to unexpected or malformed messages.<br/><br/>This protects against the "Kerberos Server Spoofed Packet Amplification DoS" attack. The attack causes two Kerberos servers to send each other error messages in a ping pong style.<!---->
''Status:''
pbx_mobility.cpp
      
      
=== DHCP server/client ARP based address validation did not work ===
=== Potential Trap because of recursive loop, if "incomplete" deastination used at a Node to invalid name/number ===


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|}
|}
The DHCP server should check if an address is already in use by a device in the network before the address is offered to a client. The DHCP client should check an address offered by a server in case the server does no such check. Both checks were broken since V8 hotfix3.    <!---->
Check for loop implemented (merge from v10, v9)<!---->
      
      
=== PBX Mobility: Better handling of disconnect from mobile phone (potential trap on release collisions) ===
=== H.450: Bad encoding of DivertingLegInformation4 arguments ===


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|}
|}
Fixes from v9 development merged<!---->
DivertingLegInformation4 content coding was wrong.<br/>Wireshark displayed it as malformed.<br/><br/>Note:<br/>This fix causes interoperability problem with phones with older (non-fixed) firmware versions!<br/>Phones also require an updated firmware if PBX is updated.<!---->
''Status:''
pbx_mobility.cpp
      
      
=== IP-DECT idle display update for anonymous endpoints ===
=== PBX: Phone config was not sent to phone, if phone was power cycled shorty after registration ===


{|
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|[http://mantis.innovaphone.com/view.php?id=69280 69280]
|}
|}
No idle display update is done in the DECT system if not configured in the IP-DECT Master. This fix correct the behavior also for anonymous endpoints.<!---->
The new registration after the power cycle was not detected as new registration but as re-transmission of the previous registration, so it was not reported to the PBX and no phone config was sent<!---->
      
      
=== Webdav client must URL decode content of href element in PROPFIND response body ===
=== SIP: NOTIFY sent after 302 moved temporarily ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54101 54101]
|[http://mantis.innovaphone.com/view.php?id=69282 69282]
|}
|}
According to RFC-2518 the href XML Element is URL encoded.<!---->
After processing "302 moved temporarily" on an outbound call a NOTIFY (sipfrag) was sent.<!---->
      
      
=== PBX potential trap when turning OFF ===
=== IP-DECT: New radio BMC firmware PCS05Ak ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54120 54120]
|[http://mantis.innovaphone.com/view.php?id=69468 69468]
|}
|}
If a PBX configured as master is turned off (setting mode to OFF) a trap could happen<!---->
The new radio BMC firmware PCS05Ak for the IP1200 fixes a trap by the DECT system if more than 255 DECT users without an endpoint subscription are sent to it.<!---->
''Status:''
pbx.cpp, ep_lib.cpp
      
      
=== directory search object transfer ignored cfu of called object ===
=== PBX: Reject calls without media, if no known facility ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54122 54122]
|[http://mantis.innovaphone.com/view.php?id=69477 69477]
|}
|}
The transfer of the directory search object should use the transfer method in a way, that checks the called object for cfu etc. and this is done now.<!---->
Fixes compatibility issues between versions. For example presence subscription sessions from v8 phones being forwarded to voicemail<!---->
      
      
=== Removed class="pad" attribute (because type-specific XSLs do not use this either) ===
=== PBX: Filter for internal or external calls at CFs did not work CFB or CFNR if call already diverted ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54153 54153]
|[http://mantis.innovaphone.com/view.php?id=69483 69483]
|}
|}
Removed class="pad" attribute (because type-specific XSLs do not use this either).<br/>Causes mis-alignment of data cells.<br/>See fault.png<!---->
Problem:<br/><br/>User A has CFU to User B<br/>User B has CFNR for ext. Calls only to User C<br/><br/>An internal call to A was diverted to B (ok) and after no response diverted to C (nok)<!---->
      
      
=== GSM License was not sent to slave correctly ===
=== PBX Waiting: No ringback when doing two-stage dialing to a Gateway/Trunk object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54239 54239]
|[http://mantis.innovaphone.com/view.php?id=69531 69531]
|}
|}
The Version was missing. Because of this a license was not freed if not used anymore.<!---->
A local ringback is now switched on, when receiving ALERT from called party<!---->
      
      
=== PBX SOAP Method UserInfo did not work right after UserCall ===
=== phone: assume an outbound call to be an external call if connected number info is missing in connect event ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54261 54261]
|[http://mantis.innovaphone.com/view.php?id=69581 69581]
|}
|}
As long as the call to the local phone was not connected this call was ignored.<!---->
In certain ISDN configurations the PBX can not provide the connected number info in the connect event for an outbound call. In this case the the call was assumed to be an internal call and consequently was not recorded when transparent recording of external calls was configured.<br/>Now an external call is assumed in this case.<br/> <br/><!---->
      
      
=== H.323 compatibility issue with missing mandatory fields ===
=== phone: VLAN signaling priority could not be configured via phone menu ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54262 54262]
|[http://mantis.innovaphone.com/view.php?id=69633 69633]
|}
|}
The fields multipleCalls an maintainConnection are marked as mandatory in the asn1 definition. Some H.323 implementations reject messages if these fields are missing even though decoding is possible.<!---->
Under "Menu/Administration/IP Settings/VLAN" there was only a "VLAN Priority" menu item. This menu item did override the 'Priority RTP Data' value but not the 'Priority Signaling' value as entered via WEB configuration.<br/>Now the items "Prio. RTP Data" and "Prio. Signaling" replace the "VLAN Priority" item.<!---->
''Status:''
h323sig.cpp
      
      
=== One way voice with SRTP, and transfer executed by slowstart endpoint accross PBXs ===
=== IPxx10-sata: trap after config /trace /track activation ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54283 54283]
|[http://mantis.innovaphone.com/view.php?id=69642 69642]
|}
|}
Call from phone with SRTP configured and registration with password to a slowstart endpoint on another PBX, which does transfer back to another SRTP phone on original PBX causes the call between the two SRTP phones to be one-way-voice<!---->
Instruccion was accessing uninitialized pointer.<!---->
''Status:''
h323sig.cpp
      
      
=== memory leak when logging to a TCP log server which did connect but did not consume the sent data fast enough ===
=== IP6010: RSTP did not work ===


{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54304 54304]
|[http://mantis.innovaphone.com/view.php?id=69731 69731]
|}
|}
too much data was buffered when a TCP log server accepted the connection but did  process log data at a lower rate as it was produced (or no data at all).  <!---->
When connecting ETH0 in RSTP mode to an HP Pro Curve switch the switch changed the port state to blocked after negotiation phase<!---->
''Status:''
files: mv78x00_drv.cpp, mv78x00_drv.h
      
      
=== HTTP client: User-Agent header must conform to syntax rules of rfc-2616 ===
=== SIP: Trap when handling NOTIFY(application/qsig) ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54355 54355]
|[http://mantis.innovaphone.com/view.php?id=69771 69771]
|}
|}
IIS rejected PUT request due to illegal User-Agent value:<br/><br/>PUT /innovaphone//bced28b8e909d311a6f70090331b341f.pcap HTTP/1.1<br/>User-Agent: innovaphone IP3028.00 hotfix4 [80500.11/8050011/200]<br/>Host: 10.0.0.3<br/>Transfer-Encoding: chunked<br/><br/>...<br/><br/>HTTP/1.1 500 Server Error<br/>Server: Microsoft-IIS/5.1<br/>Date: Fri, 16 Jul 2010 06:15:01 GMT<br/>Connection: close<br/>Content-Type: text/html<br/>Content-Length: 86<br/><br/><html><head><title>Error</title></head><body>Parametr jest niepoprawny. </body></html><!---->
Traps if no progress indicator present in tunneled DISCONNECT message.<!---->
      
      
=== PBX: Transfer of parked call failed ===
=== IP6010: SRTP using AES-192 and AES-256 did not work ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54372 54372]
|[http://mantis.innovaphone.com/view.php?id=69828 69828]
|}
|}
A tranfer of a call, which was parked by the remote peer, did result in a call without media<!---->
Due to a bug in the encryption driver of the IP6010, only AES-128 worked on this platform.<!---->
''Status:''
pbx.cpp, pbx.h, pbx_api.h
      
      
=== dyn. PBX license limits did not work ===
= V8 Hotfix21 (80500.60) =
Changes included in Version 8 hotfix21
[http://mantis.innovaphone.com/view.php?id=69991 Definition]
 
== New Features ==


{|
 
|Status
   
=== Gateway: Forward Display Info received from ISDN Setup to H.323 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="green">Closed</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54433 54433]
|[http://mantis.innovaphone.com/view.php?id=70562 70562]
|}
|}
- if the total usage of a license over all PBXs exceeded the limit of a single PBX, no registration at this PBX was not possible anymore<br/>- The total usage was displayed at a dyn PBX. This needs to be hidden from a dyn. PBX admin<br/>- a dyn. PBX should only be able to sub-license to slave PBXs up to the limit<!---->
needed for compatibility with SecuGATE LI30<!---->
''Status:''
inno_lic.cpp, inno_lic.h
      
      
=== SIP: Duplicate Call-ID on forked calls ===
=== phone: LED mode of Join Group function key can be set both for idle and for active state ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54456 54456]
|[http://mantis.innovaphone.com/view.php?id=71247 71247]
|}
|}
Causes trouble when forked calls go to same destination.<br/>INVITE may be considered as looped.<!---->
sometimes the "not in group" state must be signaled as the exception<!---->
''Status:''
sip.cpp
      
      
=== Cause code in PROGRESS not forwarded by PBX ===
=== phone: Mic Off/On controllable via Soap:UserRc(&lt;call&gt;,14/15) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54466 54466]
|[http://mantis.innovaphone.com/view.php?id=71721 71721]
|}
|}
This is a problem since on QSIG lines a PROGRESS with cause indicates a call clearing, whereas a PROGRESS without cause could be just indication of inband info.<!---->
To allow Soap app's control of the mute key<!---->
''Status:''
=== Other new Features ===
pbx.cpp
   
=== PBX Mobility: A call to a busy mobile endpoint should be rejected in case of twin phone ===


{|
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|-
|Id
|valign=top nowrap=true|[http://mantis.innovaphone.com/view.php?id=71747 71747]
|[http://mantis.innovaphone.com/view.php?id=54469 54469]
|valign=top nowrap=true|jfr
|phone_coldfire(OEM device): keypad light and display can be switched off
|}
|}
If a user is talking on the mobile phone even without mobility and a call comes in to the fixed phone, which causes a call to the mobile phone because of mobility, this incoming call should be rejected with cause busy and the fixed phone should stop ringing<!---->
 
''Status:''
   
pbx_mobility.cpp
== Bug Fixes ==
 
 
      
      
=== Potential trap during boot ===
=== VM, email attachments weren't sent for https URLs ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54479 54479]
|[http://mantis.innovaphone.com/view.php?id=69965 69965]
|}
|}
If flashman marks the first flash segment as unused (state=0) the device may trap during boot<!---->
i.e. voicemail wave attachments<!---->
''Status:''
boot.c bootxxxx.y
      
      
=== Trap in media handling under high load when closing SRTP channels ===
=== SIP: Reject unsupported method types with "SIP/2.0 405 Method Not Allowed" ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54615 54615]
|[http://mantis.innovaphone.com/view.php?id=70526 70526]
|}
|}
An assertion happend in the code, which was put in the trace buffer like<br/><br/>70:2793:594:6 - SRTP.57814 default(827790b8): serial_event(710)<br/>70:2793:594:7 - Assertion failed line 717 in common/os/os.cpp, object deleted<br/><!---->
Not ignoring them.<br/><br/>PING sip:tel3@PBX0 SIP/2.0<br/>Via: SIP/2.0/UDP 172.16.77.14:5060;branch=z9hG4bK937906956;rport<br/>From: ;tag=3520474<br/>To: <sip:tel3@PBX0><br/>Call-ID: 193626070<br/>CSeq: 20 PING<br/>Contact: <sip:tel3@172.16.77.14><br/>Max-Forwards: 70<br/>Content-Length: 0<br/><br/>SIP/2.0 405 Method Not Allowed<br/>Via: SIP/2.0/UDP 172.16.77.14:5060;branch=z9hG4bK937906956;rport<br/>From: <sip:tel3@PBX0>;tag=3520474<br/>To: <sip:tel3@PBX0><br/>Call-ID: 193626070<br/>CSeq: 20 PING<br/>Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH<br/>Content-Length: 0<br/><br/><!---->
''Status:''
media.cpp, os.cpp, os.h
      
      
=== Potential trap when removing CF cards while writing files ===
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54666 54666]
|[http://mantis.innovaphone.com/view.php?id=70675 70675]
|}
|}
A file_event_close was sent twice, so that it was sent to an already deleted object.<br/>Use a flag to prevent a second file_event_close.<!---->
After closing regstration Dectmaster starts another call.<br/>Call is rejected, but signaling enity is deleted before call object.<!---->
      
      
=== Problems with PBX OEM version (License, UI) ===
=== SIP: No route processing if neither Record-Route header nor Contact header is present ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54787 54787]
|[http://mantis.innovaphone.com/view.php?id=70971 70971]
|}
|}
Change relevant only for a single OEM<!---->
Misleading trace message:<br/>  sip_call::process_routing(0xA8) Unsupported transport protocol: sip:user@domain.com;user=phone<!---->
      
      
=== Config Wizard did not handle blanks in SIP trunk parameters ===
=== when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54802 54802]
|[http://mantis.innovaphone.com/view.php?id=71246 71246]
|}
|}
The parameter (e.g. the authentikation name) was cut off at the blank<!---->
after a template has been edited unchanged information units inherited from predecessor templates must be removed from the edited template. this did not work for the dialing location and thus a later change in a predecessor template had no effect. <!---->
''Status:''
config_wizard.txt
      
      
=== Potential trap with cascaded waiting queues ===
=== SIP: No media after accepting a waiting call ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54804 54804]
|[http://mantis.innovaphone.com/view.php?id=71288 71288]
|}
|}
This trap was introduced by<br/><br/>http://wiki.innovaphone.com/index.php?title=Support:DVL-Roadmap_Firmware_V8#Trap_if_Waiting_Queue_Announcement_reaches_end_while_doing_DTMF_two_stage_dialing<br/><br/>in v8hf3<!---->
Call waiting on a phone.<br/>Going onhock while another call is waiting starts ringer.<br/>After going offhook again the waiting call is accepted, but no media in both directions.<!---->
''Status:''
pbx.cpp
      
      
=== Gateway Routing: Enblock calls should not match mappings with additional digits ===
=== phone: send config to PBX only when the config was edited on phone ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54828 54828]
|[http://mantis.innovaphone.com/view.php?id=71387 71387]
|}
|}
Enblock calls did match to mappings with additional digits and were rejected as incomplete. Now the search is continued and a matching mappings after this are executed.<!---->
A config from an older PBX may contain duplicate elements which are stripped by the phone. I such a stripped config is sent back to the PBX the PBX will return the old config again.<!---->
''Status:''
gk.cpp
      
      
=== PBX admin with 'all-objects' rights should be allowed to edit filters ===
=== SIP: Interop with Nortel CS1000 SIPLine GW ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54831 54831]
|[http://mantis.innovaphone.com/view.php?id=71426 71426]
|}
|}
Inconsistent, because such an admin could remove filters from user object.<!---->
Nortel sends 183/Progress with 'sendrecv' answer<br/>followed by UPDATE with 'inactive' offer<br/>followed by UPDATE with 'sendrecv' offer.<br/><br/>Innovaphone SIP stack remains in 'inactive' state.<!---->
''Status:''
pbx_admin.cpp, pbx_global.xsl
      
      
=== ISDN PRI (US): Channel status 'out of service' should be cleared with channel restart ===
=== SIP: Interoperability with MX-ONE ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54888 54888]
|[http://mantis.innovaphone.com/view.php?id=71480 71480]
|}
|}
The US ISDN protocols allow the switch to set channels out-of-service. Normally this should only happen if subscription to channels changes, but it can be seen under error conditions as well. A channel restart procedure should clear this status<!---->
A semi-attended transfer fails if MX-ONE sends INVITE(Replaces)<br/>instead of 200/OK when connecting a call.<!---->
''Status:''
q931.cpp
      
      
=== ISDN: FACILIYT as first response to SETUP caused call clearing ===
=== SIP: Trap on timer expiration during call release ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54907 54907]
|[http://mantis.innovaphone.com/view.php?id=71699 71699]
|}
|}
This could happen if a Facility we were sending with the SETUP was not accepted by the other switch. Problem occured with Hipath and diverted calls.<!---->
Media negotiation watchdog timer expired after final SIG_REL went to app.<br/>But before app deleted the call object.<!---->
      
      
=== Mobility: Reject calls from mobile phone with cause user-busy or reject only ===
=== phone: display info provided by SETUP or CONNECT was ignored ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54928 54928]
|[http://mantis.innovaphone.com/view.php?id=71727 71727]
|}
|}
This is a workaround for a bug in the T-Mobile network, which is sending an ALERT if a absent mobile phone is called before the call release.<!---->
only the display info provided by an INFO event was handled  <!---->
''Status:''
pbx_mobility.cpp
      
      
= V8 Hotfix  6 (80500.20) =
= V8 Hotfix22 (80500.61) =
Changes included in Version 8 hotfix6
Changes included in Version 8 hotfix22
[http://mantis.innovaphone.com/view.php?id=54972 Definition]
[http://mantis.innovaphone.com/view.php?id=71744 Definition]


== New Features ==
== New Features ==
Line 4,845: Line 5,033:


      
      
=== Modified interface for OEM password complexity ===
=== Debug information on assertion ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55087 55087]
|[http://mantis.innovaphone.com/view.php?id=71961 71961]
|}
|}
OEMs can now implement a module for checking password complexity<!---->
More debug information on default event handler.<!---->
''Status:''
files: <br/>./common/lib/lib.mak <br/>./common/interface/interface.mak <br/>./common/interface/pwd_complex_api.h <br/>./common/interface/pwd_complex_api.cpp <br/>./ascom/pwd_complex/pwd_complex.h <br/>./ascom/pwd_complex/pwd_complex.cpp <br/>./box/command/command.h <br/>./box/command/command.cpp <br/>./dect/users/dectusers.cpp <br/><br/>
      
      
=== OEM password complexity for Kerberos users ===
=== SIP: Get display information from Call-Info header in register response ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55091 55091]
|[http://mantis.innovaphone.com/view.php?id=72448 72448]
|}
|}
The Kerberos module can now check the complexity of user passwords if this is implemented by the OEM software.<!---->
Get display information from Call-Info header in 200/OK <!---->
''Status:''
files:<br/>kerberos_db.cpp
      
      
=== Simplified administration UI for some OEMS ===
=== PBX: Forward original received ISDN display element to picking up or forwarded call ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55137 55137]
|[http://mantis.innovaphone.com/view.php?id=73278 73278]
|}
|}
Some items in the adminstration user interface can now be hidden by setting special xml-modes (admin-basic,admin-advanced).<!---->
In the display element from ISDN there could be vital information from equipment like crypto gateways. This should be available also if the call was picked or forwarded.<!---->
''Status:''
   
files:<br/>- ./dect/users/dectusers.cpp<br/>- ./dect/master/dectmaster.cpp<br/>- ./platform/platform.mak<br/>- ./platform/asc_diagnostics_basic.xml<br/>- ./platform/asc_diagnostics_hdr_basic.xml<br/>- ./platform/dect_hdr.xml<br/>- ./platform/eth0_hdr.xml<br/>- ./platform/left_menu.xml<br/>- ./box/httpfiles/reset_hdr.xml<br/>- ./common/platform/ip1201.cpp<br/>- ./box/command/command.h<br/>- ./box/command/command.cpp
== Bug Fixes ==
 
 
      
      
=== Hide some pages and items on admin UI while OEM provisioning is running ===
=== TCP: Roundtrip measurement wrong in case of packet loss ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55162 55162]
|[http://mantis.innovaphone.com/view.php?id=71985 71985]
|}
|}
While the provisioning module of an OEM is active, special xml-modes are set that can be used to hide items from the administration interface.<!---->
In case of packet loss, way to high round trip values were measured. If the packet-loss was to high, this could result in a constantly increasing re-transmission timeout value.<!---->
''Status:''
files:<br/>./ascom/httpfiles/asc_ntp.xsl<br/>./ascom/httpfiles/asc_dectfty.xsl<br/>./common/platform/ip1201.h<br/>./common/platform/ip1201.cpp<br/>./common/service/ntp/ntp.cpp<br/>./dect/fty/dectfty.cpp
      
      
=== IP-DECT OEM location monitor function change ===
=== SIP: Trap on IP-DECT when re-configuring PBX link ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55294 55294]
|[http://mantis.innovaphone.com/view.php?id=72190 72190]
|}
|}
For OEM modules the location monitor is changed.<!---->
85:2195:425:7 - REG_PRI.4 default(8102be48): serial_timeout<br/>85:2195:425:7 - Assertion failed line 748 in common/os/os.cpp, object deleted<br/><!---->
''Status:''
''Status:''
dectmaster.cpp
Merged to 09-80500
      
      
=== DTMF feature call completion can be also used for no response ===
=== Scheduling improved to avoid processes not being scheduled during long flashman operations ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55309 55309]
|[http://mantis.innovaphone.com/view.php?id=72243 72243]
|}
|}
The feature is not only usable after a busy call, but also after a call with no response.<!---->
In version 7 it could happen, that IP and other processes were not scheduled any more during periods of long flashman operations (e.g. bootcode update or reorganizing flash).<br/><br/>In version 8 and higher there was already a fix for this problem, but this included special handling of the flashman priority level, which was not a good solution even if it worked.<!---->
      
      
=== Update client option for short URL ===
=== SIP: Cleanup failed (resources leaking) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55324 55324]
|[http://mantis.innovaphone.com/view.php?id=72284 72284]
|}
|}
For OEM http server the update client should not append additional options to the update server URL.<!---->
Call and channel objects were not freed sometimes<br/>when INVITE was followed by CANCEL very fast.<!---->
''Status:''
update.h, update.cpp
      
      
=== SIP: Detect remote party identity change ===
=== PBX SOAP: Called Number presentation not correct for calls to 'local' objects ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55329 55329]
|[http://mantis.innovaphone.com/view.php?id=72396 72396]
|}
|}
Remote party update did not work in all cases:<br/>If initial INVITE got no identity header, but re-INVITE contains identity header.<!---->
If an object is marked as local, the PBX prefix should not be included in the called number.<br/><br/>This is a fix, which is merged from v9 and higher back into v8<!---->
''Status:''
sip.cpp/h
      
      
=== IP-DECT OEM configuration options for registration speed ===
=== update - scfg command could hang when the HTTP session was broken or prematurely closed by the server ===


{|
{|
Line 4,954: Line 5,134:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55499 55499]
|[http://mantis.innovaphone.com/view.php?id=72708 72708]
|}
|}
For an OEM PBX it is necessary to configure the user's registration speed to this PBX. Used only in the OEM DECT device.<!---->
in consequence update script processing was stopped until reboot<!---->
''Status:''
dectmaster.h, dectmaster,cpp.
      
      
=== SIP: Added Microsoft propriatary extension "ms-acceptedby" for OCS compatibility ===
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55510 55510]
|[http://mantis.innovaphone.com/view.php?id=72729 72729]
|}
|}
A forked call that is accepty elsewhere is counted as "missed call" by OCS unless Microsoft specific extension is add to Reason header.<br/>  Reason: SIP;cause=200;text="OK";ms-acceptedby="sip:user@domain.com"<br/>According to [MS-SIPRE].pdf<!---->
When Dectmaster registers user at PBX using SIP protocol<!---->
      
      
=== A DHCP client with "/keep on" should not fall back to dicsover mode if the lease is due ===
=== PBX: Called Name displayed when calling an object with forking was wrong ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55561 55561]
|[http://mantis.innovaphone.com/view.php?id=72735 72735]
|}
|}
"/keep on" forces reusing the remembered lease if no DHCP server is responding after boot. But if the server failed to respond to the final rebind request for a regularly obtained lease a new recovery was started.<br/>Now in this case the lease is used further, a request for the lease and an ARP requests to check if the IP address is not assigned to another device are sent in regular intervals. <!---->
The name of the forking destination was displayed instead of the name of the called object<!---->
      
      
=== SIP: Hide product information in reject responses ===
=== PBX: No Audio if call thru Waiting Queue DTMF destination, was transfered to BC-Conf ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55620 55620]
|[http://mantis.innovaphone.com/view.php?id=72746 72746]
|}
|}
Don't be kind to SIP scan tools.<!---->
Problem caused by call state management error in PBX for calls connected without alert if alert was received later<!---->
''Status:''
siptrans.cpp
      
      
=== Include modes into configuration page of update client ===
=== SIP: Memory leak during transfer ===


{|
{|
Line 5,002: Line 5,178:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55669 55669]
|[http://mantis.innovaphone.com/view.php?id=73003 73003]
|}
|}
Needed for OEM specific XSL.<!---->
Occured on internal testing only (002-conf-with-bcast.xml)<!---->
   
= V8 Hotfix25 (80500.65) =
Changes included in Version 8 hotfix25
[http://mantis.innovaphone.com/view.php?id=78245 Definition]
 
== New Features ==
 
 
      
      
=== Phone: Problems with 'Presence' Fkey ===
=== HTTP-Client: MD5-sess authentication ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55785 55785]
|[http://mantis.innovaphone.com/view.php?id=77773 77773]
|}
|}
Presence Fkey requires working presence subscription.<br/>Presence subscription may fail from time to time due to several reasons.<br/>Reliable re-establishment is required.<!---->
HTTP Digest Authentication with alogrithm=MD5-sess.<br/>Choose the first supported "WWW-Authenticate" line from 401 response headers.<br/><br/>Needed for new versions of IIS.<br/><br/><!---->
''Status:''
''Status:''
phonesig.cpp
http://wiki.innovaphone.com/index.php?title=Support:DVL-Feature_Requests#HTTP_Client
      
      
== Bug Fixes ==
== Bug Fixes ==
Line 5,023: Line 5,207:


      
      
=== SIP: Media-negotiation after call transfer failed (no audio) ===
=== IP6010: Wrong timer under high load ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54442 54442]
|[http://mantis.innovaphone.com/view.php?id=71001 71001]
|}
|}
Re-negotiation after call transfer failed.<br/>Results into no-audio condition.<!---->
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQ´s a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs.<br/>-The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled.<br/><!---->
''Status:''
''Status:''
sip.cpp/h
ip6010.cpp<br/>ip6010.h
      
      
=== send busy tone from PBX dtmf object for not working cf with diversion filter ===
=== ip6010/3010/1060: Ethernet transmit packet length is sometimes wrong ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=54978 54978]
|[http://mantis.innovaphone.com/view.php?id=77774 77774]
|}
|}
If a diversion filter is set on a user and the dialed diversion to the pbx dtmf object is not allowed, a busy tone and a reject cause is now sent by the dtmf object.<!---->
Sometimes old content of the tx dma descriptor was used by the ethernet MAC.<br/>Now the memory write buffers are drained before enabling the tx dma.<!---->
''Status:''
mv78x00_drv.cpp<br/>mmu.S
      
      
=== IP-DECT Master call list OEM link and call state ===
=== ip6010/3010/1060: Ethernet receive packet sometimes delayed ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55026 55026]
|[http://mantis.innovaphone.com/view.php?id=77781 77781]
|}
|}
For OEM devices the call clear link doesn't work.<br/>Call state for the outgoing party is shown as "off-hook".<!---->
Sometimes the rx descriptor are processed with the next tx event.<br/>Now the rx queue is processed completely in on interrupt.<br/><!---->
''Status:''
''Status:''
dectmaster_call.xsl, dectmaster.cpp
mv78x00_drv.cpp<br/>mv78x00_drv.h
      
      
=== No Media event was generated even everything was normal for unanswered CC exec on IP-DECT ===
=== Gateway: Trap when interworking Call Completion ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55177 55177]
|[http://mantis.innovaphone.com/view.php?id=78228 78228]
|}
|}
Could happen for other traffic cases as well like rejected CC exec<!---->
Trap when interworking Call Completion.<br/><br/>LOG CALL 6 A:Call    ->                        / PRI2::->*::<br/>R_CALL free error c18a59b8<br/><!---->
''Status:''
dectradio.cpp, media.cpp
      
      
=== Point to Multipoint ISDN Maps need to set Type ISDN for CGPN-Out Map ===
=== TLS flow control damaged in versions 7 and 8 ===


{|
{|
Line 5,080: Line 5,264:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55184 55184]
|[http://mantis.innovaphone.com/view.php?id=78377 78377]
|}
|}
If not the mapping does not work for some networks and always the default number is used for outgoing calls as calling party number<!---->
The following fix was not good:<br/>#75004: TLS: Flow control for incoming data<br/><br/>Therefore TLS did not work correctly in the following releases:<br/>v7hotfix35 and v7hotfix36<br/>v8hotfix23 and v8hotfix24<br/><br/>No problem in version 9.<br/><!---->
''Status:''
gk.cpp
      
      
=== SIP: Digest authentication is rejected if username contains non-us-ascii characters ===
=== SIP: Be save against sudden death of SIP caller ===


{|
{|
Line 5,093: Line 5,275:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55217 55217]
|[http://mantis.innovaphone.com/view.php?id=78460 78460]
|}
|}
Digest authentication is rejected if username contains non-us-ascii characters.<br/>Expected special characters to be URL encoded, but most clients send it UTF8 encoded.<!---->
Lifetime of an INVITE trasnaction is not limited by any timeout<br/>after provisional response has been send/received.<br/>Sudden death of a caller make calls hang forever.<br/>Now overall lifetime of an INVITE server transaction is limited to 3 minutes.<br/>After expiration fimnal reject response is sent and call is released.<!---->
      
      
=== H.323: Cause received with PROGRESS message got lost ===
=== IP6000: Traps in DSP driver under high load ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55248 55248]
|[http://mantis.innovaphone.com/view.php?id=78591 78591]
|}
|}
This could result in calls to busy subscribers in a QSIG PBX to terminate with "recovery on time expiry" instead of "user busy"<!---->
under high load timing may change. Checks in driver relaxed to take this into account.<!---->
''Status:''
h323sig.cpp
      
      
=== SIP: Outgoing call (early, not connected) was not canceled (sometimes) on ISDN interworking scenario ===
=== SIP: Wrong number of waiting messages (MWI) ===


{|
{|
Line 5,117: Line 5,297:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55277 55277]
|[http://mantis.innovaphone.com/view.php?id=78890 78890]
|}
|}
An incoming DISCONNECT with progress indicator did not caused the outgoing SIP call to be canceled.<!---->
MWI: Number of voice messages not decoded from incoming NOTIFY(application/simple-message-summary).<br/>Was either 1 or 0.<!---->
''Status:''
sip.cpp
      
      
=== Gateway: divertingLeg2 was not passed in some cases ===
=== IP6010/3010/1060/0010: RSTP not working ===


{|
{|
Line 5,130: Line 5,308:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55310 55310]
|[http://mantis.innovaphone.com/view.php?id=79251 79251]
|}
|}
divertingLeg2 got lost during re-routing in Gateway.<br/>E.g. routing each call over TONE caused the divertingLeg2 to disappear.<!---->
RSTP packets were sent to but not received from switch port <!---->
''Status:''
checked in to 8.00,09-80500
   
= V8 Hotfix26 (8079900) =
Changes included in Version 8 hotfix26
[http://mantis.innovaphone.com/view.php?id=79737 Definition]
 
== New Features ==
 
 
      
      
=== Webdav: Handling of failed TCP when writing to file ===
=== Phones: Switch for phoneapp to disable auto-answer ===


{|
{|
Line 5,141: Line 5,329:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55460 55460]
|[http://mantis.innovaphone.com/view.php?id=80233 80233]
|}
|}
Webdav client needs handling of TCP error when writing to file<!---->
Disable/enable auto-answer support on phoneapp level.<br/><br/><!--<br/>phonesig_if.h<br/>phonesig.cpp<br/>--><!---->
   
== Bug Fixes ==
 
 
      
      
=== TEL interface: '#11' not callable if feature codes enabled ===
=== IP1060 IP3010 IP6000 IP6010: DSP packet debug didnt show some packets, version endian ,and dsp-trace port was wrong ===


{|
{|
Line 5,152: Line 5,344:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55537 55537]
|[http://mantis.innovaphone.com/view.php?id=79754 79754]
|}
|}
If feature codes are enabled for a TEL interface, the number '#11' without anything else can not be dialled.<br/>To fix please submit gateway's general page with the OK button or do a factory reset.<!---->
cleanup<!---->
''Status:''
''Status:''
config.h, relay_general.xsl
ac_491.cpp<br/>debug.h<br/>ac_dsp3.cpp<br/>trace.xsl<br/>
      
      
=== ARP requests/replies returned to the sender should be ignored ===
=== PBX Waiting: Missing ringback on call forward after announcement ===


{|
{|
Line 5,165: Line 5,357:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55560 55560]
|[http://mantis.innovaphone.com/view.php?id=87674 87674]
|}
|}
It was observed that in WLAN environments broadcasted ARP requests/replies may be received by the sender again. This results in some problems when DHCP checks if an IP address is not used by another device via ARP. Now returned requests/replies are simply ignored. <!---->
This was a collateral damage of<br/><br/>fix: #81370: PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== T.38 doesnt work if the call is transferred from a IP-Phone to a fax device ===
=== PBX Waiting: DTMF overlap dialing or blind transfer to same Waiting object was rejected with busy ===


{|
{|
Line 5,176: Line 5,368:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55569 55569]
|[http://mantis.innovaphone.com/view.php?id=87681 87681]
|}
|}
Affects IP2x IP30x fax gateways, the ipphone needs no update <!---->
Even if this was caused by a CFB or CFU on the dialed destination<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
''Status:''
   
ac_dsp3.cpp<br/>v7:<br/>ac494004.h <br/>ac498004.h
= V8 Hotfix 28 (80804) =
Changes included in Version 8 hotfix28
[http://mantis.innovaphone.com/view.php?id=82179 Definition]
 
== New Features ==
 
 
      
      
=== DECT: Trap while initiating blind transfer when using SIP as PBX protocol ===
=== Debug information on assertion ===


{|
{|
Line 5,189: Line 5,387:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55581 55581]
|[http://mantis.innovaphone.com/view.php?id=81973 81973]
|}
|}
0:0246:363:3 - GK-CALL free error 9481a58c<br/>0:0246:363:4 - last free=DECTMASTER-RADIO len=6<br/>0:0246:363:4 - caller=0x943796d0<br/>0:0246:363:4 - HEXDUMP<br/>      00000000 - 05 80 38 30  31 31                                  ..8011          <br/>0:0246:363:4 - BUFFER-FREE: obj at 0x9481a574 inconsistent<br/>0:0246:363:4 - HEXDUMP<br/><br/>Fixed in dectmaster.cpp<!---->
More debug information on default event handler.<!---->
      
      
=== Kerberos problem with encrypted password data containing null bytes ===
== Bug Fixes ==
 


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=55692 55692]
|}
Encrypted Kerberos passwords that are stored using LDAP may contain null bytes. Therefore they must not be handled as strings but as binary data when reading them.<!---->
''Status:''
files: kerberos_ldap.cpp
      
      
=== Phone: Make PBX-initiated calls don't look like transferred calls ===
=== HTTP-Server: Configuration of "Public compact flash access" did not work for all cases ===


{|
{|
Line 5,213: Line 5,402:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55784 55784]
|[http://mantis.innovaphone.com/view.php?id=82064 82064]
|}
|}
Do not send CT_SETUP.<!---->
E.g. /DRIVE/CF0/Neuer Ordner/ does not work, because HTTP request contains escaped sequences.<br/><br/><!--<br/>http.cpp<br/>--><!---->
      
      
=== "Join Group" function key lost state after a PBX reboot when the phone config was stored on the PBX ===
=== Gateway CDR with '0. 0' charge amount ===


{|
{|
Line 5,224: Line 5,413:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55790 55790]
|[http://mantis.innovaphone.com/view.php?id=82359 82359]
|}
|}
The Join Group function key lost it's state and did not work anymore after a PBX reset because the the phone config sent by the PBX after reregistration was not evaluated at the phone again. <!---->
Should be '0.00' instead<br/><!--<br/>fty.cpp<br/>--><!---->
      
      
=== flash variables may get lost after reboot (because  of an earlier  trap in the critical phase of flash garbage collection) ===
=== H.323:No Media for calls with reverse media to a H.323/SIP exclusive Code Media Relay interface ===


{|
{|
Line 5,235: Line 5,424:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55797 55797]
|[http://mantis.innovaphone.com/view.php?id=82408 82408]
|}
|}
Two valid segments bearing the same data are left back when a fragmented segment is compacted into a new one and the box traps after the new segment has been validated but before the old segment has been marked invalid.<br/>Because of a wrong comparison this situation was not resolved after reboot. Instead of deleting one of the segments the new segment was used until completely filled. Therafter all further allocations failed. This situation could only be cleared by a reset to factory defaults.<br/>Now, if the flash user is permitted to use only one segment (for example VARS on most boxes) the old segment is invalidated and the new compacted segment remains. If the flash user is permitted to use more segments (for example LDAP) the new segment is invalidated because it's not known which of the old segments was compacted.<br/> <br/> <!---->
The execlusive coder/media relay config is used to avoid media negotiation problems with carrier which do not support media renegotiations. In case of a call with reverse media to such an interface, this did not work. This happens for example if a CFNR is configured at a Waiting Queue which redirects a call, which received an announcement from the Queue to such interface.<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== PBX potential trap when parsing SOAP XML ===
=== Debug "HTTP_GET LOG_HTTP.1: retry, authentication failed" removed ===


{|
{|
Line 5,246: Line 5,435:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55812 55812]
|[http://mantis.innovaphone.com/view.php?id=82499 82499]
|}
|}
No child element found in SOAP XML<br/><!---->
<!-- httpclient_i.cpp --><!---->
      
      
=== Possible buffer overrun when reading/writing fat volumn id ===
=== SIP: Trap during call handling ===


{|
{|
Line 5,257: Line 5,446:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55858 55858]
|[http://mantis.innovaphone.com/view.php?id=82544 82544]
|}
|}
There was a possible buffer overrun when reading/writing the fat volumn id.<!---->
Trap during call handling<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Display name contained bad characters in some cases ===
=== SIP: SRTP key exchange failed ===


{|
{|
Line 5,268: Line 5,457:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55891 55891]
|[http://mantis.innovaphone.com/view.php?id=82616 82616]
|}
|}
Uninitialized buffer content presented as name identification.<!---->
Bug in base64 decoding of SRTP key.<br/><br/><!--<br/>sdp.cpp<br/>--><!---->
      
      
= V8 Hotfix 7 (80500.27) =
= V8 Hotfix 29 (80807) =
Changes included in Version 8 hotfix7
Changes included in Version 8 hotfix29
[http://mantis.innovaphone.com/view.php?id=56817 Definition]
[http://mantis.innovaphone.com/view.php?id=83649 Definition]


== New Features ==
== New Features ==
Line 5,280: Line 5,469:


      
      
=== SIP: Distinctive ring tones ===
== Bug Fixes ==
 
 
   
=== failure of analog ports of ip28 ===


{|
{|
Line 5,287: Line 5,480:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55948 55948]
|[http://mantis.innovaphone.com/view.php?id=82488 82488]
|}
|}
Handling of "Alert-Info: internal".<br/>Triggers special ring tone.<!---->
ip28 analogue ports do not react to incoming calls and hook-off. Problem could only be solved by reset.<!---->
''Status:''
sip.cpp
      
      
=== SIP: Send P-Asserted-Identity header in 180/Ringing ===
=== phone: when scrolling directory search results sometimes one of  the numbers of a contact was not displayed ===


{|
{|
Line 5,300: Line 5,491:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=56091 56091]
|[http://mantis.innovaphone.com/view.php?id=84362 84362]
|}
|}
Some UAC do not show called party's display name when added to To header by UAS.<br/>We now provide PAI header in provisional responses also containing the called party's display name.<!---->
the tag characters assigned to the different numbers were not included in sort order. <!---->
''Status:''
siptrans.cpp/h<br/>sip.cpp
      
      
=== Gatway: Call completion interworking on called side did not work ===
=== SIP: Trap during channel handling ===


{|
{|
Line 5,313: Line 5,502:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=56214 56214]
|[http://mantis.innovaphone.com/view.php?id=84800 84800]
|}
|}
Call completion on called side did not work<br/><br/>Thanks to Georg Hartwig for giving us his precious support during developent!<!---->
Rare trap when re-assigning channels.<br/><br/><!--<br/>sip.cpp/h<br/>medialib.h<br/>--><!---->
''Status:''
   
relay.cpp/h<br/>q950.cpp/h<br/>q931.cpp/h<br/>q931_nt.cpp<br/>q931_te.cpp<br/>nt_tbl.tbl<br/>te_tbl.tbl<br/>fty.cpp/h
= V8 Hotfix 30 (80811) =
Changes included in Version 8 hotfix30
[http://mantis.innovaphone.com/view.php?id=85034 Definition]
 
== New Features ==
 
 
   
== Bug Fixes ==
 
 
      
      
=== SIP Interworking: CGPN in display name of From URI ===
=== AD Replication: Configuration Buffer Increased ===


{|
{|
Line 5,326: Line 5,525:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=56504 56504]
|[http://mantis.innovaphone.com/view.php?id=86211 86211]
|}
|}
SIP Interworking: Get CGPN from display name of From URI<!---->
Was too small for many maps<br/><!--<br/>ldaprep.cpp/.h<br/>--><!---->
      
      
=== A DHCP client with  "/keep on" should send DISCOVER requesting the last assigned address after boot (not a REQUEST) ===
= V8 Hotfix 31 (80815 ) =
Changes included in Version 8 hotfix31
[http://mantis.innovaphone.com/view.php?id=86427 Definition]
 
== New Features ==


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56543 56543]
|}
In WLAN networks with more than one DHCP Server REQUESTing the last assigned address after boot needs more time to switch to a new server if the server providing this address has gone. <!---->
   
=== Configuration Option to keep Routes over a PPP interface always active ===


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56711 56711]
|}
To guarantee that certain connections are only established over a virtual private network, routes over a PPP interface need to be kept active in routing table even while the PPP interface is down. This is done now by checking<br/>  "Configuration/IP/PPP-Config/PPP<n>/Always keep Routes active"<br/>For enabled PPP interfaces which are not up the current routing state (active/skipped) is displayed in addition to the interface state under<br/>  "Configuration/IP/Routing" <br/><br/><!---->
      
      
== Bug Fixes ==
== Bug Fixes ==
Line 5,356: Line 5,541:


      
      
=== Gateway: Trap if Name Out or other fields with very long content ===
=== Gateway: #11 could not be dialed on analog interfaces with feature codes enabled ===


{|
{|
Line 5,363: Line 5,548:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55941 55941]
|[http://mantis.innovaphone.com/view.php?id=86819 86819]
|}
|}
A buffer overrun could happen if very long strings were used as input values<!---->
This is a featiure code used on DECT systems and it was not disabled on analog interfaces<br/><!--<br/>relayfty.cpp<br/>dtmffty.cpp<br/>--><!---->
''Status:''
gk.cpp
      
      
=== PBX: Unknown filter did not work anymore in version 8 ===
=== PBX: Trap if a Hold was attempted for a call without media ===


{|
{|
Line 5,376: Line 5,559:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55944 55944]
|[http://mantis.innovaphone.com/view.php?id=86874 86874]
|}
|}
The unknown filter could be configured, but was not applied to calls made by endpoints registered as unknown.<!---->
Could be caused by a misbehaving application or voip device<br/><!--<br/>pbx.cpp<br/>--><!---->
''Status:''
pbx.cpp
      
      
=== Firmware update failure on ip4001 ===
=== (clone of #80623) SIP: Calls may remain in clearing state ===


{|
{|
Line 5,389: Line 5,570:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=55981 55981]
|[http://mantis.innovaphone.com/view.php?id=88134 88134]
|}
|}
On the IP4001 the hwbuild string is computed using the boot flags to see if the box is in production mode. This causes a flash access conflict if the info screen is shown during a flash write ( firmware upload ). <!---->
SIP calls may remains undeleted.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
cpu.cpp cpu.h
      
      
=== Gateway: Overlap Dialing routes did not work as expected ===
= V8 Hotfix32 (80816.00) =
Changes included in Version 8 hotfix32
[http://mantis.innovaphone.com/view.php?id=89003 Definition]
 
== New Features ==
 
 
   
== Bug Fixes ==
 


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56006 56006]
|}
- sometimes '#' was added to the outgoing call even if 'Add #' was not configured<br/>- enbloc calls were terminated by a route with '.' as incomplete if not enough digits, even if matching routes followed<!---->
''Status:''
relay.cpp, gk.cpp
      
      
=== IP2x IP30x: Missing tones on BRI interface with SIP implementations that send RTP prior to coder negotiation ===
=== PBX: Potential trap when receiving unknown presence activity ===


{|
{|
Line 5,415: Line 5,593:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=56010 56010]
|[http://mantis.innovaphone.com/view.php?id=98043 98043]
|}
|}
This is the problematic scenario:<br/>The IP302 BRI interface is registered on a SIP proxy.<br/>An outgoing call is placed, the SIP proxy sends a STATUS 180 Ringing without SDP information. <br/>The remote side sends RTP data (with inband information) to the IP302.<br/>This switches off the IP302 generated tone, but the remote tone is cannot be used since the SDP is missing in the STATUS 180 message.<br/><br/>Now we ignore RTP with unknown coder for switching off the tone.<!---->
In the respective version unknown activities are mapped to "busy"<br/><!--<br/>fty.cpp - rollback of this change<br/>h450.cpp<br/>--><!---->
''Status:''
   
ac_dsp3.cpp
= V8 Hotfix33 =
Changes included in Version 8 hotfix33
[http://mantis.innovaphone.com/view.php?id=98530 Definition]
 
== New Features ==
 
 
   
== Bug Fixes ==
 
 
      
      
=== SIP: Switch to fax did not work in some cases ===
=== SIP: Wrong encoding of proprietary response header ===


{|
{|
Line 5,428: Line 5,616:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=56076 56076]
|[http://mantis.innovaphone.com/view.php?id=98235 98235]
|}
|}
Sometimes switch to audio occured immediately after switch to t.38<!---->
200/OK for REGISTER delivers endpoint's alias list.<br/>Encoded in proprietary response header "P-Alias".<br/>Encoding specifier was wrong.<br/><br/>Was:<br/>  P-Alias: 2,17,uranus%2Ck%FCmmel<br/>Must be:<br/>  P-Alias: 1,17,uranus%2Ck%FCmmel<br/><br/><!--<br/>sipmsg.cpp<br/>--><!---->
''Status:''
sip.cpp
      
      
=== Call Completion on Busy to diverted destination failed ===
=== SIP: SDP version not increased when answering an offer where only media-mode has changed ===


{|
{|
Line 5,441: Line 5,627:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=56243 56243]
|[http://mantis.innovaphone.com/view.php?id=98739 98739]
|}
|}
with the call rejection no informtion about the final destination (leg1 info) was sent, so the call completion was tried with the original called destination.<!---->
If remote side changes from 'sendrecv' to 'inactive'<br/>the SDP answer follows this change of media-mode,<br/>but SDP version was not increased.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
pbx.cpp
      
      
=== PBX: Multiple mobility destinations with delay not handled optimal ===
=== SIP: Do not add payload type 13 to media description for fax ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56302 56302]
|}
- if no local phone was registered, all mobility destinations were called right away. Now the destination with the shortes delay is called right away and the others later according difference in delay<br/><br/>- if local phone was busy the mobility destinations was only called after delay. The one with the shortes delay should be called first and then the others.<!---->
''Status:''
pbx_mobility.cpp, pbx_mobility.h, pbx.cpp, pbx_api.h
   
=== PBX: Groups could not be configured for objects with empty PBX setting ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56307 56307]
|}
Empty PBX setting means the object is handled as it has the local PBX set. So the local groups should be selectable<!---->
''Status:''
pbx_admin.cpp, pbx.cpp
   
=== Always allow local authentication in boot mode ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56396 56396]
|}
As Kerberos does not work in boot mode, the disable local authentication flag must be ignored there.<!---->
''Status:''
Files: command.cpp
   
=== SIP: Switch to t.38 was answered with audio instead of 488 reject ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56404 56404]
|}
In case t.38 is not enable, a switch to t.38 was not rejected with 488.<br/>SDP answer with currently active audio coder was send instead.<!---->
   
=== PBX: Errors when creating or changing Mobility objects were not displayed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56411 56411]
|}
If an error was detected (e.g. duplicate number) saving of the object was prohibited, but no error message as for other objects was displayed<!---->
''Status:''
pbx_edit_mobility.xsl
   
=== PBX-SOAP: Admin function could not be used to configure some new parameters ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56419 56419]
|}
like phone-config, description, ...<!---->
''Status:''
pbx.cpp, pbx.h
   
=== IP-DECT R-key handling for OEM protocol ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56469 56469]
|}
The R-key for an OEM protocol does not work.<!---->
   
=== Support for packetization up to 80ms ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56566 56566]
|}
60ms was the limit before<!---->
''Status:''
h323ch.cpp
   
=== IP-DECT FTY with TSIP and SIPS ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56580 56580]
|}
The feature codes do not work with TSIP, the local cf does not work with TSIP and SIPS.<!---->
   
=== IP-DECT: No Audio was received during call waiting ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56616 56616]
|}
This was another collateral damage from<br/><br/>fix: #55177: No Media event was generated even everything was normal for unanswered CC exec on IP-DECT<!---->
   
=== Changing the do-not-disturb user setting has no effect if do-not-disturb function key configured and present ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56743 56743]
|}
problem: Changing the do-not-disturb user setting has no effect if do-not-disturb function key configured and present<br/><br/>solution: fixed in code<br/><br/>files: phone/user/phone_user.cpp<br/><br/>products: all IPxxx telephones<br/><br/>risks: none<br/><br/><!---->
   
= V8 Hotfix 8 (80500.28) =
Changes included in Version 8 hotfix8
[http://mantis.innovaphone.com/view.php?id=56818 Definition]
== New Features ==
   
=== Gatway: Do not pass through SRTP key if "Enable SRTP" not activated ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=55767 55767]
|}
Pass through SRTP key only if "Enable SRTP" is activated<!---->
''Status:''
channel.h<br/>sip.cpp<br/>gk.cpp<br/>h323ch.cpp
   
=== PBX: Only 8 IP Filters possible, no indication if maximum reached ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56764 56764]
|}
Number increased to 32. If 32 Filters are configured no field to enter a new one is displayed<!---->
''Status:''
pbx.cpp, pbx.h, pbx_api.h, pbx_admin.cpp, pbx_global.xsl
   
=== PBX: Filters to even restrict registration with password ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56888 56888]
|}
The existing filters only restricted registration to the PBX without password. Now in addition to this registration with password can be restricted as well.<!---->
''Status:''
pbx.cpp, pbx.h, pbx_api.h, pbx_admin.cpp, pbx_global.xsl, pbx_admin_hdr.xml
   
=== DTMF facilities: new MWI modes for an OEM protocol ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56953 56953]
|}
New modes for message waiting indication added in the DTMF facility module. There are used for an OEM protocol in OEM IP-DECT devices.<!---->
   
=== SIP: Allow to receive messages larger than 2560 bytes ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57081 57081]
|}
There was a limitation for incoming SIP messages at 2560 bytes.<!---->
   
=== IP-DECT: anonymous login; master id checks/traces ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57104 57104]
|}
For anonymous handsets login additional master id checks and traces added.<!---->
   
=== make function keys on the phone-ui unmodifiable and unviewable ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57212 57212]
|}
problem: by setting a function key readonly mask (config change PHONE USER /funclock-ro-mask <mask> or web-ui: Phone->Protect->Function keys not modifiable on the phone-> <mask>), one can now determine a set of function key types which can only be set thru a web-ui and can only be viewed but not modified through phone-ui (see http://wiki.innovaphone.com/index.php?title=Howto:Disable_Function_Key_Modification_On_Phone_UI)<br/><br/>solution: fixed in code<br/><br/>files: phone/user/*<br/><br/>products: all telephones<br/><br/>risks: none<br/><!---->
   
=== SIP: Use registration's Contact-URI as Request-URI on calls to endpoints only ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57300 57300]
|}
Registered gateways get a Request-URI containing the destination number<!---->
   
=== Automated Kerberos configuration triggered by a special VAR ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57330 57330]
|}
A box can now be advised to join a Kerberos realm by writing an XML-Command to variable CMD0/KCMD.<!---->
''Status:''
command.h<br/>command.cpp<br/>command.xsl<br/><br/>http://wiki.innovaphone.com/index.php?title=Howto:How_to_configure_Kerberos_using_commands#Automated_Client_Configuration_.28V8_Hotfix8_and_later.29
   
=== IP-DECT: Kerberos configuration options for radio device configuration ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57339 57339]
|}
Now it is also possible to configure the Kerberos client if the radio device in discovery mode is configured by the master. The new feature #57330 is used.<!---->
   
=== IP-DECT: Messaging options and XML message type support ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57413 57413]
|}
New configuration page "DECT - Messaging" for the IP-DECT messaging alert signal options. The enable option replaces the IP Master option "Enable messaging to PBX".<br/>The XML message type is supported now. With XML messages it is possible to change the alert signal message dependent.<br/>The message priority can be considered if enabled: the SIP priority "emergency" changes the alert signal to alarm and the priority "non-urgent" changes it to silence.<!---->
   
=== Decoding of special XML entities ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57451 57451]
|}
Implement decoding of the following entities: &amp;lt; &amp;gt; &amp;quot; &amp;apos; &amp;amp;<!---->
''Status:''
files: xml.cpp
   
=== IP-DECT: log messages for MSF calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57512 57512]
|}
Log messages for MSF calls added.<!---->
   
=== IP-DECT: MSF module option disable ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57560 57560]
|}
With the option /disable it is possible to disable the DECT MSF module.<!---->
   
=== VM, URL parameter "$_noctl=true" allows to reject control-calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57571 57571]
|}
Control calls may reach a VM object unintentionally. Such calls can now be rejected.<!---->
   
=== Gateway: If Moh Mode is configured set 'exclusive coder' checkmark as well on UI ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57654 57654]
|}
The MOH Mode implies that exclusive coders are used<!---->
''Status:''
relay_edit_phys.xsl
   
=== Phone: Show presence note on 'partner' fkey label ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57687 57687]
|}
Show presence note (if availbale) on 'partner' fkey label.<br/>If no text note is avalable, activity is shown (as usual).<!---->
   
=== update service 'provision' option to request earlier and faster polling in provisioning mode ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57799 57799]
|}
In provisioning mode the update service should start polling the update server as soon as possible and not use the default delay.<br/>This can be configured now by<br/><br/>    config add UP1 /provision <n><br/><br/><n> defines the delay in seconds of the first poll, subsequent polls start after (previous delay * 2) seconds. The maximum delay between polls is 60 seconds.<br/><br/>    config add UP1 /provision 0<br/>or<br/>    config rem UP1 /provision<br/><br/>switches back to the default or the configured polling interval<br/><br/><!---->
   
== Bug Fixes ==
   
=== PBX: Checking if a call matches an pending call-completion request was wrong ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56706 56706]
|}
If a call completion is pending and the user calls the destination with the pending CC or the user retries successfully the call independent of the pending CC, we want to avoid to signal this CC. For this we match any calls to pending CCs. Sometimes this resulted in matches even if there was none and pending CCs were cleared which shouldn't<!---->
''Status:''
pbx.cpp
   
=== PBX: Trap if duplicate "Long Name" in Database ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56774 56774]
|}
It may happen that on a replicated PBX temporarily multiple objects with the same Long Name (cn) exist. In the case the PBX restarted.<!---->
''Status:''
pbx.cpp
   
=== PBX: CFNR configured at Waiting not executed correctly on transfer to Waiting ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56775 56775]
|}
under some circumstances not executed at all and sometime without waiting for No Response Timeout<!---->
''Status:''
pbx.cpp
   
=== Gatway: Suspend/Resume on call completion interworking ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56827 56827]
|}
Suspend/Resume signaling on call completion interworking did not interwork<!---->
   
=== PBX Mobility: Trap if call to mobile phone scheduled for recall is cleared and SOAP monitoring is on ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56847 56847]
|}
If call is put on hold by the mobile phone and then the mobile phone hangs up, the PBX tries to recall the mobile phone. If the held party hangs up in this situation with SOAP monitoring of the mobile phone active, a trap happens<!---->
''Status:''
pbx_mobility.cpp
   
=== Trap on call completion with mobility over dtmf object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56882 56882]
|}
When using call completion with mobility over the dtmf object, the PBX crashed.<br/>Now call completion over mobility is rejected.<br/><!---->
   
=== Disconnect from DTMF/ICP/Directory search object didn't work with mobility ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56883 56883]
|}
The disconnect from the DTMF, ICP and Directory search objects didn't work with mobility, as it was wrongly called.<!---->
   
=== PBX Mobility: CLIR did not work correctly ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56899 56899]
|}
A call was sent without number, but it should have been sent with Number Presentation restricted option set.<!---->
''Status:''
ep_lib.cpp
   
=== SIP: Keep ringing calls longer than 3 min ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56901 56901]
|}
An INVITE client transaction was canceled 180 secs after "180 Ringing" have been received.<br/><!---->
   
=== IP-DECT: Load sharing for trunks (OEM protocol) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=56942 56942]
|}
Load sharing for trunks does not work. It is used for an OEM protocol.<!---->
   
=== Trap: When handling call completion request from ISDN ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57113 57113]
|}
Trap: When handling call completion request from ISDN<!---->
''Status:''
relay.cpp<br/>q931.cpp<br/>pppif.cpp<br/>signal.cpp/h
   
=== Qsig Leg2 Info decoding could fail ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57126 57126]
|}
Qsig Leg2 Info decoding could fail<!---->
   
=== Protect TLS socket against collision of SOCKET_RECV and SOCKET_SHUTDOWN ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57130 57130]
|}
It was possible that a collision of SOCKET_RECV from the application and SOCKET_SHUTDOWN from the TLS socket occured. This could lead to a trap because the application was already deleted when the SOCKET_RECV_RESULT was sent.<!---->
''Status:''
tls.cpp
   
=== Missing "Recall possible" text in status line ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57196 57196]
|}
problem: Missing "Recall possible" text in status line <br/><br/>solution: fixed in call<br/><br/>files: phone/app/app_cc.cpp [box/phone]/forms/[lcd/]forms_gen.cpp<br/><br/>products: all telephones<br/><br/>risks: none<br/><br/> <br/><!---->
   
=== PBX: Call from mobile endpoint could not be picked up with DTMF group pickup ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57204 57204]
|}
pickup was rejeceted<!---->
''Status:''
pbx.cpp
   
=== v9 Replication Compliance ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57274 57274]
|}
Fixes addressing UTF-8 conversions<!---->
   
=== SIP: Some interop tweaks did not work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57354 57354]
|}
Some module options did not work after reboot:<br/> /no-hr-notify<br/> /prefer-pai<br/><!---->
   
=== SIP: Fix for video calls through broadcast user ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57504 57504]
|}
When initiating a video call towards broadcast user, an offer/offer collision may occur in the PBX.<br/>The PBX must select the video coder (not only audio coder) in this case.<!---->
   
=== IP-DECT: Pickup, caller id update ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57509 57509]
|}
Fix for the caller id display update after call pickup.<!---->
   
=== SIP: Decoding of special Contact-URIs ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57523 57523]
|}
sip:2031;phone-context=cdp.udp@dpp.nortel:5070;maddr=47.166.92.207;transport=udp<br/>The port information was not extracted from phone-context parameter.<br/>Format used by Nortel only.<!---->
   
=== SIP: SDP attribut annexb=no was missing ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57533 57533]
|}
If G.729 Annex B was disabled it must be explicitely announced,<br/>because no mentioning annexb is interpreted as annexb=yes.<br/><!---->
   
=== Tones: Ringback cadence for Ireland not correct ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57545 57545]
|}
Ringing tone - Ireland<br/>Freq: 400+450<br/>Cadence: 0.4 on 0.2 off 0.4 on 2.0 off<br/><br/><!---->
   
=== PBX: Trap when handling presence subscription for VM object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57578 57578]
|}
Trap when handling presence subscription for VM object<!---->
''Status:''
pbx.cpp
   
=== Allow dtmf features park/unpark for calls from voicemail object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57582 57582]
|}
Currently, calls from the voicemail object to the dtmf object were cancelled, as all calls from non user objects have been cancelled.<br/>Now, the features park and unpark are allowed.<!---->
   
=== SNMP, ifSpeed wrong ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57610 57610]
|}
SNMP, ifSpeed wrong<!---->
   
=== IP-DECT: MSF CLMS messages ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57612 57612]
|}
Now CLMS messages can be sent with the MSF module.<!---->
   
=== VM: trailing '#' in CDPN let's diverted call to VM fail ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57649 57649]
|}
VM: trailing '#' in CDPN let's diverted call to VM fail<!---->
   
=== Filter did not work correctly with local objects and overlap sending ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57652 57652]
|}
For checking the filter in case of overlap sending, the number including the Node prefix was used regardless if the node prefix was dialed or not.<!---->
   
=== automatic or manual recording cannot be stopped if the recorded call is not the currently active call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57685 57685]
|}
Automatic or manual recording could not be stopped if the recorded call was not the currently active call.<br/>If the Redial-key is used to toggle recording this is intended behaviour because otherwise the Redial-key could not be used to transfer the non-recorded active call.<br/>If a 'Recording' function key is used to toggle recording there is no need for this restriction.<br/><br/>Now a 'Recording' function key stops automatically or manual started recording any case. <!---->
   
= V8 Hotfix 9 (80500.32)  =
Changes included in Version 8 hotfix9
[http://mantis.innovaphone.com/view.php?id=57750 Definition]
== New Features ==
   
=== SIP: Suppress Annex B of G.729 if "Silence Compression" is not enabled at the interface ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57540 57540]
|}
Suppress Annex B of G.729 if "Silence Compression" is not enabled at the interface<!---->
   
=== permit to send log messages, alarms and events via HTTPS with and without checking the server certificate ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57785 57785]
|}
Both for the log server and for the alarm/event forward server HTTPS can be configured now.<br/>But because distribution of certifcates a may be problematic if there is a big number of clients checking the server certificate can be supressed by<br/><br/>  config add LOG0 /tls-unchecked<br/><!---->
   
=== IP-DECT: OEM device GUI ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57993 57993]
|}
Some little changes for a DECT OEM device for the GUI.<!---->
   
=== IP-DECT: TONE interface ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58041 58041]
|}
The tone inferface is added to the IP1200.<!---->
   
=== product_id 153,154 added ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58122 58122]
|}
these new IDs are needed for IP152 based phone versions <!---->
   
=== PBX dtmf group feature marks dynamic in groups ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58536 58536]
|}
As the PBX dtmf group feature shows all dynamic in and out groups, the displayed name of dynamic in groups will be preceeded with '* ' now.<!---->
   
=== SIP: Mapping of "403 Forbidden" into "Q.931 Requested circuit/channel not available" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58635 58635]
|}
Previously mapped into "Q.931 Call rejected"<br/>Better mapped into "Q.931 Requested circuit/channel not available" in order to trigger re-routing at the Gateway<!---->
   
=== SIP: Support of P-Called-Party-ID ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58748 58748]
|}
Get CDPN of incoming SIP calls from P-Called-Party-ID if present.<br/><!---->
   
=== 30s Timeout for dialing too short ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58783 58783]
|}
When putting someone on hold with 'R' there was a timeout of 30s until the consultation call was terminated. This could be too short to find the one to whom to transfer the call.<br/><br/>The protocol timeout in H.323 (TO302) was increased from 30s to 120s<!---->
''Status:''
h323sig.cpp
   
=== PBX: Don't apply Send Number to Recording calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58878 58878]
|}
For recording it is usually needed to know the real number<!---->
''Status:''
pbx.cpp
   
=== MWI key with configurable DTMF signaling type for message center calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58980 58980]
|}
Some users must force inband DTMF for certain SIP providers but our Voice Mail requires out of band DTMF signaling.<br/>Now the type of DTMF signaling to be used for calls to the message center can be configured at the MWI key.  <!---->
   
=== phone: disable call intrusion via partner key when recording is active ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65918 65918]
|}
Call intrusion cannot be performed while recording is active:<br/>- recording establishes a 3party conference between local party, remote party and recorder.<br/>- call intrusion establishes a 3party conference between local party and the two remote parties<br/>- recording and call intrusion at the same time would require a 4party conference which cannot be set up because the phone has only 2 DSP coder channels.<br/><br/>Now if any kind of recording is configured call intrusion is neither offered in 'recall' menu nor performed via partner key.<!---->
   
== Bug Fixes ==
   
=== Disabling local authentication also turned off module authentication ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57863 57863]
|}
When Kerberos was configured on a box and the local admin accounts were disabled, logging and PBX administration using PBX users did not work anymore.<!---->
''Status:''
files: command.cpp
   
=== SIP: Transfer handling at Gateway may cause on-way-audio ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57906 57906]
|}
I some scenarios where REFER is handled at the Gateway to transfer a local media call leg (e.g. ISDN) to any other call leg.<!---->
   
=== IP-DECT: no digits en-bloc timeout ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=57925 57925]
|}
The timeout of the en-bloc timer is changed for the case that no digits are dialed. This fixes the Aastra PBX block bug.<!---->
   
=== Resuming TLS sessions did not work correctly ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58013 58013]
|}
The server now ensures that session IDs are unique by adding a timestamp and a serial number. This increases the size of session IDs from 16 bytes to 24 bytes.<br/><br/>Also IP addresses were not handled correctly by the session cache.<!---->
''Status:''
tls.cpp
   
=== QSIG Call Complettion to MD110 failed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58372 58372]
|}
QSIG Call Complettion to MD110 failed<!---->
   
=== phone directory collating sort  order unexpected ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58386 58386]
|}
The ordinal of the space character was higher than that of any alphameric character, thus for example "Smith Eric" was displayed behind "Smithson Eric".<br/>The ordinal of the space character is now 0.  <!---->
   
=== SIP: Don't send empty P-Asserted-Identity in provisional response ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58493 58493]
|}
SIP/2.0 183 Session Progress<br/>Via: SIP/2.0/TCP 10.64.32.2:14937;branch=z9hG4bK6728a259<br/>From: ""<sip:850@10exchange.wschneider.com;user=phone>;epid=123A3A4D16;tag=c755636afc<br/>To: <sip:00763773033@10.64.64.1;user=phone>;tag=3908677425<br/>Call-ID: d0248a8c-a324-454b-807a-923c30c1e24b<br/>CSeq: 34 INVITE<br/>Contact: <sip:00763773033@10.64.64.1:5060;user=phone;transport=TCP><br/>Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH<br/>Content-Length: 230<br/>Content-Type: application/sdp<br/>Server: (innovaphone IP800/8.00 dvl [tac-1.11108:/8050028/400])<br/>Supported: replaces,privacy,answermode,from-change,100rel,timer,histinfo<br/>P-Asserted-Identity: <br/>P-Sig-Options: Sending-Complete<br/><!---->
   
=== Invalid duplicate DTMF object caused the PBX to trap ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58514 58514]
|}
A false config with an invalid DTMF object (name like DTMF#pickup_group) caused the PBX to trap.<br/>Such an object will be ignored now.<!---->
   
=== Pickup function key display discards leading letter on transferred call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58520 58520]
|}
problem: Pickup function key display discards leading letter on transferred call, so the first letter or number of the calling party is always missing <br/><br/>solution: fixed in code<br/><br/>files: phone/app_disp.cpp<br/><br/>products: all telephones<br/><br/>risks: none<br/><br/><br/><!---->
   
=== trap on late CHANNEL_INIT to relay_media_relay::serial_event() ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58524 58524]
|}
A null pointer was referenced when a CHANNEL_INIT was passed to an object in closing state  <br/><br/><!---->
   
=== AD-replicator: xml-show-namingcontexts leaks memory ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58564 58564]
|}
a memory leak occurred every time when clicked on Configuration/LDAP/Replicator(AD)/DN/"Show Options"<br/><!---->
   
=== Do not disconnect calls to directory search object from master/slave user ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58587 58587]
|}
Calls from a master/slave user where disconnected by the directory search object.<br/>These calls are allowed now.<!---->
   
=== Phone: Light up partner fkey even on active state ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58589 58589]
|}
While the phone itself is in active state (non-idle) a partner fkey lamp did not light up when partner's presence indicate 'on-the-phone' activity.<br/>Only in idle state the lamp indicated that partner is 'on-the-phone'.<!---->
   
=== SIP: Dialog-Info did not show "confirmed" state ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58594 58594]
|}
"proceeding" was indicated instead.<br/>Caused Problems on snom phones.<!---->
   
=== Soap::UserPickup() sometimes didn't work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58665 58665]
|}
Soap::UserPickup() sometimes didn't work<!---->
   
=== Call Intrusion across PBXs did not work (intrude call at slave from master) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58710 58710]
|}
There was a fix already for this, but this covered only intrude at master from slave.<!---->
''Status:''
pbx.cpp<br/>pbx.h
   
=== Gateway Routes with CDPN map to number containing '#' did not work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58737 58737]
|}
The number starting with the '#' was omitted.<br/><br/>Collateral damage of fix: #56006: Gateway: Overlap Dialing routes did not work as expected<!---->
''Status:''
gk.cpp
   
=== PBX Trunk Object: Incomplete destination did not work for incoming incomplete enblock calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58755 58755]
|}
collateral damage of fix: #54357: PBX Node 'incomplete Number' destination did not work for block dial calls<!---->
''Status:''
pbx.cpp
   
=== DRAM /Firmware upload stops sometimes ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58769 58769]
|}
Depending on the timing the upload hangs.<br/>Seen with the innovaphone test program and minifirmware<!---->
''Status:''
servlet_post_file.cpp
   
=== Gateway: Trap on early RELEASE from calling side ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58780 58780]
|}
If the caller stops calling at an early stage, a trap may occur:<br/><br/>0:0806:591:0 - LOG CALL 15 Alloc<br/>0:0806:591:3 - LOG CALL 15 A:Call    ->                        / PRI2::->*::<br/>0:0806:597:0 - LOG CALL 15 B:Call    100->226                  / PRI2:5336100:->RP2:226:<br/>0:0806:701:3 - LOG CALL 15 A:Rel    100->226                  / PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry<br/>0:0806:712:3 - LOG CALL 15 Media    100->226                  G711A,20(0,0,0)/G711A,20(0,0,0) PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry<br/>0:0806:713:7 - LOG CALL 15 B:Alert  100->226                  G711A,20(0,0,0)/G711A,20(0,0,0) PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry<br/>0:0806:714:0 - TRAP: 0x10<br/><!---->
   
=== PBX: Name Identification was not forwarded with forked call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58786 58786]
|}
With call forking the original calling name id was not forwarded<!---->
''Status:''
pbx.cpp
   
=== PBX: Trap if 'Escape dialtone from' is configured to a non-existent object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58789 58789]
|}
Check implemented to use internal TONE interface in this case<!---->
''Status:''
pbx.cpp
   
=== SIP: re-INVITE without SDP offer was rejected with 504 Server Timeout in 'held' state ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58822 58822]
|}
re-INVITE without SDP offer was rejected with 504 Server Timeout if received on an inactive session.<br/><!---->
   
=== SIP: Handling of reject of re-INVITE without SDP offer was incomplete ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58824 58824]
|}
Handling of reject of re-INVITE without SDP offer was incomplete.<br/>Need to generate dummy offer for app.<!---->
   
=== send PROGRESS after CALL-PROC to stop 10s T310 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58839 58839]
|}
sometimes too short to forward a call<!---->
   
=== IP-DECT: potential trap ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58920 58920]
|}
Potential trap in DECT devices fixed.<br/>Trap identification:<br/>XCPT: no 2 (TLB load)  pc 94273278  ra 94273254  va 0000000c<br/><!---->
   
=== Gateway: A call counter with name containing blank or other special character created problems ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58944 58944]
|}
It could be configured, but if another map was added to the same route the config was corrupted<!---->
''Status:''
gk.cpp
   
=== Trap on CF remove while files are deleted ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58984 58984]
|}
When files are deleted from the CF card and the card is removed or has an error, the box could trap.<!---->
   
=== Potential trap if routes with DTMF output combined with pause chars (',') are used for calls without channel or out-of-channels ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59012 59012]
|}
In this situation pause digits are passed to a channel, which does not exits. This causes the trap.<br/>Could also be dialed pause characters on a call-independent signaling.<!---->
''Status:''
relay.cpp
   
= V8 Hotfix10 (80500.33) =
Changes included in Version 8 hotfix10
[http://mantis.innovaphone.com/view.php?id=59505 Definition]
== New Features ==
   
=== Call Forwarding Function Key with "Apply 'Always' Setting Only" checkmark  (CFU  Only) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59077 59077]
|}
If "Apply 'Always' Setting Only" is checked the Function key toggles onls over the 'Always' (i.e. CFU) entries and keeps other existing diversions untouched.<br/>Thus CFB or CFNR diversions set at the phone or at the PBX are not changed when toggling this key.<!---->
   
=== SIP: Registration lookup by attribute 'username' of Authorization header ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59078 59078]
|}
Registration lookup by attribute 'username' of Authorization header (not only on anonymized calls)<!---->
   
=== x509: Support for DNS names in SubjectAltName extension of certificates ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59171 59171]
|}
Create self-signed certificates and certificate requests that contain a DNS name in the SubjectAltName extension. Display the DNS name in the certificate details.<!---->
''Status:''
Files: x509.cpp, x509.h, x509asn1.h, request.xsl, certificate_create.xsl, certificate.xsl, oids_asn1.h
   
=== SIP: Support for another Contact-URI parameter in REGISTER ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59174 59174]
|}
+u.sip!model.ccm.cisco.com<!---->
   
=== SIP: Interop feature "X-cisco-srtp-fallback" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59198 59198]
|}
Required for SRTP sessions<!---->
   
=== H.323-Q.931-Interworking - display text provided in the Display Information Element of an ISDN Information Message on phone ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59506 59506]
|}
The text provided in the Display Information Element of an ISDN Information Message was silently discarded. Now it is displayed in the phone status line.  <br/><!---->
   
=== SIP: Interop feature "X-cisco-sis-3.0.0" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59533 59533]
|}
Required for SRTP sessions<!---->
   
=== Debug: Support to identify bad objects ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59714 59714]
|}
Only mem-clients are allowed be deleted dynamically.<!---->
   
== Bug Fixes ==
   
=== H.323 Remote address was not checked for calls coming in on special trunks with non-standard ports ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58958 58958]
|}
This is no problem which affects innovaphone standard products. It is only for H.323 trunks configured with fixed remote and local address and port.<!---->
''Status:''
h323sig.cpp
   
=== Interworked Control-Calls without Facilities Shall Stop in Relay ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59009 59009]
|}
Interworked Control-Calls without Facilities Shall Stop in Relay<!---->
   
=== PBX Exec Object: A number Map object to be used to call exec directly ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59066 59066]
|}
A number map can be put in exec secretary or direct call groups to call the exec thru this Map Object directly. This did not work for calls from IP Phones, which sent a source name with the call.<!---->
''Status:''
pbx_exec.cpp
   
=== PBX: Trap if using SOAP Version Method if PBX not started ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59071 59071]
|}
null pointer access happens in this case<!---->
''Status:''
pbx_xml.cpp
   
=== DHCP client: "Wait for selected Server" timeout was not applied after a DHCP restart ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59076 59076]
|}
When the DHCP client receives a DHCP restart request a timer is setup to trigger the restart. The failure happens when an offer arrives before this timer fires.<!---->
   
=== SIP: Media negotiation problem when processing INVITE without SDP ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59082 59082]
|}
Media negotiation problem when processing INVITE without SDP<!---->
   
=== H.323: Don't send a call-independent-signaling call without facilities ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59088 59088]
|}
This could happen if a QSIG call was interworked, with facilities we do not support<!---->
''Status:''
h323_tbl.tbl<br/>h323sig.cpp<br/>h323sig.h<br/>phonesig.cpp
   
=== send PROGRESS after CALL-PROC to stop 10s T310 - in ISDN Stack not PBX ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59195 59195]
|}
sending PROGRESS in the PBX could have some unwanted side effects, like a Cisco Callmanager believing that there is actual in-band media available<!---->
''Status:''
pbx.cpp<br/>q931.cpp<br/>q931.h<br/>te_tbl.tbl<br/>nt_tbl.tbl<br/>isdn_interop.xsl
   
=== H.323 slowstart avoid sending duplicate TerminalCapabilitySet messages ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59203 59203]
|}
If a media re-negotiation happened on a remote system at a time the local H.245 channel was not even established, it could happen that a sequence of TCS, TCS0 and TCS again was sent to a calling system. This irritated especially a Cisco Call Manager.<br/><br/>This happened for example, if a call was received from the call manager on one PBX, which was routed to another PBX on which a CFNR was configured.<!---->
''Status:''
h323ch.cpp
   
=== H.323 Slowstart media renegotiation did not work if TCS was not yet received ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59248 59248]
|}
This caused a CFNR not being executed (call was cleared on the original called endpoint, but was not sent to new destination) for calls from Cisco Call Manager<!---->
''Status:''
h323ch.cpp
   
=== PBX Mobility: Filters were not evaluated for mobility calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59398 59398]
|}
Calls from mobile phones thru the mobility object were not affected by filter configurations for the user<!---->
''Status:''
pbx_mobility.cpp
   
=== SNMP, ifSpeed wrong ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59504 59504]
|}
SNMP, ifSpeed wrong<!---->
   
=== SIP: Media negotiation problem in some early media scenarios ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59711 59711]
|}
SIP/H323 interworking problem.<br/>Call was terminated with "504 Server Time-out" and "Recovery on timer expiry (102)"<!---->
''Status:''
sip.cpp
   
=== Phone IP150 - dialling numbers containing asterisks '*' does not work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59768 59768]
|}
if in offhook mode the asterisk key is pressed for a short time the key is ignored, if it is pressed longer it is evaluated as mute key.<!---->
   
=== SIP: Registration refresh interval not parsed from REGISTER response if behind NAT ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59826 59826]
|}
Registration refresh interval not parsed from REGISTER response if behind NAT.<br/>Wrong handling of 'received' and 'rport' parameters in Via header (RFC-3581).<!---->
''Status:''
REGISTER sip:talk.arcstel.netpbx5.net SIP/2.0<br/>Proxy-Authorization: Digest username="1295_1",realm="talk.arcstel.netpbx5.net",nonce="12935856813:4d1dfa2cd75027df50e51d433f90d3a6",response="09e7e72f21d1772b12b73dffb5b51e3c",uri="sip:talk.arcstel.netpbx5.net",qop=auth,cnonce="b35c9f24e909d311",nc=00000001,algorithm=MD5<br/>Via: SIP/2.0/UDP 192.168.0.34:2057;branch=z9hG4bK-E9764661;rport<br/>From: <sip:1295_1@talk.arcstel.netpbx5.net>;epid=00013e01b12b;tag=847121008<br/>To: <sip:1295_1@talk.arcstel.netpbx5.net><br/>Call-ID: fc72cde0e909d3119b2500013e01b12b@192.168.0.34<br/>CSeq: 1001 REGISTER<br/>Contact: <sip:1295_1@192.168.0.34:2057;transport=UDP>;expires=3600<br/>Content-Length: 0<br/>Expires: 3600<br/>Max-Forwards: 70<br/>User-Agent: (Ascom IP-DECT Base Station/ [4.1.24/4.1.24/IPBS1-A3/4C])<br/>Allow-Events: reg,dialog,message-summary,presence<br/><br/>SIP/2.0 200 OK<br/>Via: SIP/2.0/UDP 192.168.0.34:2057;received=89.233.254.81;branch=z9hG4bK-E9764661;rport=58537<br/>From: <sip:1295_1@talk.arcstel.netpbx5.net>;epid=00013e01b12b;tag=847121008<br/>To: <sip:1295_1@talk.arcstel.netpbx5.net>;tag=5439c50a<br/>Call-ID: fc72cde0e909d3119b2500013e01b12b@192.168.0.34<br/>CSeq: 1001 REGISTER<br/>Contact: <sip:1295_1@192.168.0.34:2057;transport=UDP>;expires=54<br/>User-Agent: Advoco/5.0.3046<br/>Content-Length: 0
   
=== PBX: Call was possible from registration as standby PBX ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59844 59844]
|}
A standby PBX registers at the active PBX to check if it is alive. This registration could be misused for calls. It could be done with H.323 and SIP. This fix prohibits calls from this registration and allows registration with H.323 only<!---->
''Status:''
pbx.cpp
   
=== Phone - switch off microphone while sending DTMF as voice data, increase volume of DTMF tones sent as voice data  ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59846 59846]
|}
When "Registration x/General/No DTMF Detection" is checked DTMF tones are sent as voice data. Detection of such tones at the receiving side may fail when mixed with microphone input.  <!---->
   
=== PBX CDR records with a size  near 1kB or larger were garbled when sent via HTTP ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59966 59966]
|}
PBX CDR records with a size near 1kB or larger were garbled when sent via HTTP because of an encoding bug. Locally logging worked correct.    <!---->
   
=== Phone: Translation for presence activities ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60119 60119]
|}
  Abwesend, Beschäftigt, Mittagessen, Besprechung, Urlaub<br/>  Away, Busy, Lunch, Meeting, Vacation<br/>  Parti, Occupé, Déjeuner, Réunion, Vacances<br/>  Assente, Occupato, Pranzo, Riunione, Ferie<br/>  Ausente, Ocupado, Comida, Reunión, Vacaciones <br/>  Fravær, Opptatt, Lunsj, Møte, Ferie<br/><!---->
   
=== PBX: Trap if a call from mobile endpoint was diverted to a waiting queue, with altert Timeout ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60161 60161]
|}
A NULL pointer access happend in this case while sending the ALERT message<!---->
''Status:''
pbx_wait.cpp
   
= V8 Hotfix11 (80500.34) =
Changes included in Version 8 hotfix11
[http://mantis.innovaphone.com/view.php?id=60189 Definition]
== New Features ==
   
=== SIP: Interop flag for Avaya: /no-t38-in-initial-offer ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59176 59176]
|}
config change SIP /no-t38-in-initial-offer<br/>Can be used to suppress T.38 capability indication in initial SDP offer.<br/>A switch to T.38 fax mode may follow, if T.38 is enabled at the interface.<!---->
   
=== SIP: Add PAI/PPI header to 200/Ok for INVITE ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60249 60249]
|}
Some SIP servers wants us to send P-Asserted-Identity/P-Preferred-Identity header in final INVITE response.<!---->
   
=== IP-DECT: number map for incoming calls (OEM) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60294 60294]
|}
Number map for incoming calls added for OEM devices.<!---->
   
=== SIP: Add PAI/PPI header to 181 response for INVITE ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60438 60438]
|}
To get full identity information of the new remote partner<!---->
   
=== SIP: Module option /share-local-port ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60542 60542]
|}
This option forces outbound TCP signaling connection to be bound to the same local port as the signaling interface is listening on.<br/>(In order to make the remote peer do connection reuse)<!---->
   
== Bug Fixes ==
   
=== SIP: Handling of re-INVITE w/o SDP offer while in held (inactive) state ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60296 60296]
|}
A re-INVITE w/o SDP offer while in held (inactive) state must be answered with 200/Ok containing an sendrecv offer (not inactive).<!---->
   
=== SIP: SRTP re-negotiation failed sometimes ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60387 60387]
|}
After switching to non-encrypted media (MOH) the re-negotiation for encrypted media failed (on CCM).<!---->
   
=== PBX: Slave license update period too short ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60390 60390]
|}
was 100s (v8) or 10s (v7) should be 10min<!---->
''Status:''
pbx.h
   
=== Gateway: Trap on early RELEASE from calling side ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60400 60400]
|}
Trap when Notification Indicator is received with ALERT while peer call is released already.<!---->
   
=== IP-DECT: potential trap ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60406 60406]
|}
Some pointer checks are added to prevent traps.<!---->
   
=== PBX Waiting object: Problem with announcements from Boolean Object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60421 60421]
|}
The announcement worked, but if DTMF dialing to another Waiting object was done, DTMF dialing on this second Waiting object did not work anymore.<!---->
''Status:''
pbx.cpp<br/>pbx_api.h<br/>pbx_wait.cpp
   
=== PBX CF Loop detection indicated loop with CFNR even if there was no loop ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60427 60427]
|}
A CFNR loop is only detected if the CFNRs are executed because of no registration. The loop was detected with a single Object without registration instead of only detecting the loop if all objects are without registration<!---->
''Status:''
pbx.cpp
   
=== H.323: If INFO was sent with cdpn and kp it could happen that it was forwarded with cdpn in SETUP and kp in INFO ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60443 60443]
|}
If a call was established by the application (or incoming signaling) without dialing information and then before the TCP connection was established a INFO message was sent with keypad and called-party-number, the call (SETUP) was sent with the called-party-number followed by an INFO with keypad.<br/><br/>This could result in a duplication of the dialed digits.<br/><br/>Only in special OEM scenarios, because keypad is usually not used.<!---->
''Status:''
h323_tbl.h
   
=== editing function keys via WEB interface broken after invalid characters have been entered in an e164 number field ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60468 60468]
|}
xml syntax characters like < > &amp; entered in a number field were not encoded on output and thus garbled the xml structure    <!---->
   
=== Memory leak when configuring H.323 NAT ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60474 60474]
|}
Memory leak when configuring H.323 NAT<!---->
   
=== possible trap with enabled trace flag on CF checkdisc ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60513 60513]
|}
The box could trap while checking the card, if the trace flag for CF0 was enabled.<!---->
   
=== PBX/SOAP: Potential trap when disconnecting a mobility call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60538 60538]
|}
If a SOAP application (e.g. TAPI) disconnects a call to/from a mobile user, a trap could happen<!---->
''Status:''
pbx_xml.cpp
   
=== PBX DECT System object: DECT parameters got lost, when changing critical flag ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60565 60565]
|}
The object was written back to flash without the parameters stored by the DECT system<!---->
''Status:''
pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_dect.cpp<br/>pbx_dect.h
   
=== PBX SOAP Admin: Critical flag could not be set in object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60568 60568]
|}
The attribute "critical" was not allowed<!---->
''Status:''
pbx.cpp
   
=== Ldap Replication, Problems with Percent-Char in Password ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60611 60611]
|}
Ldap Replication, Problems with Percent-Char in Password<!---->
   
=== Optional display of text provided in the Display Information Element of an ISDN Information Message ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60612 60612]
|}
The text provided in the Display Information Element of an ISDN Information Message is displayed at the phone status line.<br/>This may be supressed now by checking "Phone/Preferences/Hide Display Info from ISDN Providers" <br/><!---->
   
=== SIP: Authentication issue (AVAYA-SM interworking) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60712 60712]
|}
Another re-try with authentication required.<!---->
   
=== Group Indication with a diverting number of zero length caused a encoding error ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60715 60715]
|}
The number should not be sent at all. This happend if a group indication was to be sent from a call which was diverted by an object without number<!---->
''Status:''
h450.cpp
   
=== PBX Waiting: Don't forward DTMF to announcement source ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60838 60838]
|}
Announcement source could be a boolean object and DTMF could change the state of the boolean<!---->
''Status:''
pbx_wait.cpp
   
=== IP-DECT: cause code changed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60958 60958]
|}
The cause code is changed to "cause unassigned number" if the call is released because no radios are available.<!---->
   
=== Fix for SIP requests with 10+ header instances ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61014 61014]
|}
Response to following INVITE request did not returned all Via headers:<br/><br/>INVITE sip:229@192.168.193.181:2058;transport=UDP SIP/2.0<br/>Record-Route: <sip:145bf82@192.168.193.210;transport=udp;lr><br/>Record-Route: <sip:192.168.193.219:15060;lr;sap=433098584*1*016asm-callprocessing.sar-624908352~1296718381566~-535462628~1><br/>From: "H323-2" ;tag=8084387dbc40e01d7f4d42da8200<br/>To: <sip:229@localdomain.com><br/>Call-ID: 8084387dbc40e01d8f4d42da8200<br/>CSeq: 1 INVITE<br/>Via: SIP/2.0/UDP 192.168.193.210;rport;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099903-AP;ft=192.168.193.210~13c4<br/>Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099903<br/>Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199901<br/>Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199900<br/>Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bK8084387dbc40e01d7f4d42da8200-AP;ft=6565<br/>Via: SIP/2.0/TLS 192.168.193.104;branch=z9hG4bK8084387dbc40e01d7f4d42da8200;avaya-cm-term-reaction=shortcut<br/>Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099899-AP;ft=7355<br/>Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099899<br/>Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199897<br/>Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199896<br/>Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bK8084387dbc40e01d9f4d42da8200-AP;ft=6565<br/>Via: SIP/2.0/TLS 192.168.193.104;branch=z9hG4bK8084387dbc40e01d9f4d42da8200<br/>Supported: 100rel,histinfo,join,replaces,sdp-anat,timer<br/>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH<br/>User-Agent: Avaya CM/R016x.00.1.510.1 AVAYA-SM-6.1.0.0.610012<br/>Contact: "H323-2" <sip:201@192.168.193.104:5061;transport=tls><br/>Accept-Language: en<br/>Accept-Contact: *;+avaya-cm-line=1<br/>Alert-Info: <cid:internal@localdomain.com>;avaya-cm-alert-type=internal<br/>History-Info: <sip:229@localdomain.com>;index=1<br/>History-Info: "229" <sip:229@localdomain.com>;index=1.1<br/>Min-SE: 1200<br/>P-Asserted-Identity: "H323-2" <sip:201@localdomain.com><br/>Record-Route: <sip:145bf82@192.168.193.210;transport=tls;lr><br/>Record-Route: <sip:192.168.193.219:15061;transport=tls;lr;sap=433098584*1*016asm-callprocessing.sar-624908352~1296718381477~-535462632~1><br/>Record-Route: <sip:145bf82@192.168.193.210;transport=tls;lr><br/>Record-Route: <sip:192.168.193.104:5061;transport=tls;lr><br/>Session-Expires: 1200;refresher=uac<br/>Content-Type: application/sdp<br/>Content-Length: 178<br/>P-Location: SM;origlocname="Interoplab";termlocname="Interoplab"<br/>Max-Forwards: 63<br/><br/>v=0<br/>o=- 1296719515 1 IN IP4 192.168.193.104<br/>s=-<br/>c=IN IP4 192.168.193.105<br/>b=AS:64<br/>t=0 0<br/>m=audio 2564 RTP/AVP 8 18 96<br/>a=fmtp:18 annexb=no<br/>a=rtpmap:96 telephone-event/8000<br/><!---->
   
=== SIP: Do not send INFO(dtmf) before call is connected ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61025 61025]
|}
Do not send INFO(dtmf) before dialog is in confirmed state.<!---->
   
= V8 Hotfix12 (80500.36) =
Changes included in Version 8 hotfix12
[http://mantis.innovaphone.com/view.php?id=60894 Definition]
== New Features ==
   
=== Phone: New config option "Proxy" for SIP registrations ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=59396 59396]
|}
Now DNS names can be specified.<br/>Replaces config option "Primary Server Address".<!---->
   
=== phone: " reject if busy" option for incoming announcement calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61412 61412]
|}
In some scenarios it's required that announcement calls are not accepted when the phone is busy.<!---->
   
=== v8 Firmware for IP6010, IP3010, IP1060, IP0010 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61522 61522]
|}
Version 8 Firmware will be released for the new IP6010 Gateway familiy as part of a hotfix release.<!---->
   
=== IP-DECT: Abnormal call release error event ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61705 61705]
|}
Now the DECT Master sends an error event to the event logger every time if an abnormal call release occurs.<!---->
   
=== new: phonesig api method to restart registration process without deregistration ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62165 62165]
|}
WLAN phones we need a way to restart a RAS registration when coming back from a out-of-coverage condition to syncronize the handsets and PBX's registration state.<!---->
   
== Bug Fixes ==
   
=== IP2x/30x: T.38: Option for  high speed data redundancy  ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60866 60866]
|}
to configure this option use <br/>http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl<!---->
   
=== IP2x/30x: T.38: Calling tone (CNG) detect didnt work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60879 60879]
|}
to configure this option use <br/>http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl<!---->
   
=== IP3xx: Trap if switching a PBX from Standy to Off ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=60956 60956]
|}
This happens because we try to unregister from a CONF interface, which does not exist on the IP3xx platform<!---->
''Status:''
pbx.cpp
   
=== SIP: Trap when receicing provisional response for obsolete INVITE ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61035 61035]
|}
In overlap dialing scenarios overlapping INVITE client transactions are used.<br/>Same Call-ID, different CSeq and different To-URI.<!---->
   
=== SIP: Read PAI/PPI header when receiving MESSAGE request ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61086 61086]
|}
Read PAI/PPI header when receiving MESSAGE request in order to get calling party identity<!---->
   
=== Phone: Memory leak when deleting SIP registration ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61132 61132]
|}
Failed to delete registration, but only if trying to delete during state "rgistration failed due to no response from server".<!---->
   
=== H.450 encoding problem with call-transfer and diverting facilities, if length of number was 0 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61222 61222]
|}
A zero lenght number cannot be encoded, it must be omited from the message<!---->
''Status:''
h450.cpp
   
=== SIP: Bug in handling of re-direct responses ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61264 61264]
|}
New remote port was not respected when maddr parameter was present in redirection URI.<br/>E.g.<br/><br/>  sip:662@10.0.77.46:4432;user=phone;transport=Tcp;maddr=10.0.77.46;x-mss-call-id=a515c882e909d311874700903306177f%4010.0.77.70 <!---->
   
=== IP2x/IP30x: T38: Missing "no signal indications" on remote initiated T.38 session ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61273 61273]
|}
This solves a problem with SIP-Provider behing a NAT router on outgoing fax calls.<br/><!---->
   
=== Critical Flag at DECT System Object disappears ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61318 61318]
|}
If the DECT system is replicated from the PBX and systems settings are changed on the DECT system, the critical flag on the DECT System object in the PBX is lost<!---->
''Status:''
dectusers.cpp<br/>dectusers.h
   
=== Calls redialled from call list were not set up with CLIR although CLIR was active for the original call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61321 61321]
|}
The CLIR setting of the original call was saved in the call list but not applied when the call was redialled from list.<!---->
   
=== CFNR at PBX object, was executed on call to busy endpoint ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61323 61323]
|}
should only be executed registration down or no respone at all<!---->
''Status:''
pbx.cpp
   
=== phone: function key Boolean Object with 'Toggle State' checked did not display the correct state sometimes ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61368 61368]
|}
This happened when the state of the boolean object was toggled from 'manual-on' to 'automatic-off' state at the PBX or by another phone with such a key. It did not happen when with a key where the 'Toggle State' checkmark was not set.  <!---->
   
=== SIP: No overlap sending if 'sending complete' was declared ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61472 61472]
|}
Do not start overlapping INVITE transaction for new dialing digit if 'sending complete' was indicated for the call.<!---->
   
=== PBX phone config templates could overrun when a big number of function keys was configured ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61476 61476]
|}
There was a general 4kB size limitation for attributes read from LDAP directory which was too small for the 'phone' attribute of a config template.    <!---->
   
=== Webdav: Bad encoding of special characters in XML properties ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61505 61505]
|}
Bad encoding of file/folder names containing special characters.<!---->
   
=== do not open multiple HTTP sessions when forwarding a big number of alarms in a short time ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61527 61527]
|}
when alerm forwarding is active the fault handler passed new alarms immediately to the forwarding httpclient and httpclient opens a new session when there is no idle session.<!---->
   
=== PBX: Boolean Function Key was not updated when joining group ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61590 61590]
|}
For the Boolean function key it is required to receive Group Indications from the Boolean object, which does not happen if the phone is not member of the group (dynamic out). When joining the group an update should be sent to the phone.<!---->
''Status:''
pbx.cpp<br/>pbx.h<br/>pbx_gi.cpp<br/>pbx_gi.h<br/>pbx_bool.cpp
   
=== Possible to configure use of Feature Codes on Basic Rate ISDN ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61620 61620]
|}
This configuration option is not useful on ISDN BRIs. In fact it usually results in unexpected behaviour.<br/><br/>This option is removed from the user interface.<!---->
''Status:''
ip800/platform/config.h<br/>ip24/platform/config.h
   
=== IP-DECT: OEM module update function ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61671 61671]
|}
The update function for an OEM module was changed.<!---->
   
=== IP-DECT: trap with call transfer ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61676 61676]
|}
Null pointer trap with call transfer and release event from the DECT side.<br/>Trap identification, IP1200, V8 Hotfix 10:<br/>XCPT: no 2 (TLB load)  pc 943fd6d4  ra 94278e9c  va 0000000c<br/><!---->
   
=== PBX-SOAP: Admin function removed password if Object Long Name (cn) was changed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61725 61725]
|}
If the cn is changed the object must be identified by guid an the password of this old object is to be used<!---->
''Status:''
pbx.cpp
   
=== PBX-SOAP: Admin function could not be used to configure "phone-config" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61726 61726]
|}
"phone-config" was missing in the list of allowed attributes<!---->
''Status:''
pbx.cpp
   
=== SNMP, If Index sometimes missing in interfaces walk ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=61985 61985]
|}
SNMP, If Index sometimes missing in interfaces walk<!---->
   
=== SIP: Very large SIP request headers were rejected with 414 Request-URI Too Long ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62033 62033]
|}
SIP request headers larger than 2000 bytes were rejected with 414 Request-URI Too Long<!---->
   
=== ISDN, QSIG, NT, Invalid Progress message was sent ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62190 62190]
|}
The mandatory Progress Indicator was missing in Progress message when rejecting a call. This could cause that the inband busy tone could not be sent.<!---->
''Status:''
nt_tbl.h
   
= V8 Hotfix13 (80500.37) =
Changes included in Version 8 hotfix13
[http://mantis.innovaphone.com/view.php?id=63025 Definition]
== New Features ==
   
=== CAS E1 3bit pulse dialing ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62191 62191]
|}
Support for CAS E1 3bit pulse dialing, which is sometimes used instead of DTMF addressing.<!---->
   
=== RPCAP uses system time instead of uptime now ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62745 62745]
|}
A wireshark capture with RPCAP will now receive packet timestamps with the system time and not the uptime anymore.<!---->
   
=== Gateway Routing: Support of '?' wildcards in CGPN and CDPN output ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62809 62809]
|}
In the routing table digits received at places marked with '?' are forwarded to the respective '?' in the output number. This works for CDPN and CGPN maps in routes. It does not work in interface maps<!---->
''Status:''
gk.cpp<br/>gk.h
   
== Bug Fixes ==
   
=== SIP: INVITE after redirect must not contain the old remote tag ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62263 62263]
|}
INVITE after redirect did contain the old remote tag.<br/>Now it is cleared before new INVITE is sent to new destination.<!---->
   
=== SIP: Expect early inband information if 180 with SDP answer is received ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62275 62275]
|}
Expect early inband information if 180 with SDP answer is received<!---->
   
=== PBX Quickdial: Transferscenario leaves orphaned call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62311 62311]
|}
PBX Quickdial: Transferscerio leaves orphaned call<br/>The orphaned call remains under PBX/Calls and cannot be cleared.<!---->
   
=== License: License upload shows error "No licenses available" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62318 62318]
|}
"No licenses available" when uploading license XML.<!---->
   
=== Do SRTP Re-keying when doing media renegotiation ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62325 62325]
|}
Using the same SRTP key could be a security issue. When after a transfer the same SRTP keys are used, in theory the party doing the transfer could still decrypt the SRTP even if not in this call anymore<!---->
''Status:''
h323ch.cpp<br/>media.cpp<br/>channel.cpp<br/>channel.h
   
=== phone: a call unparked by a phone with recording active was released instead of reconnected  ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62367 62367]
|}
When the phone receices the SETUP indicating the unparked call the call should be automatically connected and become the active call. This failed because the currently active call was not put on hold before and thus there was no free DSP cannel to connect the unparked call. <!---->
   
=== Polish Language could not be configured in the PBX Phone Config ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62410 62410]
|}
The table entry for polish language was missing<!---->
   
=== General btree library problem: Potential Trap if many outgoing registrations need to be retried ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62428 62428]
|}
Actually the problem is in the commonly used btree library, but there are not that many cases in which the libray is used in a way that create the problem<!---->
''Status:''
btree.cpp
   
=== PBX Waiting: Limited DTMF targets could be added using Internet Exporer ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62432 62432]
|}
URL size limitiation of IE -> use POST instead<!---->
''Status:''
pbx_edit_waiting.xsl
   
=== PBX Waiting: Connected Number handling different from normal Connected Number Handling ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62437 62437]
|}
This caused different behaviour whether the operator answered the call on a SIP or H.323 phone. In case of SIP the Connected Number was sent, in case of H.323 not<!---->
''Status:''
pbx.cpp<br/>pbx.h
   
=== SIP: Media negotiation failed when interworking with H.323 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62439 62439]
|}
When calling from H323 to a user with multiple registrations<br/>and the called user accepts on one of its (SIP type) secondary registration,<br/>the media negotiation can fail.<!---->
   
=== PBX: Progress Indicator in Alert not forwarded by PBX ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62483 62483]
|}
This could result in in-band info not played at receiving phone in case no progress incator was sent in previous message of same call<!---->
''Status:''
pbx.cpp
   
=== Call Completion to MD110 didn't work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62512 62512]
|}
Call Completion to MD110 didn't work<!---->
   
=== VM, Smtp authentication sometimes in-place, although not required ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62571 62571]
|}
VM, Smtp authentication sometimes in-place, although not required<br/><!---->
   
=== SIP: Media negotiation issue ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62606 62606]
|}
Handling of re-INVITE w/o SDP offer in 'held' state requires change.<br/><!---->
   
=== PBX: Blind transfer with consultation to mobile endpoint -&gt; Retrieve missing ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62638 62638]
|}
The caller is put on hold for the consultation, but is not retrieved when the transfer happens. If the caller is SIP, this results in no media sent.<!---->
''Status:''
pbx.cpp
   
=== Possible trap on certain compact flash operations ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62703 62703]
|}
There has been the possibility of a trap on certain compact flash file operations.<br/>This trap has been fixed.<!---->
   
=== DHCP client: timeout for response to a REQUEST too small in some case ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62709 62709]
|}
When the DHCP client REQUESTs an OFFERed address a variable timeout (min 2 seconds) is set up. In the case in question the server always responds to DISCOVERs and REQUESTs with a delay of a little bit more than 2 seconds and thus a new DISCOVER was triggered a short time before the ACK arrived.<br/>To overcome this problem the minimum timeout is changed to 5 seconds which should be enough for any server.  <!---->
   
=== ADSP driver: initialization changed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62869 62869]
|}
The ADSP2191 initialization is changed. This fixes some missed voice channels in conference calls.<!---->
   
=== Diagnostic/Tracing on IP6000: Trace flag on TEL could not be cleared ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62914 62914]
|}
once set, it could only be cleared with a !config change command<!---->
''Status:''
tracing.xsl
   
= V8 Hotfix14 (80500.47) =
Changes included in Version 8 hotfix14
[http://mantis.innovaphone.com/view.php?id=63026 Definition]
== New Features ==
   
=== New flash S29GL256P90/S29GL128P90 on IP1200 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=58643 58643]
|}
This flash is used on new IP1200 devices.<br/>Bootcode downgrade to older bootcode is disabled.<br/>If the bootcode is downgraded the bootcode version is shown as 1013.<!---->
   
=== SNMP, innoColdStart Trap to be sent only after sw failure or button reset ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63160 63160]
|}
Settlement of a feature request to have the innoColdStart SNMP trap indicate severe reboot reasons only.<br/><!---->
   
=== DECT: GUI password input limit info ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63349 63349]
|}
The user password is truncated to 15 signs. Now the input field is limited and an info is shown.<!---->
   
=== support for external ringer unit ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63358 63358]
|}
some special purpose phones may be equipped with an external ringer unit. the information controlling the internal ringer is now passed to the module controlling the external ringer unit.<!---->
   
== Bug Fixes ==
   
=== H.323: Don't send a call-independent-signaling call without facilities and user-user information ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62961 62961]
|}
This fix is related to the fix #59088.<br/>A call-independent-signaling call without facilities should not be sent, but if it has got a user-user information, it should be sent.<br/>This fixes the DECT messaging problem on the IP1200.<!---->
''Status:''
h323sig.cpp
   
=== TCP: Ack was not sent under special conditions with re-transmissions ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62965 62965]
|}
This could cause the breaking of a TCP connection in case of packet loss, even if the packet loss was not too bad<!---->
''Status:''
tcp.cpp
   
=== Trap when processing webdav requests ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=62980 62980]
|}
Trap when webdav request session were terminated irregularly.<!---->
   
=== SIP: Bad encoding of To-URI in INVITE when handling REFER with special chars in user part of Refer-To URI ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63030 63030]
|}
Refer-To: <sip:+49231395710880_(399)@172.20.173.104><br/>received with REFER was mangled into<br/>To: <sip:%2049231395710880_(399)@172.20.173.104><br/>and send in INVITE<!---->
   
=== HTTP-Server: Closing connection after transaction causes trouble with Webdav client ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63045 63045]
|}
NetDrive client fails when uploading files<!---->
''Status:''
http.cpp
   
=== Webdav: Bug when handling GET with Range header ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63131 63131]
|}
When applied on a zero length file this response was returned:<br/><br/>\tHTTP/1.1 206 Partial Content<br/>\tDate: Tue, 12 Apr 2011 14:52:23 GMT<br/>\tServer: innovaphone Virtual Appliance / 9.00 dvl [xxx/1000/0]<br/>\tAccept-Ranges: bytes<br/>\tContent-Type: application/octet-stream<br/>\tContent-Length: 0<br/>\tContent-Range: bytes 0-4294967295/0<br/><br/>Error response "416 Requested Range Not Satisfiable" must be returned instead.<!---->
   
=== Webdav: Don't keep zero-length files open on server side ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63133 63133]
|}
In case of large files, NetDrive performes GET operation between PUT0 and PUT.<br/>The actual PUT was rejected with 500 error resonse then.<!---->
   
=== 62879: ISDN, QSIG, NT: No Disc Option can be used to send PROGRESS instead of DISC - fix for this fix ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63209 63209]
|}
This fix from hotfix13 did only for calls on which a CALL-PROC was sent as well. For calls still in overlap dialing (only SETUP-ACK sent) it did not work<!---->
''Status:''
nt_tbl.tbl
   
=== SIP: Fix for dialog-info notification ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63249 63249]
|}
NOTIFY for dialog state 'terminated' was missing sometimes.<!---->
   
=== SIP: Trap when session timer is used ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63271 63271]
|}
Trap on collision of session timer and call release<!---->
   
=== SIP: Authentication passwords were truncated ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63321 63321]
|}
Authentication failed because password was truncated.<!---->
   
=== SIP: Not accepting calls from alternative proxy ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63327 63327]
|}
When being registered at a proxy with 2 ip addresses the gateway does not accept calls from the alternative ip address.<!---->
   
= V8 Hotfix15 (80500.49) =
Changes included in Version 8 hotfix15
[http://mantis.innovaphone.com/view.php?id=63485 Definition]
== New Features ==
   
=== DECT: Radio firmware for new handsets ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63577 63577]
|}
The new radio firmware PCS05Ah accepts new handsets with the new IPEI number range.<!---->
   
=== phone: improved czech display texts ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63998 63998]
|}
now all texts are translated to czech, previous errors were fixed (translations provided by zakharova@annexnet.cz)<!---->
   
== Bug Fixes ==
   
=== PBX: License mechanism changed to allow easy migration to new version ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63381 63381]
|}
- licences of different versions may be installed<br/>- check for min version<br/>- v8 master can act as license master for v9 licenses<br/>- applications may run on older version<!---->
''Status:''
inno_lic.cpp<br/>inno_lic.h<br/>pbx.cpp<br/>pbx_api.h<br/>pbx_general.xsl<br/>pbx_edit_loc.xsl<br/>
   
=== PBX: Trunk - don't retry call to next gateway if wrong number ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63386 63386]
|}
all gateways registered to a trunk are by definition to the same network, so a rerouting is useless, if the cause indicates that the dialed number was wrong<!---->
''Status:''
q931lib.h
   
=== Command traps in minifirmware on joining or leaving Kerberos realms ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63415 63415]
|}
Because command does not check if kerberos_client_provider::provider is null.<br/><br/>Files: command.cpp<!---->
   
=== TEL and PRI1-4 not contained in 'PPP connection port' dropdown menu on ip6010, ip3010 and ip1060 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63419 63419]
|}
'PPP connection port' dropdown should contain TEL and PRI1-4<!---->
''Status:''
ip_config.cpp
   
=== ip0010 wizard configures PRI1, gateway/interfaces shows PRI1 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63430 63430]
|}
PRI1-L1 must be renamed into PRI1-CLK<!---->
''Status:''
config.h, ip6010.cpp
   
=== HTTP-Client: Bad encoding of uri parameter in digest authentication ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63469 63469]
|}
Uri parameter in digest authentication was not URL encoded<!---->
   
=== Gateway: Outgoing Call Completion did not work when outgoing call was routed through TONE interface ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63517 63517]
|}
Outgoing CC request did not went out to ISDN interface.<!---->
   
=== SIP: Message buffer too small for REGISTER request for re-try with authentication ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63539 63539]
|}
On some installations a change-of-nonce at server side may cause volatile "Registration down error" on client side.<!---->
   
=== certain non latin-1 characters entered via WEB interface or provided by an external LDAP Server cause display errors ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63591 63591]
|}
entering such characters via copy/paste as when editing a PBX object may result in an xml-error when showing PBX objects.<br/>when such characters are provided by an external LDAP Server to a phone the display may get cleared.<br/>Now such characters are transcribed to a single latin1 character or replaced by a '-' if no transscription is available. <!---->
   
=== Web-UI: PBX password length is limited to 15 chars ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63640 63640]
|}
Added tooltip and fixed maxlength attribute on input elements.<!---->
   
=== License: Character encoding problem ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63645 63645]
|}
Character encoding problem<!---->
   
=== config download may trap when malformed LDAP config data  has been uploaded ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63678 63678]
|}
a buffer overrun happens on config download when a "mod cmd FLASHDIR0 add-view nnn cn=..." line with a length > 63 characters has been uploaded.  <!---->
   
=== Presence functionality is not available when registered via H323 at a non-innovaphone PBX ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63745 63745]
|}
Presence operations via H323 are encoded in private facility elements which are unknown to a non-innovaphone PBX. Presence control calls sent to such a PBX may be misunderstood and routed back as normal voice call to the sending phone.<br/>Thus no presence control calls must be sent to such a PBX.<!---->
   
=== Trap when starting from flash_stick ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63752 63752]
|}
and flash memory not yet programmed with bootcode<!---->
''Status:''
ip6010.cpp
   
=== SIP: Allocated message size to small for INVITE redirect response (Avaya) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63829 63829]
|}
Memory allocation is a bit to tight to fit the message due to many Via headers.<br/><br/>INVITE sip:3003@192.168.150.140:2059;transport=UDP SIP/2.0<br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=udp;lr><br/>Record-Route: <sip:192.168.150.114:15060;lr;sap=315810451*1*016asm-callprocessing.sar1905633216~1304428214402~-1054885358~1><br/>Via: SIP/2.0/UDP 192.168.150.115;rport;branch=z9hG4bKC0A896726E7526620194612-AP;ft=192.168.150.115~13c4<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526620194612<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194610<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194609<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bK0e2106b7388e016424db9a29200-AP;ft=11786<br/>Via: SIP/2.0/TCP 192.168.150.118;branch=z9hG4bK0e2106b7388e016424db9a29200;avaya-cm-term-reaction=shortcut<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bKC0A896726E7526620194608-AP;ft=12651<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526620194608<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194606<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194605<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bK0e2106b7388e018424db9a29200-AP;ft=11786<br/>Via: SIP/2.0/TCP 192.168.150.118;branch=z9hG4bK0e2106b7388e018424db9a29200<br/>Via: SIP/2.0/TCP 192.168.150.84;branch=z9hG4bK200_f1774512c29cc2e5cd78966_I2371<br/>User-Agent: Avaya one-X Deskphone AVAYA-SM-6.1.1.0.611023 Avaya CM/R016x.00.1.510.1<br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=tcp;lr><br/>Record-Route: <sip:192.168.150.114:15060;transport=tcp;lr;sap=315810451*1*016asm-callprocessing.sar1905633216~1304428214355~-1054885362~1><br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=tcp;lr><br/>Record-Route: <sip:192.168.150.118;transport=tcp;lr><br/>Session-Expires: 1800;refresher=uac<br/>Content-Type: application/sdp<br/>Content-Length: 215<br/>...<!---->
   
=== IP152: Flash access not working with version 8050047 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64009 64009]
|}
With fix #58643 16 bit access to spansion flash doesnt work<!---->
''Status:''
boot_coldfire.mak common.mak flash_coldfire.c
   
=== No received cause code should be treated as 'normal clearing' ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64043 64043]
|}
Was sometimes treated as cause code to do re-routing. This happened esspecially with multiple registrations to v8 gateway object. A call sent successfully to the gateway on the first regsitration was sent again on the second registration after call clearing.<!---->
''Status:''
q931lib.cpp<br/>relay.cpp
   
=== missing response 'reset required' when changing PRIx-Lx config options ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64055 64055]
|}
changing i.e. the ,NT-Mode' config option didn't show the 'reset required' link button after pressing 'OK'.<!---->
''Status:''
falc56_drv.cpp, config.h ipac_drv.cpp V9:falc56_drv.xsl
   
=== PBX: Transfer Recall timer was not started if destination was ringing after blind transfer ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64064 64064]
|}
After a blind transfer without consultation to a busy destination the recall timer should be started as soon as the destination is not busy anymore and the call is delivered<!---->
''Status:''
pbx.cpp
   
=== Gateway: Allow interface maps for analog interfaces as well ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64068 64068]
|}
Was prohibited in the past, but there are uses for this.<!---->
''Status:''
ip24/config.h
   
=== Conference on IP6000 Hardware 200 and lower not working with v8hf14 and v9 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64132 64132]
|}
The ADSP serial port has been changed from SPORT1 to SPORT0 for the IP6010.<br/>Old IP6000 hardware has the SPORT0 not connected, so now SPORT1 is again used on IP6000.<br/><!---->
   
=== PBX: Potential Trap on calls to exec, map or waiting object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64135 64135]
|}
under some rare circimstances, which are unfortunatly not known, there could be a NULL pointer access<!---->
''Status:''
pbx_exec.cpp<br/>pbx_wait.cpp<br/>pbx_map.cpp
   
= V8 Hotfix16 =
Changes included in Version 8 hotfix16
[http://mantis.innovaphone.com/view.php?id=64216 Definition]
== New Features ==
   
=== SIP: Ignore History-Info URI not containing user part ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64211 64211]
|}
Calls from Avaya PBX were indicated as diverted/redirected calls<br/>since they have History-Info header.<br/>But URI in History-Info header does not have userpart.<br/>Ignore that.<br/><!---->
   
=== SIP: Treat "Privacy:off" like "Privacy:none" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64692 64692]
|}
Treat "Privacy:off" like "Privacy:none" when receiving INVITE<!---->
   
=== H.323: Display call state in "Signaling Timeout" error log ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65194 65194]
|}
To provide better indication about the nature of the problem<!---->
''Status:''
h323sig.cpp
   
=== SIP: New config file option /add-cn-capability ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65313 65313]
|}
Required for mediation server (lync) interoperability.<br/>Otherwise mediation server complains:<br/>  "The Gateway peer does not support comfort noise"<!---->
   
=== Gateway: Allow sending of Date/Time in Connect on ISDN interfaces ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65445 65445]
|}
Was missing in the User Interface, so it could not be configured<!---->
''Status:''
relay_edit_phys.xsl<br/>config.h of ip800, ip24, ip3000, ip6000, ip6010
   
== Bug Fixes ==
   
=== SIP: Media negotiation problem during transfer to early media source ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63422 63422]
|}
test\\9.00\\relay\\early-media failed<!---->
   
=== PPPOE: specific configuration not reachable from config web page ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64192 64192]
|}
problem: PPPOE: specific configuration not reachable from config web page, so no new PPPOE can be configured (already present ones run though), also ISDN part always visible<br/><br/>solution: fixed<br/><br/>files: ip_pppif.xsl (now check for PPPOE0, PPPOE1 and PPPOE2 types)<br/><br/>products: all (gateways effected)<br/><br/>risks: none<br/><!---->
   
=== Out Of Memory Trap when running VM without prompt files ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64243 64243]
|}
When calling into a Voice Mail object without prompt files<br/>memory objects are allocated at high rate without being freed.<br/>Memory is freed at disconnect.<br/>This may cause a OOM trap when call stays connected for a longer time.<br/><!---->
   
=== SIP: P-Asserted-Identity in UPDATE not working ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64289 64289]
|}
PAI with changed remote party identification was not handled<br/>if also Session-Expires header was present in UPDATE request.<!---->
   
=== supress "Send Number" for calls triggered by a 'Dial' function key with 'Send as Control Call' checked ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64365 64365]
|}
When using a 'Dial' function key with 'Send as Control Call' checked to control a call recording device the unique original calling party number must be passed to the recorder. The 'Send Number' configured in the the PBX user object may be the same for a group of phones and does not identify a certain phone.<!---->
   
=== Trunk Park/Pickup (line keys) did not work anymore ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64373 64373]
|}
Collateral damage from fix<br/><br/>fix: #61590: PBX: Boolean Function Key was not updated when joining group<!---->
''Status:''
pbx_gi.cpp<br/>pbx_gi.h<br/>pbx.cpp (v9 only)<br/>pbx.h (v9 only)<br/>pbx_api.h (v9 only)
   
=== ring-back tone missing in certain call fork cases ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64410 64410]
|}
a SIG-PROGRESS indicating in-band-info received after SIG_ALERT stopped the ring-back tone and if no in-band-info is provided the user misses the ringback tone. Now  the ringback  tone is stopped and started again if there is no RTP data received within 500 milliseconds.<!---->
''Status:''
Already fixed in V9 final
   
=== memory leak check missing for last parked call info ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64445 64445]
|}
when a call is parked using the 'Park' function key info about the parked call is kept for later checks when the call is unparked again. the leak check for this info was missing. <!---->
   
=== Timeout when calling Mobile endpoint which does not send alert ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64563 64563]
|}
Some SIP carriers do not send correct alert but only something which can be translated to CALL-PROC. In this case the CALL-PROC was not forwarded to the caller and therefore the call timed out after 12s<!---->
''Status:''
pbx_mobility.cpp
   
=== A PBX user with "Full PBX Administration" Rights could not edit phone configuration ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64572 64572]
|}
The configuration pages could be opened once but after changing an item the input was disabled<!---->
   
=== A Bootcode Update could disrupt the Media stream for some seconds ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64631 64631]
|}
This was observed on phone devices with relatively slow flash memory when a bootcode update took place while a call was active.  <!---->
   
=== phone: picking up a call failed sometimes ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64679 64679]
|}
Sometimes pressing the partner, pickup or park key to pick up an alerting or parked call had no effect.<!---->
   
=== phone: prevent the pc port of the ethernet switch from receiving frames directly from the phone firmware ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64689 64689]
|}
In some cases is not desired that frames sent by the phone firmware via the cpu port are recieved by the pc port. This may be prevented now by<br/>    config add ETH0 /isolate-pc  <!---->
   
=== PBX User Interface did not work with Groups containing XML reserved characters (&amp;amp;,&lt;,&gt;,...) or non-ascii ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64695 64695]
|}
XML or URI encoding was missing in some paces. The browser could not display the page.<br/>This happend when using the left PBX/Group tree for nvigation<!---->
''Status:''
pbx_admin.cpp<br/>pbx_objs_left.xsl<br/>pbx_objs_right.xsl
   
=== TLS: Error on processing huge handshake messages ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64702 64702]
|}
The current implementation does not work with handshake messages that are bigger than 8 kilobytes. Especially the CertificateRequest message that is used for MTLS can be bigger.<br/><br/>files: tls.cpp<!---->
   
=== PBX v5 SoftwarePhones licenes did not work on v9 or v8 PBX ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64709 64709]
|}
An old v5 SoftwarePhone license installed on a v9 PBX did not work for v5 SoftwarePhones<!---->
''Status:''
inno_lic.cpp<br/>inno_lic.h
   
=== PBX:OEM Voicemail license did not work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64769 64769]
|}
collateral damage from supporting licenses from different versions on a PBX<!---->
''Status:''
inno_lic.cpp
   
=== SIP: Interworking of "Q.931 CALL PROCEEDING" into "183 Session Progress" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64770 64770]
|}
Required if only CALL PROCEEDING and no ALERTING is received.<!---->
   
=== ip800 trace telling wrong information about power source ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64826 64826]
|}
PCBs since V300 cannot detect POE power and trace therefore told 'not powered'.<!---->
''Status:''
ip800.cpp
   
=== SIP: Interworking of calls with Q.931 Bearer Capability "Unrestricted digital information" rejected ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64932 64932]
|}
Calls with Q.931 Bearer Capability "Unrestricted digital information" were rejected.<!---->
   
=== Qsig: leg1Info sent with ALERT msg, instead of in FACILITY msg ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64948 64948]
|}
Problem: Siemens ACWin got confused after upgrade v6->v7. Qsig mandates to send the leg1Information within a FACILITY message (while H.323 does also allow for an ALERT to carry the leg1).<br/><br/>Solution: Detect and treat this case accordingly within the relay/gateway.<br/><br/>Files: relay.cpp<br/><br/>Risk: none<!---->
   
=== H.323: Don't forward G.729B capability if silience compreession not enabled ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65133 65133]
|}
This solves quality issues some SIP provider have with G.729B.<!---->
''Status:''
h323ch.cpp
   
=== SIP-H323 calls with SRTP: No media after multiple Hold/Retrieve ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65185 65185]
|}
After first Hold/Retrieve there was no SRTP, after the next Hold/Retrieve very often no media<!---->
''Status:''
h323ch.cpp
   
=== PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65261 65261]
|}
If "Execute Member Diversions" was checked a call to Broadcast was also sent to CFB destination<!---->
''Status:''
pbx_bc.cpp
   
=== Gateway: Not possible to enter wildcards ('.') in interface maps ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65280 65280]
|}
wrong check for correct value<!---->
''Status:''
gk.cpp
   
=== Gateway: Configured signaling port got lost, when ediiting interface maps ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65303 65303]
|}
The signaling port was reset to the standard port when saving interface mappings<!---->
''Status:''
gk.cpp
   
=== UTF-8  Conversion Wrong Since HF15 On Little-Endian Machines (e.g. IP3010) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65317 65317]
|}
LDAP Replication produced wrong contents<br/>Affects IP6010, 3010, 0010, 1060<!---->
   
=== Trap in rarely used OS function bufman::remove ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65338 65338]
|}
could result in negative length of buffer<!---->
''Status:''
os.cpp
   
=== buffer overflow in fat32 method ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65344 65344]
|}
The borders of a static buffer has been exceeded.<br/><!---->
   
=== PBX-SOAP: Trap if initiating multiple outgoing calls from a Waiting object at the same time ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65418 65418]
|}
Some applications do this to deliver voice messages<!---->
''Status:''
pbx_wait.cpp<br/>pbx_wait.h
   
=== Gateway: Record URL at SIP interface was lost when Internal registration was configured ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65443 65443]
|}
UI problem<!---->
''Status:''
gk.cpp
   
=== PBX: Call-Intrusion could result in wrong name display ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65462 65462]
|}
esspecially for silent intrusion<!---->
''Status:''
signal.cpp<br/>h450asn1.h<br/>pbx.cpp<br/>pbx.h
   
= V8 Hotfix17 (09-80500.55) =
Changes included in Version 8 hotfix17
[http://mantis.innovaphone.com/view.php?id=65485 Definition]
== New Features ==
   
=== QSIG: Avaya expect Progress Indicator with external calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66074 66074]
|}
Avaya uses the Progress indicator 'Interworking with a public network' to identify a call as external. This Progress Indicator is now added for calls from a Number NOT with private numbering plan (which is our way to identify internal calls)<!---->
''Status:''
q931.cpp
   
=== ISDN: New interop flag to forward network provided or checked cli only ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66183 66183]
|}
Useful if the real calling number is needed and not a number provided by CLIP no screening<!---->
''Status:''
q931.cpp<br/>q931.h<br/>isdn_interop.xsl
   
== Bug Fixes ==
   
=== SIP: Session refresh was taken as session modification ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=63310 63310]
|}
Local SRTP key was re-calculated after re-INVITE for session refreh was received.<br/>Causes SRTP decode error at remote side.<br/>CUCM scenario<!---->
   
=== IP6010, IP6000: Use optimized memcpy ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64587 64587]
|}
Use of load/store multiple and shifts for 32 bit alignment speeds up memcpy by a factor of approx 2<br/><br/>Orginal memcpy<br/><info product="IP6010" mips="800Mips"><br/><memcpy bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><read bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><write bytes="1000000" time="2ms" speed="470.588Mbyte/s"/><br/><stack_memcpy bytes="1000000" time="7ms" speed="133.333Mbyte/s"/><br/><uncached_memcpy bytes="1000000" time="41ms" speed="24.169Mbyte/s"/><br/><aes bytes="1000000" time="135ms" speed="7.373Mbyte/s"/><br/><sha bytes="1000000" time="70ms" speed="14.260Mbyte/s"/><br/></info><br/> <br/>Optimized memcpy:<br/><info product="IP6010" mips="800Mips"><br/><memcpy bytes="1000000" time="1ms" speed="888.888Mbyte/s"/><br/><read bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><write bytes="1000000" time="2ms" speed="421.052Mbyte/s"/><br/><stack_memcpy bytes="1000000" time="7ms" speed="142.857Mbyte/s"/><br/><uncached_memcpy bytes="1000000" time="15ms" speed="64.000Mbyte/s"/><br/><aes bytes="1000000" time="138ms" speed="7.200Mbyte/s"/><br/><sha bytes="1000000" time="70ms" speed="14.285Mbyte/s"/><br/></info><br/><br/>CPU load with the test test/9.00/box/dsp/ip6010 shows approx 1% lower CPU load.<br/>Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate, compared to 9708Kbyte/s with the old memcpy.<br/><br/>With ECC enabled the CPU load was 19% / 21% without SRTP and 31% / 33% with SRTP<br/>With ECC Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate10309<!---->
''Status:''
ip6010.mak ip6000.mak arm.mak box/arm/memcpy.S<br/><br/>v8: ip6010.mak, box/box.mak, box/memcpy.S
   
=== Incorrect rpcap timestamp after TRACE LOST messages ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=64915 64915]
|}
The RPCAP timestamp (Wireshark) after a TRACE LOST message was incorrect, as the TRACE LOST message contained an incorrect timestamp.<!---->
   
=== VM, Project script didn't run for endpoints having "Send Number" configured ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65456 65456]
|}
VM, Project script didn't run for endpoints having "Send Number" configured<!---->
   
=== Kerberos: Do not allow registration of multiple databases for one realm name ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65589 65589]
|}
This happened when a box hosted multiple PBXes with the same system name.<br/><br/>files: <br/>kerberos_if.cpp<br/>kerberos_kdc.h (v9 only)<br/>kerberos_kdc.cpp<br/>kerberos_db.cpp<!---->
   
=== DECT: Trap during registration up handling ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65698 65698]
|}
Trap in DECT Master fixed. It occurs if the master endpoint is in delete state and a RAS registration up event is received.<!---->
   
=== MWI does not work in various Node/Pbx combination ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65750 65750]
|}
MWI does not work in various Node/Pbx combination<!---->
   
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65798 65798]
|}
Occurred on IPBL[4.1.22]<!---->
   
=== SIP: Fix for SDP answer to SDP offer with "a:inactive" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65863 65863]
|}
Interop with CUCM.<br/>Should return RTP/AVP(inactive) if offer was RTP/AVP(inactive).<br/>Not not RTP/SAVP(inactive).<!---->
   
=== Message Waiting Interrogation: Result message coding wrong ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65912 65912]
|}
a malformed message was displayed in wireshark<!---->
''Status:''
h450.cpp<br/>h450asn1.h
   
=== SIP: Set CLIR if display string of From-URI contains "Anonymous" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65925 65925]
|}
Not only if userpart of From-URI contains "anonymous".<!---->
   
=== ip6010 - same MAC address was assigned to ETH0 and ETH1 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65939 65939]
|}
this results in problems when both interfaces are connected to the same LAN segment <!---->
   
=== PBX-SOAP: Don't provide caller number if CLIR was used on call to monitored endpoint ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65944 65944]
|}
If this was an internal call, the PBX knows the calling number anyway, but it should not be sent on SOAP<!---->
''Status:''
pbx_xml.cpp
   
=== PBX-SOAP: UserDTMF did not send DTMF to Voicemail or Waiting Objects ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65958 65958]
|}
It only sent DTMFs to a VOIP connection<!---->
''Status:''
pbx_xml.cpp
   
=== Gateway SIP Interfaces: Could not configure internal registration for a disabled interface ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65975 65975]
|}
and if a interface was disabled afterwards, the config for the internal registration was lost<!---->
''Status:''
gk.cpp
   
=== SIP: Trap when receicing provisional response with RSeq header ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65986 65986]
|}
Trap when trying to send PRACK<!---->
   
=== ip6010 - frame loss on ethernet ports running in a VLAN ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66028 66028]
|}
receiving of VLAN tagged frames did not work stable, when running ping -t over a longer time a frame loss from 5 to 10 percent was reported <!---->
   
=== PBX Broadcast: CFNR was executed only after No Response Timeout even if no member ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66032 66032]
|}
If there is no member in the broadcast group, a CFNR configured at the Broadcast object should be executet immediatelly.<br/><br/>This was a collateral damage from hotfix<br/><br/>65261: PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions" <!---->
''Status:''
pbx_bc.cpp
   
=== IP3010/6010: fax problems ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66110 66110]
|}
* CED is not transfered <br/>* Wrong T38 encoding in V8 <!---->
''Status:''
ac_dsp3.cpp ( AC491 doesnt want the V21/V22... relay bits set )<br/>config.h ( config.h, X missing, on V9 this parameter is not needed )
   
=== PBX: Missing Group Indications when SIP phone is monitoring ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66148 66148]
|}
If a SIP phone is monitored by another SIP phone,<br/>there are GI's missing if the monitored SIP phone is calling.<!---->
   
=== DECT: Delete duplicate LDAP 'pbx' &lt;gw&gt; items ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66174 66174]
|}
Now duplicate LDAP 'pbx' <gw> items are deleted by the DECT users module.<!---->
   
=== PBX Trunk: Prefix was added to connected number even if no connected number present ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66213 66213]
|}
The PBX then displayed just the Trunk prefix as remote number on the calls page when the call was connected.<!---->
''Status:''
pbx_trunk.cpp
   
=== PBX-SOAP: FindUser should not show hidden objects ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66216 66216]
|}
Could be confusing<!---->
''Status:''
pbx_xml.cpp
   
=== IP6010-CF: Kingston compact flash was not recognized ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66269 66269]
|}
the card was not recognized because a register was wrongly initialized.<!---->
   
=== SIP: Bug in SDP handling ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66274 66274]
|}
If value of the session id and version in the o line are zero.<!---->
   
=== phone: Hexadecimal values instead of descriptive texts were displayed for some rare disconnect causes ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66343 66343]
|}
"0x57 - unknow cause" was displayed instead of "user not a CUG member". Mainly german descriptive texts were missing.<!---->
   
= V8 Hotfix18 (80500.57) =
Changes included in Version 8 hotfix18
[http://mantis.innovaphone.com/view.php?id=66417 Definition]
== New Features ==
   
=== X.509: Add key usage to certificate requests ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66413 66413]
|}
The Microsoft CA (standard) does not write the key usage into the certificate if it is not specified in the request.<!---->
   
=== DHCP-client  monitors ethernet link down/up events and revalidates current lease after link up ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67006 67006]
|}
This prevents problems when a device is hot plugged to another network.<br/>Further this helps to overcvome a problem with certain cable modems.    <!---->
   
== Bug Fixes ==
   
=== SOAP, Send leg2Info.originalCalled Info ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66422 66422]
|}
As CallInfo.No with type="leg2orig"<!---->
''Status:''
pbx_xml.cpp
   
=== PBX CF Filter for external calls did not work as expected in case of chained CFs ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66599 66599]
|}
A filter for external calls did not match if the external call was forwarded already by an internal user<!---->
''Status:''
pbx.cpp
   
=== Gateway: Trap in case of collision of hold and clearing from remote ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66642 66642]
|}
This could happen on gateways with analog interfaces if the R-Key was pressed right when the other side hung up<!---->
   
=== H.323 potential trap if AlertingNumber is received ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66710 66710]
|}
is no problem with existing equipment, because we don't know of any sending an AkertingNumber. Could become an problem if we do this sometimes in the future<!---->
   
=== H.323 Coding error, when forwarding tunneled SDP in some cases ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66727 66727]
|}
This could happen if during call setup a media negotiation happened on a call with a SIP and a H.323 leg.<br/><br/>This happened for example if a call was received from a SIP Trunk to a Quickdial object in the PBX. The outgoing call from Quickdial could fail because of this.<!---->
   
=== Release not forwarded in quick dial object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=66728 66728]
|}
If the called party released the call, the remote party didn't get the release.<!---->
   
=== possible noise in PRI connections with ip6010 ip3010 ip1060 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67302 67302]
|}
some few gateways may produce noise when using the PRI ports. This can be fixed with a new CPLD code contained in future firmware.<!---->
''Status:''
cpld.h
   
= V8 Hotfix19 (80500.58) =
Changes included in Version 8 hotfix19
[http://mantis.innovaphone.com/view.php?id=67521 Definition]
== New Features ==
   
=== ip200a/230/240:  handset conversations can be monitored in a directly connected  headset ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67666 67666]
|}
This feature is required for a special application and is supported only for ip200a/230/240 phones with a directly connected headset (non DHSG).<br/>It is enabled via<br/>  config add INCA_DSP /handset-spy <volume><br/>whith <volume> in the range from 1..8  <!---->
   
== Bug Fixes ==
   
=== IPxx10: error handling in sata driver ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67229 67229]
|}
Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.<!---->
   
=== DECT: IP6000/IP6010/... default config Master mode off ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67479 67479]
|}
Now the Dect Master is in mode off by default for the IP6000/IP6010/...<!---->
   
=== VM: Trap while processing self-forwarded call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67570 67570]
|}
VM: Trap while processing self-forwarded call<!---->
   
=== SIP: Uninitialized data in SDP offer/answer ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67617 67617]
|}
Applies to G.726 exclusive calls only.<!---->
   
=== SIP: Interoperability with Lync and media-bypass ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67645 67645]
|}
Ack contained wrong To-Tag when calling a lync client in media-bypass scenario.<br/>Results into call drop after 30 seconds.<!---->
   
=== PBX: Don't forward original diverting_leg2 info if divertion is executed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67686 67686]
|}
The leg2 information which is generated when executing an diversion already contains theoriginal called number from previous diversions, so the old leg2 info is not needed anymore. In fact it is harmfull if the call is received by an application only looking at the first leg2 info (e.g. Voxtron)<!---->
   
=== PBX: License accounting in centralized licensing scenario wrong if master not available ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67698 67698]
|}
When the master is available the slave stores the licenses from the master including the usage. This stored usage included the licenses used by the slave itself, so if after a reset the master was not available the local usage just added to this.<br/><br/>Now from the stored usage the local usage is subtracted.<!---->
   
=== PBX Trunk: Problem with Forking to trunk if multiple GWs are registered to Trunk ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67720 67720]
|}
If one of the gateways rejected the call (no channel, not connected, ...), the original call from which was forked was disconnected<!---->
   
=== SIP: Fix for early media from Waitng Queue ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67775 67775]
|}
PROGRESS after ALERT was not handled by SIP stack.<br/>Now 183 Session Progress with SDP is send after 180 Ringing w/o SDP.<!---->
   
=== H.323: A name_id of length 0 resulted in invalid H.450 coding ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67796 67796]
|}
An empty name identification received was forwarded in H.323 as invalid H.450. Such a name is now forwarded as 'name not available'.<!---->
   
=== H.323 Malformed packet ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67803 67803]
|}
The ASN.1 encoder had a bug under one special condition: For a constrained character string with a maximum length of more or equal to 16bits, with an effective length of zero, the padding for octett alignment was missing for the zero length bitfield containing the string.<br/><br/>In H.323 this only happens for the CallIdentity used for H.450 call transfer message in case of blind transfer without consultation.<br/><br/>This fix breaks compatibility with earlier versions, for this reason this fix is available for version 9,8,7 and 6.<br/><br/>If phones and PBX with versions containing and not containing this fix are mixed  the following problems will occur:<br/>- A blind transfer without consultation (initiated with the redial key) is not possible<br/>- A call which was transfered without consultation is not displayed at the transfered-to phone as transfered<br/><!---->
   
=== SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67819 67819]
|}
If in incoming SIP was routed to multiple destinations<br/>the final session could be media-relay although not configured.<!---->
   
=== SIP: DNS problem when SRV response provides no additional records ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67907 67907]
|}
If 2-step resolving is required (SRV and A) the service port<br/>of the SRV response got lost and default SI Pport 5060 was used.<!---->
   
=== SIP: Trap when configuring STUN server on a SIP/TCP or SIP/TLS interface ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67923 67923]
|}
STUN is for SIP/UDP only.<!---->
   
=== PBX: Master/Slave compatibility problem with version 9 and version 8 and non-ascii characters in PBX name ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=67956 67956]
|}
In version 8 only latin1 characters were allowed, which means in unicode the high byte was always 0. So it could be ignored and when sending location information between master and slave sometimes the high byte contained 0xff.<br/><br/>In version 9 this non-ascii location information was not correct unicode at all.<br/><br/>The problem happened only if non-ascii characters were used when naming a PBX.<!---->
   
=== PBX: End of call intrusion was not signaled to the phone ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68007 68007]
|}
The call intrusion tone was generated even if the intrusion was terminated<!---->
   
=== phone_inca:  "ETH0/Isolate PC Link" checkmark could not be cleared via WEB UI once set ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68098 68098]
|}
Only a WEB UI problem, a "config rem ETH0 /isolate-pc" did help.<!---->
   
=== SIP: Interoperability with LinkSys SPA3102 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68174 68174]
|}
LinkSys SPA3102 gives "g729a" as RTP payload type mapping:<br/><br/>    v=0<br/>    o=- 510843041 510843041 IN IP4 192.168.10.20<br/>    s=-<br/>    c=IN IP4 192.168.10.20<br/>    t=0 0<br/>    m=audio 16404 RTP/AVP 18 100 101<br/>    a=rtpmap:18 G729a/8000<br/>    a=fmtp:18 annexb=no<br/>    ...<br/><br/>Needs to be handled.<!---->
   
=== Gerneral/Admin page was broken if too many authentication servers were configured ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68231 68231]
|}
The number of authentication servers is now restricted to 10.<!---->
   
=== phone: intrusion call started in handset mode is not terminated when going on hook when TAPI or operator run on PBX ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68249 68249]
|}
With TAPI or operator running on the PBX the the signaling of a busy condition is changed such that a disconnect instead of a release is sent. The disconnect was not handled correctly, the hookswitch state was lost and the next on-hook signal was ignored. TThus teh call could be terminated with the disc-key only.<br/><!---->
   
=== IP-DECT: Adding OEM radios to Kerberos realm did not work with passwords containing special characters ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68377 68377]
|}
The password was not URL-decoded when reading it from the UI.<!---->
   
=== DTMF user configuration with invalid checkbox check for presence setting ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68383 68383]
|}
The check of the checkmark of the presence setting was wrong.<!---->
   
=== X509: Fix for reading innovaphone info from flash ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68435 68435]
|}
Parsing the innovaphone info text was incorrect<!---->
   
=== License: Be safe against factory reset during license invalidation ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68447 68447]
|}
If factory reset is done before license invalidation procedure is complete,<br/>will keep you from completing the license invalidation.<br/>Now the procedure can be completed even after factory reset.<!---->
   
=== phone: DHSG headset not reset to idle after a hookswitch signal in idle state ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68567 68567]
|}
most DHSG headsets generate a hookswich signal and enter voice mode when taken out of basestation. This hookswitch signal was simply ignored.<br/>Now the voice mode is cleared after one second if there is no other DHSG event before.<!---->
   
= V8 Hotfix20 (80500.59) =
Changes included in Version 8 hotfix20
[http://mantis.innovaphone.com/view.php?id=69989 Definition]
== New Features ==
   
=== ISDN interop issue with SecuGATE LI 30 from Sirrix ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69168 69168]
|}
The SecuGATE LI30 is sending/receiving ISDN INFO messages in Call Proceeding State (State 3 and state 9), which was not supported<!---->
   
=== Allow multiple HTTP IP address filters (allowed stations) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69645 69645]
|}
synced from V9<!---->
''Status:''
http.cpp<br/>http.h<br/>http.xsl
   
== Bug Fixes ==
   
=== Gateway: Allow configuration of username and password for ENUM/SIP interfaces ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68147 68147]
|}
For rare where remote destination server asks for authentication.<br/>(And all remote destination servers ask for same auth or remote destination server s always the same.)<!---->
   
=== SIP/TCP: Transport error when connection is closed by client ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68578 68578]
|}
If transaction client closes connection before final response has been sent,<br/>the server tries to open a new connection toward ephemeral port of closed connection.<!---->
   
=== SIP: Fix for Dialog-Info notification ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68581 68581]
|}
Send an empty dialig-info XML after inbound subscription.<br/>Required for interop with Grandstream GXP2010.<!---->
   
=== SIP: Problem decoding INFO(application/dtmf-relay) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68667 68667]
|}
DTMF digit was not decoded from message body if whitespace between EQUAL and DIGIT.<br/>E.g. Signal= 5<!---->
   
=== Phone: Changing config option /sip-hold does not call for reset ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68691 68691]
|}
Reset is required and 'reset required" must be displayed.<!---->
   
=== Kerberos: Protect against ping pong attacks ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68822 68822]
|}
Do not answer with an error message to unexpected or malformed messages.<br/><br/>This protects against the "Kerberos Server Spoofed Packet Amplification DoS" attack. The attack causes two Kerberos servers to send each other error messages in a ping pong style.<!---->
   
=== Potential Trap because of recursive loop, if "incomplete" deastination used at a Node to invalid name/number ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68862 68862]
|}
Check for loop implemented (merge from v10, v9)<!---->
   
=== H.450: Bad encoding of DivertingLegInformation4 arguments ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=68868 68868]
|}
DivertingLegInformation4 content coding was wrong.<br/>Wireshark displayed it as malformed.<br/><br/>Note:<br/>This fix causes interoperability problem with phones with older (non-fixed) firmware versions!<br/>Phones also require an updated firmware if PBX is updated.<!---->
   
=== PBX: Phone config was not sent to phone, if phone was power cycled shorty after registration ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69280 69280]
|}
The new registration after the power cycle was not detected as new registration but as re-transmission of the previous registration, so it was not reported to the PBX and no phone config was sent<!---->
   
=== SIP: NOTIFY sent after 302 moved temporarily ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69282 69282]
|}
After processing "302 moved temporarily" on an outbound call a NOTIFY (sipfrag) was sent.<!---->
   
=== IP-DECT: New radio BMC firmware PCS05Ak ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69468 69468]
|}
The new radio BMC firmware PCS05Ak for the IP1200 fixes a trap by the DECT system if more than 255 DECT users without an endpoint subscription are sent to it.<!---->
   
=== PBX: Reject calls without media, if no known facility ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69477 69477]
|}
Fixes compatibility issues between versions. For example presence subscription sessions from v8 phones being forwarded to voicemail<!---->
   
=== PBX: Filter for internal or external calls at CFs did not work CFB or CFNR if call already diverted ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69483 69483]
|}
Problem:<br/><br/>User A has CFU to User B<br/>User B has CFNR for ext. Calls only to User C<br/><br/>An internal call to A was diverted to B (ok) and after no response diverted to C (nok)<!---->
   
=== PBX Waiting: No ringback when doing two-stage dialing to a Gateway/Trunk object ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69531 69531]
|}
A local ringback is now switched on, when receiving ALERT from called party<!---->
   
=== phone: assume an outbound call to be an external call if connected number info is missing in connect event ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69581 69581]
|}
In certain ISDN configurations the PBX can not provide the connected number info in the connect event for an outbound call. In this case the the call was assumed to be an internal call and consequently was not recorded when transparent recording of external calls was configured.<br/>Now an external call is assumed in this case.<br/> <br/><!---->
   
=== phone: VLAN signaling priority could not be configured via phone menu ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69633 69633]
|}
Under "Menu/Administration/IP Settings/VLAN" there was only a "VLAN Priority" menu item. This menu item did override the 'Priority RTP Data' value but not the 'Priority Signaling' value as entered via WEB configuration.<br/>Now the items "Prio. RTP Data" and "Prio. Signaling" replace the "VLAN Priority" item.<!---->
   
=== IPxx10-sata: trap after config /trace /track activation ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69642 69642]
|}
Instruccion was accessing uninitialized pointer.<!---->
   
=== IP6010: RSTP did not work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69731 69731]
|}
When connecting ETH0 in RSTP mode to an HP Pro Curve switch the switch changed the port state to blocked after negotiation phase<!---->
''Status:''
files: mv78x00_drv.cpp, mv78x00_drv.h
   
=== SIP: Trap when handling NOTIFY(application/qsig) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69771 69771]
|}
Traps if no progress indicator present in tunneled DISCONNECT message.<!---->
   
=== IP6010: SRTP using AES-192 and AES-256 did not work ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69828 69828]
|}
Due to a bug in the encryption driver of the IP6010, only AES-128 worked on this platform.<!---->
   
= V8 Hotfix21 (80500.60) =
Changes included in Version 8 hotfix21
[http://mantis.innovaphone.com/view.php?id=69991 Definition]
== New Features ==
   
=== Gateway: Forward Display Info received from ISDN Setup to H.323 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=70562 70562]
|}
needed for compatibility with SecuGATE LI30<!---->
   
=== phone: LED mode of Join Group function key can be set both for idle and for active state ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71247 71247]
|}
sometimes the "not in group" state must be signaled as the exception<!---->
   
=== phone: Mic Off/On controllable via Soap:UserRc(&lt;call&gt;,14/15) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71721 71721]
|}
To allow Soap app's control of the mute key<!---->
=== Other new Features ===
{|
|-
|valign=top nowrap=true|[http://mantis.innovaphone.com/view.php?id=71747 71747]
|valign=top nowrap=true|jfr
|phone_coldfire(OEM device): keypad light and display can be switched off
|}
   
== Bug Fixes ==
   
=== VM, email attachments weren't sent for https URLs ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=69965 69965]
|}
i.e. voicemail wave attachments<!---->
   
=== SIP: Reject unsupported method types with "SIP/2.0 405 Method Not Allowed" ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=70526 70526]
|}
Not ignoring them.<br/><br/>PING sip:tel3@PBX0 SIP/2.0<br/>Via: SIP/2.0/UDP 172.16.77.14:5060;branch=z9hG4bK937906956;rport<br/>From: ;tag=3520474<br/>To: <sip:tel3@PBX0><br/>Call-ID: 193626070<br/>CSeq: 20 PING<br/>Contact: <sip:tel3@172.16.77.14><br/>Max-Forwards: 70<br/>Content-Length: 0<br/><br/>SIP/2.0 405 Method Not Allowed<br/>Via: SIP/2.0/UDP 172.16.77.14:5060;branch=z9hG4bK937906956;rport<br/>From: <sip:tel3@PBX0>;tag=3520474<br/>To: <sip:tel3@PBX0><br/>Call-ID: 193626070<br/>CSeq: 20 PING<br/>Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH<br/>Content-Length: 0<br/><br/><!---->
   
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=70675 70675]
|}
After closing regstration Dectmaster starts another call.<br/>Call is rejected, but signaling enity is deleted before call object.<!---->
   
=== SIP: No route processing if neither Record-Route header nor Contact header is present ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=70971 70971]
|}
Misleading trace message:<br/>  sip_call::process_routing(0xA8) Unsupported transport protocol: sip:user@domain.com;user=phone<!---->
   
=== when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71246 71246]
|}
after a template has been edited unchanged information units inherited from predecessor templates must be removed from the edited template. this did not work for the dialing location and thus a later change in a predecessor template had no effect. <!---->
   
=== SIP: No media after accepting a waiting call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71288 71288]
|}
Call waiting on a phone.<br/>Going onhock while another call is waiting starts ringer.<br/>After going offhook again the waiting call is accepted, but no media in both directions.<!---->
   
=== phone: send config to PBX only when the config was edited on phone ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71387 71387]
|}
A config from an older PBX may contain duplicate elements which are stripped by the phone. I such a stripped config is sent back to the PBX the PBX will return the old config again.<!---->
   
=== SIP: Interop with Nortel CS1000 SIPLine GW ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71426 71426]
|}
Nortel sends 183/Progress with 'sendrecv' answer<br/>followed by UPDATE with 'inactive' offer<br/>followed by UPDATE with 'sendrecv' offer.<br/><br/>Innovaphone SIP stack remains in 'inactive' state.<!---->
   
=== SIP: Interoperability with MX-ONE ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71480 71480]
|}
A semi-attended transfer fails if MX-ONE sends INVITE(Replaces)<br/>instead of 200/OK when connecting a call.<!---->
   
=== SIP: Trap on timer expiration during call release ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71699 71699]
|}
Media negotiation watchdog timer expired after final SIG_REL went to app.<br/>But before app deleted the call object.<!---->
   
=== phone: display info provided by SETUP or CONNECT was ignored ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71727 71727]
|}
only the display info provided by an INFO event was handled  <!---->
   
= V8 Hotfix22 (80500.61) =
Changes included in Version 8 hotfix22
[http://mantis.innovaphone.com/view.php?id=71744 Definition]
== New Features ==
   
=== Debug information on assertion ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71961 71961]
|}
More debug information on default event handler.<!---->
   
=== SIP: Get display information from Call-Info header in register response ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72448 72448]
|}
Get display information from Call-Info header in 200/OK <!---->
   
=== PBX: Forward original received ISDN display element to picking up or forwarded call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=73278 73278]
|}
In the display element from ISDN there could be vital information from equipment like crypto gateways. This should be available also if the call was picked or forwarded.<!---->
   
== Bug Fixes ==
   
=== TCP: Roundtrip measurement wrong in case of packet loss ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71985 71985]
|}
In case of packet loss, way to high round trip values were measured. If the packet-loss was to high, this could result in a constantly increasing re-transmission timeout value.<!---->
   
=== SIP: Trap on IP-DECT when re-configuring PBX link ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72190 72190]
|}
85:2195:425:7 - REG_PRI.4 default(8102be48): serial_timeout<br/>85:2195:425:7 - Assertion failed line 748 in common/os/os.cpp, object deleted<br/><!---->
''Status:''
Merged to 09-80500
   
=== Scheduling improved to avoid processes not being scheduled during long flashman operations ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72243 72243]
|}
In version 7 it could happen, that IP and other processes were not scheduled any more during periods of long flashman operations (e.g. bootcode update or reorganizing flash).<br/><br/>In version 8 and higher there was already a fix for this problem, but this included special handling of the flashman priority level, which was not a good solution even if it worked.<!---->
   
=== SIP: Cleanup failed (resources leaking) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72284 72284]
|}
Call and channel objects were not freed sometimes<br/>when INVITE was followed by CANCEL very fast.<!---->
   
=== PBX SOAP: Called Number presentation not correct for calls to 'local' objects ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72396 72396]
|}
If an object is marked as local, the PBX prefix should not be included in the called number.<br/><br/>This is a fix, which is merged from v9 and higher back into v8<!---->
   
=== update - scfg command could hang when the HTTP session was broken or prematurely closed by the server ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72708 72708]
|}
in consequence update script processing was stopped until reboot<!---->
   
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72729 72729]
|}
When Dectmaster registers user at PBX using SIP protocol<!---->
   
=== PBX: Called Name displayed when calling an object with forking was wrong ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72735 72735]
|}
The name of the forking destination was displayed instead of the name of the called object<!---->
   
=== PBX: No Audio if call thru Waiting Queue DTMF destination, was transfered to BC-Conf ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=72746 72746]
|}
Problem caused by call state management error in PBX for calls connected without alert if alert was received later<!---->
   
=== SIP: Memory leak during transfer ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=73003 73003]
|}
Occured on internal testing only (002-conf-with-bcast.xml)<!---->
   
= V8 Hotfix23 (80500.62) =
Changes included in Version 8 hotfix23
[http://mantis.innovaphone.com/view.php?id=76312 Definition]
== New Features ==
   
=== PBX-SOAP: UserHold without MOH to local User ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=75577 75577]
|}
UserHold was sending MOH to the local and the remote User. With the argument remote=true, the MOH is sent to the remote user only<!---->
   
== Bug Fixes ==
   
=== ISDN Trunk: Transfer to ISDN Trunk with TONE interface failed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=73695 73695]
|}
There was not media after the transfer<!---->
   
=== SIP: Using wrong remote port when registering ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=73784 73784]
|}
Only affects IP-DECT when handset is switched OFF and ON and if the SIP runs on non-standard port.<!---->
   
=== SIP: Handling of collision of transfer and release ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=73936 73936]
|}
If one end releases a call while the other initiates an attended transfer, a "ghost call" may remain.<br/>Resource leak.<br/><!---->
   
=== SIP: Handling of P-Alias header was wrong ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=74061 74061]
|}
Interop of v8/v7 clients with v9 PBX:<br/>New alias type "2" in P-Alias was taken as numeric.<!---->
   
=== ISDN: Send HLC with mobility calls ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=74296 74296]
|}
Some ISDN networks refuse the forwarding of a call to a mobile network if no HLC (High Layer Compatibility) Information Element indicating Telephony is included in the call.<!---->
   
=== PBX: CF at Gateway Type objects - additional dialed digits should be added to the destination ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=74348 74348]
|}
This way a CFNR at a trunk object can be used to reroute the call to another trunk.<!---->
   
=== IP6000 crypto driver: Trap when buffers are depleted ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=74935 74935]
|}
Avoid the trap and log an Event when the buffers are depleted.<!---->
   
=== TLS: Flow control for incoming data ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=75004 75004]
|}
The TLS socket has to wait for the application to process incoming data before sending the next RECV.<!---->
   
=== VM: &lt;pbx-upd-obj type="cfu"..&gt; without effect when invoked multiple times ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=75121 75121]
|}
Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.<!---->
   
=== SIP: Send "305 Use Proxy" if INVITE is received from unexpected source ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=75380 75380]
|}
Applies to registered interfaces only (e.g. phones).<!---->
   
=== TLS: Duplicate alert message on malformed ClientHelloV2 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=75509 75509]
|}
Only one alert should be sent per session.<!---->
   
=== TLS: Improved negotiation of protocol version ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=75510 75510]
|}
TLS server unnecessarily rejected ClientHello messages with TLS 1.1 and higher. Instead of rejecting it should tell the client that it wants to use TLS 1.0.<!---->
   
=== TLS: Skip empty records ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=75511 75511]
|}
TLS record layer should ignore records with zero length without doing anything.<!---->
   
=== SIP: Memory leak in SIP stack ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=76059 76059]
|}
If group indications are configured for a user<br/>and SIP phone registers at user object<br/>without subscribing for "dialog-info"<br/>box memory can be exhausted with sip_gpi_ctx objects.<!---->
   
=== Gateway: Conference interface, no voice ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=76419 76419]
|}
The ADSP firmware is changed to version 122. This fixes a bug in the conference interface of IP6000/IP6010/... which results in conference calls without voice in one direction for a single member.<!---->
   
=== AD Replication failed on objects without cn-attribute ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=76473 76473]
|}
AD Replication failed on objects without cn-attribute<!---->
   
= V8 Hotfix24 (80500.63) =
Changes included in Version 8 hotfix24
[http://mantis.innovaphone.com/view.php?id=76666 Definition]
== New Features ==
   
== Bug Fixes ==
   
=== PBX-SOAP: Call initiated by SOAP for softwarephone or IP-DECT was sent as transfered call ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=76962 76962]
|}
The result was that call diversions or busy on ... calls settings were ignored<!---->
   
=== Edss1 Interworking: divertingLegInformation2 didn't contain redirectingNumber ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=77003 77003]
|}
Edss1 Interworking: divertingLegInformation2 didn't contain redirectingNumber<!---->
   
=== RTP: Potential random trap when closing channels ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=77918 77918]
|}
Happens if there is a collision with a received packet and closing of the channel. Window for this is very small, so it should happen very rarely. Probability can increase with high load.<!---->
   
= V8 Hotfix25 (80500.65) =
Changes included in Version 8 hotfix25
[http://mantis.innovaphone.com/view.php?id=78245 Definition]
== New Features ==
   
=== HTTP-Client: MD5-sess authentication ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=77773 77773]
|}
HTTP Digest Authentication with alogrithm=MD5-sess.<br/>Choose the first supported "WWW-Authenticate" line from 401 response headers.<br/><br/>Needed for new versions of IIS.<br/><br/><!---->
''Status:''
http://wiki.innovaphone.com/index.php?title=Support:DVL-Feature_Requests#HTTP_Client
   
== Bug Fixes ==
   
=== IP6010: Wrong timer under high load ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=71001 71001]
|}
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQ´s a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs.<br/>-The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled.<br/><!---->
''Status:''
ip6010.cpp<br/>ip6010.h
   
=== ip6010/3010/1060: Ethernet transmit packet length is sometimes wrong ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=77774 77774]
|}
Sometimes old content of the tx dma descriptor was used by the ethernet MAC.<br/>Now the memory write buffers are drained before enabling the tx dma.<!---->
''Status:''
mv78x00_drv.cpp<br/>mmu.S
   
=== ip6010/3010/1060: Ethernet receive packet sometimes delayed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=77781 77781]
|}
Sometimes the rx descriptor are processed with the next tx event.<br/>Now the rx queue is processed completely in on interrupt.<br/><!---->
''Status:''
mv78x00_drv.cpp<br/>mv78x00_drv.h
   
=== Gateway: Trap when interworking Call Completion ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=78228 78228]
|}
Trap when interworking Call Completion.<br/><br/>LOG CALL 6 A:Call    ->                        / PRI2::->*::<br/>R_CALL free error c18a59b8<br/><!---->
   
=== TLS flow control damaged in versions 7 and 8 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=78377 78377]
|}
The following fix was not good:<br/>#75004: TLS: Flow control for incoming data<br/><br/>Therefore TLS did not work correctly in the following releases:<br/>v7hotfix35 and v7hotfix36<br/>v8hotfix23 and v8hotfix24<br/><br/>No problem in version 9.<br/><!---->
   
=== SIP: Be save against sudden death of SIP caller ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=78460 78460]
|}
Lifetime of an INVITE trasnaction is not limited by any timeout<br/>after provisional response has been send/received.<br/>Sudden death of a caller make calls hang forever.<br/>Now overall lifetime of an INVITE server transaction is limited to 3 minutes.<br/>After expiration fimnal reject response is sent and call is released.<!---->
   
=== IP6000: Traps in DSP driver under high load ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=78591 78591]
|}
under high load timing may change. Checks in driver relaxed to take this into account.<!---->
   
=== SIP: Wrong number of waiting messages (MWI) ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=78890 78890]
|}
MWI: Number of voice messages not decoded from incoming NOTIFY(application/simple-message-summary).<br/>Was either 1 or 0.<!---->
   
=== IP6010/3010/1060/0010: RSTP not working ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=79251 79251]
|}
RSTP packets were sent to but not received from switch port <!---->
''Status:''
checked in to 8.00,09-80500
   
= V8 Hotfix26 (8079900) =
Changes included in Version 8 hotfix26
[http://mantis.innovaphone.com/view.php?id=79737 Definition]
== New Features ==
   
=== Phones: Switch for phoneapp to disable auto-answer ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=80233 80233]
|}
Disable/enable auto-answer support on phoneapp level.<br/><br/><!--<br/>phonesig_if.h<br/>phonesig.cpp<br/>--><!---->
   
== Bug Fixes ==
   
=== IP1060 IP3010 IP6000 IP6010: DSP packet debug didnt show some packets, version endian ,and dsp-trace port was wrong ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=79754 79754]
|}
cleanup<!---->
''Status:''
ac_491.cpp<br/>debug.h<br/>ac_dsp3.cpp<br/>trace.xsl<br/>
   
=== PBX Waiting: Missing ringback on call forward after announcement ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87674 87674]
|}
This was a collateral damage of<br/><br/>fix: #81370: PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
   
=== PBX Waiting: DTMF overlap dialing or blind transfer to same Waiting object was rejected with busy ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87681 87681]
|}
Even if this was caused by a CFB or CFU on the dialed destination<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
   
= V8 Hotfix 28 (80804) =
Changes included in Version 8 hotfix28
[http://mantis.innovaphone.com/view.php?id=82179 Definition]
== New Features ==
   
=== Debug information on assertion ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=81973 81973]
|}
More debug information on default event handler.<!---->
   
== Bug Fixes ==
   
=== HTTP-Server: Configuration of "Public compact flash access" did not work for all cases ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=82064 82064]
|}
E.g. /DRIVE/CF0/Neuer Ordner/ does not work, because HTTP request contains escaped sequences.<br/><br/><!--<br/>http.cpp<br/>--><!---->
   
=== Gateway CDR with '0. 0' charge amount ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=82359 82359]
|}
Should be '0.00' instead<br/><!--<br/>fty.cpp<br/>--><!---->
   
=== H.323:No Media for calls with reverse media to a H.323/SIP exclusive Code Media Relay interface ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=82408 82408]
|}
The execlusive coder/media relay config is used to avoid media negotiation problems with carrier which do not support media renegotiations. In case of a call with reverse media to such an interface, this did not work. This happens for example if a CFNR is configured at a Waiting Queue which redirects a call, which received an announcement from the Queue to such interface.<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== Debug "HTTP_GET LOG_HTTP.1: retry, authentication failed" removed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=82499 82499]
|}
<!-- httpclient_i.cpp --><!---->
   
=== SIP: Trap during call handling ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=82544 82544]
|}
Trap during call handling<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: SRTP key exchange failed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=82616 82616]
|}
Bug in base64 decoding of SRTP key.<br/><br/><!--<br/>sdp.cpp<br/>--><!---->
   
= V8 Hotfix 29 (80807) =
Changes included in Version 8 hotfix29
[http://mantis.innovaphone.com/view.php?id=83649 Definition]
== New Features ==
   
== Bug Fixes ==
   
=== failure of analog ports of ip28 ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=82488 82488]
|}
ip28 analogue ports do not react to incoming calls and hook-off. Problem could only be solved by reset.<!---->
   
=== phone: when scrolling directory search results sometimes one of  the numbers of a contact was not displayed ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=84362 84362]
|}
the tag characters assigned to the different numbers were not included in sort order. <!---->
   
=== SIP: Trap during channel handling ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=84800 84800]
|}
Rare trap when re-assigning channels.<br/><br/><!--<br/>sip.cpp/h<br/>medialib.h<br/>--><!---->
   
= V8 Hotfix 30 (80811) =
Changes included in Version 8 hotfix30
[http://mantis.innovaphone.com/view.php?id=85034 Definition]
== New Features ==
   
== Bug Fixes ==
   
=== AD Replication: Configuration Buffer Increased ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86211 86211]
|}
Was too small for many maps<br/><!--<br/>ldaprep.cpp/.h<br/>--><!---->
   
= V8 Hotfix 31 (80815 ) =
Changes included in Version 8 hotfix31
[http://mantis.innovaphone.com/view.php?id=86427 Definition]
== New Features ==
   
== Bug Fixes ==
   
=== Gateway: #11 could not be dialed on analog interfaces with feature codes enabled ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86819 86819]
|}
This is a featiure code used on DECT systems and it was not disabled on analog interfaces<br/><!--<br/>relayfty.cpp<br/>dtmffty.cpp<br/>--><!---->
   
=== PBX: Trap if a Hold was attempted for a call without media ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86874 86874]
|}
Could be caused by a misbehaving application or voip device<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== (clone of #80623) SIP: Calls may remain in clearing state ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88134 88134]
|}
SIP calls may remains undeleted.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
= V8 Hotfix32 (80816.00) =
Changes included in Version 8 hotfix32
[http://mantis.innovaphone.com/view.php?id=89003 Definition]
== New Features ==
   
== Bug Fixes ==
   
=== PBX: Potential trap when receiving unknown presence activity ===
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98043 98043]
|}
In the respective version unknown activities are mapped to "busy"<br/><!--<br/>fty.cpp - rollback of this change<br/>h450.cpp<br/>--><!---->
   
= V8 Hotfix33 =
Changes included in Version 8 hotfix33
[http://mantis.innovaphone.com/view.php?id=98530 Definition]
== New Features ==
   
== Bug Fixes ==
   
=== SIP: Wrong encoding of proprietary response header ===
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98235 98235]
|}
200/OK for REGISTER delivers endpoint's alias list.<br/>Encoded in proprietary response header "P-Alias".<br/>Encoding specifier was wrong.<br/><br/>Was:<br/>  P-Alias: 2,17,uranus%2Ck%FCmmel<br/>Must be:<br/>  P-Alias: 1,17,uranus%2Ck%FCmmel<br/><br/><!--<br/>sipmsg.cpp<br/>--><!---->
   
=== SIP: SDP version not increased when answering an offer where only media-mode has changed ===
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98739 98739]
|}
If remote side changes from 'sendrecv' to 'inactive'<br/>the SDP answer follows this change of media-mode,<br/>but SDP version was not increased.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: Do not add payload type 13 to media description for fax ===
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id

Revision as of 00:31, 19 April 2013

This is the Firmware V8 Roadmap Document.

The release date of the next Hotfix is planed for the second monday of a month. Please note that this a scheduled and no fix date.

This article is generated automatically. Do not edit! Please see the disclaimer before using the information presented here!


V8 Release

This release adds all kind of classic PBX/TAPI Features Definition

New Features

Other new Features

44735 cb V8 License: Prevent downgrade to V7 if a V8 license needs to be removed from a box
38782 dde PBX Objects: Merge (and eliminate) Quickdial Object into Directory Search Object
1247 gd Busy on ... Calls/Twin Phones for Executive
26462 gd Multiple PBX on a single Gateway (PBX Hosting)
26463 gd PBX Hosting: Phones on private Network registered to PBX on public network
34874 gd Broadcast: Display original diverting Party as well
38147 gd PBX: Route internal calls to ... interface
38357 gd PBX Mobility Object
38358 gd PBX Call Forking
40874 gd Tunneling of sdp thru H.323 to allow sip/video accross locations
42421 gd Voice Recording in medialib
43572 jfr Ethernet Redundancy with RSTP
36000 msc Single Sign On
44096 mst switchboard, italian localization
38471 tac Implementation of Microsoft call forking header in SIP for OCS dual-forking
41652 tac Switch off Gateway CDRs completely
43438 tac OCS Federation
44718 tac FR: repeat=integer URL für webmedia/WQ
10943 tsr Call-Completion interworking to ISDN (CCB, CCNR)
35814 vsi IP72: lcd light per default an
39188 vsi New v8 ringing calls display
39381 gd use H.323 hop_count to stop call loops
19363 mst Operator rework (v8 switchboard)
40288 mst Switchboard, Reverse-LDAP Lookup Support for Estos Metadir
40404 mst Switchboard, Localization
40406 mst Switchboard, Park, De-Park
40636 mst switchboard: Read/Write User Configuration
41109 mst Switchboard, Persist Window Size+Position, Some Column-Widths
41108 mst Switchboard, Forward-LDAP Lookup Support for Estos Metadir
4326 ckl First und Last Diverting party in leg2 Info
26584 ckl determination of large pbx dimensioning
42666 mst operator, freq automatic nightswitch
7304 tac rfc4235 notifications
7650 tac SIP AOC
39234 tac SIP Presence Federation
22337 gd Phone config stored on PBX
46769 jfr display diverting party in addition to calling party on partner and pickup key
42959 mst switchboard, Maximize App' on Call-Connect. Call Info when minimized (a'la Outlook)
27130 tac Presence
53427 tac Failed to write files to CF card


Bug Fixes

Other Bug Fixes

46167 mst switchboard, XP Installation abbrechen, wenn .Net CLR < 2.0.50727.1433
13468 vsi blind transfer und FTY_CT_SETUP info ergeben irreführende Anzeige beim Ziel
44177 vsi incoming calls from SIPGATE carry same public number in both alias/name/number
47401 Trap in RELAY on call completion interworking
54183 jfr unique spelling of "Signaling" in WEB interface
42951 mst switchboard, multi-registration confusions
40375 tac SIP passwords were limited to 15 characters
40870 tac Media negotiation problem on SIP/SIP video calls
33570 tac ip72: wifi roaming and IP address changes thru dhcp after reassoc with new wifi network


V8 Hotfix 6 (80500.20)

Changes included in Version 8 hotfix6 Definition

New Features

Modified interface for OEM password complexity

Status Closed
Id 55087

OEMs can now implement a module for checking password complexity Status: files:
./common/lib/lib.mak
./common/interface/interface.mak
./common/interface/pwd_complex_api.h
./common/interface/pwd_complex_api.cpp
./ascom/pwd_complex/pwd_complex.h
./ascom/pwd_complex/pwd_complex.cpp
./box/command/command.h
./box/command/command.cpp
./dect/users/dectusers.cpp

OEM password complexity for Kerberos users

Status Closed
Id 55091

The Kerberos module can now check the complexity of user passwords if this is implemented by the OEM software. Status: files:
kerberos_db.cpp

Simplified administration UI for some OEMS

Status Closed
Id 55137

Some items in the adminstration user interface can now be hidden by setting special xml-modes (admin-basic,admin-advanced). Status: files:
- ./dect/users/dectusers.cpp
- ./dect/master/dectmaster.cpp
- ./platform/platform.mak
- ./platform/asc_diagnostics_basic.xml
- ./platform/asc_diagnostics_hdr_basic.xml
- ./platform/dect_hdr.xml
- ./platform/eth0_hdr.xml
- ./platform/left_menu.xml
- ./box/httpfiles/reset_hdr.xml
- ./common/platform/ip1201.cpp
- ./box/command/command.h
- ./box/command/command.cpp

Hide some pages and items on admin UI while OEM provisioning is running

Status Closed
Id 55162

While the provisioning module of an OEM is active, special xml-modes are set that can be used to hide items from the administration interface. Status: files:
./ascom/httpfiles/asc_ntp.xsl
./ascom/httpfiles/asc_dectfty.xsl
./common/platform/ip1201.h
./common/platform/ip1201.cpp
./common/service/ntp/ntp.cpp
./dect/fty/dectfty.cpp

IP-DECT OEM location monitor function change

Status Closed
Id 55294

For OEM modules the location monitor is changed. Status: dectmaster.cpp

DTMF feature call completion can be also used for no response

Status Closed
Id 55309

The feature is not only usable after a busy call, but also after a call with no response.

Update client option for short URL

Status Closed
Id 55324

For OEM http server the update client should not append additional options to the update server URL. Status: update.h, update.cpp

SIP: Detect remote party identity change

Status Closed
Id 55329

Remote party update did not work in all cases:
If initial INVITE got no identity header, but re-INVITE contains identity header. Status: sip.cpp/h

IP-DECT OEM configuration options for registration speed

Status Closed
Id 55499

For an OEM PBX it is necessary to configure the user's registration speed to this PBX. Used only in the OEM DECT device. Status: dectmaster.h, dectmaster,cpp.

SIP: Added Microsoft propriatary extension "ms-acceptedby" for OCS compatibility

Status Closed
Id 55510

A forked call that is accepty elsewhere is counted as "missed call" by OCS unless Microsoft specific extension is add to Reason header.
Reason: SIP;cause=200;text="OK";ms-acceptedby="sip:user@domain.com"
According to [MS-SIPRE].pdf

A DHCP client with "/keep on" should not fall back to dicsover mode if the lease is due

Status Closed
Id 55561

"/keep on" forces reusing the remembered lease if no DHCP server is responding after boot. But if the server failed to respond to the final rebind request for a regularly obtained lease a new recovery was started.
Now in this case the lease is used further, a request for the lease and an ARP requests to check if the IP address is not assigned to another device are sent in regular intervals.

SIP: Hide product information in reject responses

Status Closed
Id 55620

Don't be kind to SIP scan tools. Status: siptrans.cpp

Include modes into configuration page of update client

Status Closed
Id 55669

Needed for OEM specific XSL.

Phone: Problems with 'Presence' Fkey

Status Closed
Id 55785

Presence Fkey requires working presence subscription.
Presence subscription may fail from time to time due to several reasons.
Reliable re-establishment is required. Status: phonesig.cpp

Bug Fixes

SIP: Media-negotiation after call transfer failed (no audio)

Status Closed
Id 54442

Re-negotiation after call transfer failed.
Results into no-audio condition. Status: sip.cpp/h

send busy tone from PBX dtmf object for not working cf with diversion filter

Status Closed
Id 54978

If a diversion filter is set on a user and the dialed diversion to the pbx dtmf object is not allowed, a busy tone and a reject cause is now sent by the dtmf object.

IP-DECT Master call list OEM link and call state

Status Closed
Id 55026

For OEM devices the call clear link doesn't work.
Call state for the outgoing party is shown as "off-hook". Status: dectmaster_call.xsl, dectmaster.cpp

No Media event was generated even everything was normal for unanswered CC exec on IP-DECT

Status Closed
Id 55177

Could happen for other traffic cases as well like rejected CC exec Status: dectradio.cpp, media.cpp

Point to Multipoint ISDN Maps need to set Type ISDN for CGPN-Out Map

Status Closed
Id 55184

If not the mapping does not work for some networks and always the default number is used for outgoing calls as calling party number Status: gk.cpp

SIP: Digest authentication is rejected if username contains non-us-ascii characters

Status Closed
Id 55217

Digest authentication is rejected if username contains non-us-ascii characters.
Expected special characters to be URL encoded, but most clients send it UTF8 encoded.

H.323: Cause received with PROGRESS message got lost

Status Closed
Id 55248

This could result in calls to busy subscribers in a QSIG PBX to terminate with "recovery on time expiry" instead of "user busy" Status: h323sig.cpp

SIP: Outgoing call (early, not connected) was not canceled (sometimes) on ISDN interworking scenario

Status Closed
Id 55277

An incoming DISCONNECT with progress indicator did not caused the outgoing SIP call to be canceled. Status: sip.cpp

Gateway: divertingLeg2 was not passed in some cases

Status Closed
Id 55310

divertingLeg2 got lost during re-routing in Gateway.
E.g. routing each call over TONE caused the divertingLeg2 to disappear.

Webdav: Handling of failed TCP when writing to file

Status Closed
Id 55460

Webdav client needs handling of TCP error when writing to file

TEL interface: '#11' not callable if feature codes enabled

Status Closed
Id 55537

If feature codes are enabled for a TEL interface, the number '#11' without anything else can not be dialled.
To fix please submit gateway's general page with the OK button or do a factory reset. Status: config.h, relay_general.xsl

ARP requests/replies returned to the sender should be ignored

Status Closed
Id 55560

It was observed that in WLAN environments broadcasted ARP requests/replies may be received by the sender again. This results in some problems when DHCP checks if an IP address is not used by another device via ARP. Now returned requests/replies are simply ignored.

T.38 doesnt work if the call is transferred from a IP-Phone to a fax device

Status Closed
Id 55569

Affects IP2x IP30x fax gateways, the ipphone needs no update Status: ac_dsp3.cpp
v7:
ac494004.h
ac498004.h

DECT: Trap while initiating blind transfer when using SIP as PBX protocol

Status Closed
Id 55581

0:0246:363:3 - GK-CALL free error 9481a58c
0:0246:363:4 - last free=DECTMASTER-RADIO len=6
0:0246:363:4 - caller=0x943796d0
0:0246:363:4 - HEXDUMP
00000000 - 05 80 38 30 31 31 ..8011
0:0246:363:4 - BUFFER-FREE: obj at 0x9481a574 inconsistent
0:0246:363:4 - HEXDUMP

Fixed in dectmaster.cpp

Kerberos problem with encrypted password data containing null bytes

Status Closed
Id 55692

Encrypted Kerberos passwords that are stored using LDAP may contain null bytes. Therefore they must not be handled as strings but as binary data when reading them. Status: files: kerberos_ldap.cpp

Phone: Make PBX-initiated calls don't look like transferred calls

Status Closed
Id 55784

Do not send CT_SETUP.

"Join Group" function key lost state after a PBX reboot when the phone config was stored on the PBX

Status Closed
Id 55790

The Join Group function key lost it's state and did not work anymore after a PBX reset because the the phone config sent by the PBX after reregistration was not evaluated at the phone again.

flash variables may get lost after reboot (because of an earlier trap in the critical phase of flash garbage collection)

Status Closed
Id 55797

Two valid segments bearing the same data are left back when a fragmented segment is compacted into a new one and the box traps after the new segment has been validated but before the old segment has been marked invalid.
Because of a wrong comparison this situation was not resolved after reboot. Instead of deleting one of the segments the new segment was used until completely filled. Therafter all further allocations failed. This situation could only be cleared by a reset to factory defaults.
Now, if the flash user is permitted to use only one segment (for example VARS on most boxes) the old segment is invalidated and the new compacted segment remains. If the flash user is permitted to use more segments (for example LDAP) the new segment is invalidated because it's not known which of the old segments was compacted.

PBX potential trap when parsing SOAP XML

Status Closed
Id 55812

No child element found in SOAP XML

Possible buffer overrun when reading/writing fat volumn id

Status Closed
Id 55858

There was a possible buffer overrun when reading/writing the fat volumn id.

SIP: Display name contained bad characters in some cases

Status Closed
Id 55891

Uninitialized buffer content presented as name identification.

V8 Hotfix 7 (80500.27)

Changes included in Version 8 hotfix7 Definition

New Features

SIP: Distinctive ring tones

Status Closed
Id 55948

Handling of "Alert-Info: internal".
Triggers special ring tone. Status: sip.cpp

SIP: Send P-Asserted-Identity header in 180/Ringing

Status Closed
Id 56091

Some UAC do not show called party's display name when added to To header by UAS.
We now provide PAI header in provisional responses also containing the called party's display name. Status: siptrans.cpp/h
sip.cpp

Gatway: Call completion interworking on called side did not work

Status Closed
Id 56214

Call completion on called side did not work

Thanks to Georg Hartwig for giving us his precious support during developent! Status: relay.cpp/h
q950.cpp/h
q931.cpp/h
q931_nt.cpp
q931_te.cpp
nt_tbl.tbl
te_tbl.tbl
fty.cpp/h

SIP Interworking: CGPN in display name of From URI

Status Closed
Id 56504

SIP Interworking: Get CGPN from display name of From URI

A DHCP client with "/keep on" should send DISCOVER requesting the last assigned address after boot (not a REQUEST)

Status Closed
Id 56543

In WLAN networks with more than one DHCP Server REQUESTing the last assigned address after boot needs more time to switch to a new server if the server providing this address has gone.

Configuration Option to keep Routes over a PPP interface always active

Status Closed
Id 56711

To guarantee that certain connections are only established over a virtual private network, routes over a PPP interface need to be kept active in routing table even while the PPP interface is down. This is done now by checking
"Configuration/IP/PPP-Config/PPP<n>/Always keep Routes active"
For enabled PPP interfaces which are not up the current routing state (active/skipped) is displayed in addition to the interface state under
"Configuration/IP/Routing"

Bug Fixes

Gateway: Trap if Name Out or other fields with very long content

Status Closed
Id 55941

A buffer overrun could happen if very long strings were used as input values Status: gk.cpp

PBX: Unknown filter did not work anymore in version 8

Status Closed
Id 55944

The unknown filter could be configured, but was not applied to calls made by endpoints registered as unknown. Status: pbx.cpp

Firmware update failure on ip4001

Status Closed
Id 55981

On the IP4001 the hwbuild string is computed using the boot flags to see if the box is in production mode. This causes a flash access conflict if the info screen is shown during a flash write ( firmware upload ). Status: cpu.cpp cpu.h

Gateway: Overlap Dialing routes did not work as expected

Status Closed
Id 56006

- sometimes '#' was added to the outgoing call even if 'Add #' was not configured
- enbloc calls were terminated by a route with '.' as incomplete if not enough digits, even if matching routes followed Status: relay.cpp, gk.cpp

IP2x IP30x: Missing tones on BRI interface with SIP implementations that send RTP prior to coder negotiation

Status Closed
Id 56010

This is the problematic scenario:
The IP302 BRI interface is registered on a SIP proxy.
An outgoing call is placed, the SIP proxy sends a STATUS 180 Ringing without SDP information.
The remote side sends RTP data (with inband information) to the IP302.
This switches off the IP302 generated tone, but the remote tone is cannot be used since the SDP is missing in the STATUS 180 message.

Now we ignore RTP with unknown coder for switching off the tone. Status: ac_dsp3.cpp

SIP: Switch to fax did not work in some cases

Status Closed
Id 56076

Sometimes switch to audio occured immediately after switch to t.38 Status: sip.cpp

Call Completion on Busy to diverted destination failed

Status Closed
Id 56243

with the call rejection no informtion about the final destination (leg1 info) was sent, so the call completion was tried with the original called destination. Status: pbx.cpp

PBX: Multiple mobility destinations with delay not handled optimal

Status Closed
Id 56302

- if no local phone was registered, all mobility destinations were called right away. Now the destination with the shortes delay is called right away and the others later according difference in delay

- if local phone was busy the mobility destinations was only called after delay. The one with the shortes delay should be called first and then the others. Status: pbx_mobility.cpp, pbx_mobility.h, pbx.cpp, pbx_api.h

PBX: Groups could not be configured for objects with empty PBX setting

Status Closed
Id 56307

Empty PBX setting means the object is handled as it has the local PBX set. So the local groups should be selectable Status: pbx_admin.cpp, pbx.cpp

Always allow local authentication in boot mode

Status Closed
Id 56396

As Kerberos does not work in boot mode, the disable local authentication flag must be ignored there. Status: Files: command.cpp

SIP: Switch to t.38 was answered with audio instead of 488 reject

Status Closed
Id 56404

In case t.38 is not enable, a switch to t.38 was not rejected with 488.
SDP answer with currently active audio coder was send instead.

PBX: Errors when creating or changing Mobility objects were not displayed

Status Closed
Id 56411

If an error was detected (e.g. duplicate number) saving of the object was prohibited, but no error message as for other objects was displayed Status: pbx_edit_mobility.xsl

PBX-SOAP: Admin function could not be used to configure some new parameters

Status Closed
Id 56419

like phone-config, description, ... Status: pbx.cpp, pbx.h

IP-DECT R-key handling for OEM protocol

Status Closed
Id 56469

The R-key for an OEM protocol does not work.

Support for packetization up to 80ms

Status Closed
Id 56566

60ms was the limit before Status: h323ch.cpp

IP-DECT FTY with TSIP and SIPS

Status Closed
Id 56580

The feature codes do not work with TSIP, the local cf does not work with TSIP and SIPS.

IP-DECT: No Audio was received during call waiting

Status Closed
Id 56616

This was another collateral damage from

fix: #55177: No Media event was generated even everything was normal for unanswered CC exec on IP-DECT

Changing the do-not-disturb user setting has no effect if do-not-disturb function key configured and present

Status Closed
Id 56743

problem: Changing the do-not-disturb user setting has no effect if do-not-disturb function key configured and present

solution: fixed in code

files: phone/user/phone_user.cpp

products: all IPxxx telephones

risks: none

V8 Hotfix 8 (80500.28)

Changes included in Version 8 hotfix8 Definition

New Features

Gatway: Do not pass through SRTP key if "Enable SRTP" not activated

Status Closed
Id 55767

Pass through SRTP key only if "Enable SRTP" is activated Status: channel.h
sip.cpp
gk.cpp
h323ch.cpp

PBX: Only 8 IP Filters possible, no indication if maximum reached

Status Closed
Id 56764

Number increased to 32. If 32 Filters are configured no field to enter a new one is displayed Status: pbx.cpp, pbx.h, pbx_api.h, pbx_admin.cpp, pbx_global.xsl

PBX: Filters to even restrict registration with password

Status Closed
Id 56888

The existing filters only restricted registration to the PBX without password. Now in addition to this registration with password can be restricted as well. Status: pbx.cpp, pbx.h, pbx_api.h, pbx_admin.cpp, pbx_global.xsl, pbx_admin_hdr.xml

DTMF facilities: new MWI modes for an OEM protocol

Status Closed
Id 56953

New modes for message waiting indication added in the DTMF facility module. There are used for an OEM protocol in OEM IP-DECT devices.

SIP: Allow to receive messages larger than 2560 bytes

Status Closed
Id 57081

There was a limitation for incoming SIP messages at 2560 bytes.

IP-DECT: anonymous login; master id checks/traces

Status Closed
Id 57104

For anonymous handsets login additional master id checks and traces added.

make function keys on the phone-ui unmodifiable and unviewable

Status Closed
Id 57212

problem: by setting a function key readonly mask (config change PHONE USER /funclock-ro-mask <mask> or web-ui: Phone->Protect->Function keys not modifiable on the phone-> <mask>), one can now determine a set of function key types which can only be set thru a web-ui and can only be viewed but not modified through phone-ui (see http://wiki.innovaphone.com/index.php?title=Howto:Disable_Function_Key_Modification_On_Phone_UI)

solution: fixed in code

files: phone/user/*

products: all telephones

risks: none

SIP: Use registration's Contact-URI as Request-URI on calls to endpoints only

Status Closed
Id 57300

Registered gateways get a Request-URI containing the destination number

Automated Kerberos configuration triggered by a special VAR

Status Closed
Id 57330

A box can now be advised to join a Kerberos realm by writing an XML-Command to variable CMD0/KCMD. Status: command.h
command.cpp
command.xsl

http://wiki.innovaphone.com/index.php?title=Howto:How_to_configure_Kerberos_using_commands#Automated_Client_Configuration_.28V8_Hotfix8_and_later.29

IP-DECT: Kerberos configuration options for radio device configuration

Status Closed
Id 57339

Now it is also possible to configure the Kerberos client if the radio device in discovery mode is configured by the master. The new feature #57330 is used.

IP-DECT: Messaging options and XML message type support

Status Closed
Id 57413

New configuration page "DECT - Messaging" for the IP-DECT messaging alert signal options. The enable option replaces the IP Master option "Enable messaging to PBX".
The XML message type is supported now. With XML messages it is possible to change the alert signal message dependent.
The message priority can be considered if enabled: the SIP priority "emergency" changes the alert signal to alarm and the priority "non-urgent" changes it to silence.

Decoding of special XML entities

Status Closed
Id 57451

Implement decoding of the following entities: &lt; &gt; &quot; &apos; &amp; Status: files: xml.cpp

IP-DECT: log messages for MSF calls

Status Closed
Id 57512

Log messages for MSF calls added.

IP-DECT: MSF module option disable

Status Closed
Id 57560

With the option /disable it is possible to disable the DECT MSF module.

VM, URL parameter "$_noctl=true" allows to reject control-calls

Status Closed
Id 57571

Control calls may reach a VM object unintentionally. Such calls can now be rejected.

Gateway: If Moh Mode is configured set 'exclusive coder' checkmark as well on UI

Status Closed
Id 57654

The MOH Mode implies that exclusive coders are used Status: relay_edit_phys.xsl

Phone: Show presence note on 'partner' fkey label

Status Closed
Id 57687

Show presence note (if availbale) on 'partner' fkey label.
If no text note is avalable, activity is shown (as usual).

update service 'provision' option to request earlier and faster polling in provisioning mode

Status Closed
Id 57799

In provisioning mode the update service should start polling the update server as soon as possible and not use the default delay.
This can be configured now by

config add UP1 /provision <n>

<n> defines the delay in seconds of the first poll, subsequent polls start after (previous delay * 2) seconds. The maximum delay between polls is 60 seconds.

config add UP1 /provision 0
or
config rem UP1 /provision

switches back to the default or the configured polling interval

Bug Fixes

PBX: Checking if a call matches an pending call-completion request was wrong

Status Closed
Id 56706

If a call completion is pending and the user calls the destination with the pending CC or the user retries successfully the call independent of the pending CC, we want to avoid to signal this CC. For this we match any calls to pending CCs. Sometimes this resulted in matches even if there was none and pending CCs were cleared which shouldn't Status: pbx.cpp

PBX: Trap if duplicate "Long Name" in Database

Status Closed
Id 56774

It may happen that on a replicated PBX temporarily multiple objects with the same Long Name (cn) exist. In the case the PBX restarted. Status: pbx.cpp

PBX: CFNR configured at Waiting not executed correctly on transfer to Waiting

Status Closed
Id 56775

under some circumstances not executed at all and sometime without waiting for No Response Timeout Status: pbx.cpp

Gatway: Suspend/Resume on call completion interworking

Status Closed
Id 56827

Suspend/Resume signaling on call completion interworking did not interwork

PBX Mobility: Trap if call to mobile phone scheduled for recall is cleared and SOAP monitoring is on

Status Closed
Id 56847

If call is put on hold by the mobile phone and then the mobile phone hangs up, the PBX tries to recall the mobile phone. If the held party hangs up in this situation with SOAP monitoring of the mobile phone active, a trap happens Status: pbx_mobility.cpp

Trap on call completion with mobility over dtmf object

Status Closed
Id 56882

When using call completion with mobility over the dtmf object, the PBX crashed.
Now call completion over mobility is rejected.

Disconnect from DTMF/ICP/Directory search object didn't work with mobility

Status Closed
Id 56883

The disconnect from the DTMF, ICP and Directory search objects didn't work with mobility, as it was wrongly called.

PBX Mobility: CLIR did not work correctly

Status Closed
Id 56899

A call was sent without number, but it should have been sent with Number Presentation restricted option set. Status: ep_lib.cpp

SIP: Keep ringing calls longer than 3 min

Status Closed
Id 56901

An INVITE client transaction was canceled 180 secs after "180 Ringing" have been received.

IP-DECT: Load sharing for trunks (OEM protocol)

Status Closed
Id 56942

Load sharing for trunks does not work. It is used for an OEM protocol.

Trap: When handling call completion request from ISDN

Status Closed
Id 57113

Trap: When handling call completion request from ISDN Status: relay.cpp
q931.cpp
pppif.cpp
signal.cpp/h

Qsig Leg2 Info decoding could fail

Status Closed
Id 57126

Qsig Leg2 Info decoding could fail

Protect TLS socket against collision of SOCKET_RECV and SOCKET_SHUTDOWN

Status Closed
Id 57130

It was possible that a collision of SOCKET_RECV from the application and SOCKET_SHUTDOWN from the TLS socket occured. This could lead to a trap because the application was already deleted when the SOCKET_RECV_RESULT was sent. Status: tls.cpp

Missing "Recall possible" text in status line

Status Closed
Id 57196

problem: Missing "Recall possible" text in status line

solution: fixed in call

files: phone/app/app_cc.cpp [box/phone]/forms/[lcd/]forms_gen.cpp

products: all telephones

risks: none


PBX: Call from mobile endpoint could not be picked up with DTMF group pickup

Status Closed
Id 57204

pickup was rejeceted Status: pbx.cpp

v9 Replication Compliance

Status Closed
Id 57274

Fixes addressing UTF-8 conversions

SIP: Some interop tweaks did not work

Status Closed
Id 57354

Some module options did not work after reboot:
/no-hr-notify
/prefer-pai

SIP: Fix for video calls through broadcast user

Status Closed
Id 57504

When initiating a video call towards broadcast user, an offer/offer collision may occur in the PBX.
The PBX must select the video coder (not only audio coder) in this case.

IP-DECT: Pickup, caller id update

Status Closed
Id 57509

Fix for the caller id display update after call pickup.

SIP: Decoding of special Contact-URIs

Status Closed
Id 57523

sip:2031;phone-context=cdp.udp@dpp.nortel:5070;maddr=47.166.92.207;transport=udp
The port information was not extracted from phone-context parameter.
Format used by Nortel only.

SIP: SDP attribut annexb=no was missing

Status Closed
Id 57533

If G.729 Annex B was disabled it must be explicitely announced,
because no mentioning annexb is interpreted as annexb=yes.

Tones: Ringback cadence for Ireland not correct

Status Closed
Id 57545

Ringing tone - Ireland
Freq: 400+450
Cadence: 0.4 on 0.2 off 0.4 on 2.0 off

PBX: Trap when handling presence subscription for VM object

Status Closed
Id 57578

Trap when handling presence subscription for VM object Status: pbx.cpp

Allow dtmf features park/unpark for calls from voicemail object

Status Closed
Id 57582

Currently, calls from the voicemail object to the dtmf object were cancelled, as all calls from non user objects have been cancelled.
Now, the features park and unpark are allowed.

SNMP, ifSpeed wrong

Status Closed
Id 57610

SNMP, ifSpeed wrong

IP-DECT: MSF CLMS messages

Status Closed
Id 57612

Now CLMS messages can be sent with the MSF module.

VM: trailing '#' in CDPN let's diverted call to VM fail

Status Closed
Id 57649

VM: trailing '#' in CDPN let's diverted call to VM fail

Filter did not work correctly with local objects and overlap sending

Status Closed
Id 57652

For checking the filter in case of overlap sending, the number including the Node prefix was used regardless if the node prefix was dialed or not.

automatic or manual recording cannot be stopped if the recorded call is not the currently active call

Status Closed
Id 57685

Automatic or manual recording could not be stopped if the recorded call was not the currently active call.
If the Redial-key is used to toggle recording this is intended behaviour because otherwise the Redial-key could not be used to transfer the non-recorded active call.
If a 'Recording' function key is used to toggle recording there is no need for this restriction.

Now a 'Recording' function key stops automatically or manual started recording any case.

V8 Hotfix 9 (80500.32)

Changes included in Version 8 hotfix9 Definition

New Features

SIP: Suppress Annex B of G.729 if "Silence Compression" is not enabled at the interface

Status Closed
Id 57540

Suppress Annex B of G.729 if "Silence Compression" is not enabled at the interface

permit to send log messages, alarms and events via HTTPS with and without checking the server certificate

Status Closed
Id 57785

Both for the log server and for the alarm/event forward server HTTPS can be configured now.
But because distribution of certifcates a may be problematic if there is a big number of clients checking the server certificate can be supressed by

config add LOG0 /tls-unchecked

IP-DECT: OEM device GUI

Status Closed
Id 57993

Some little changes for a DECT OEM device for the GUI.

IP-DECT: TONE interface

Status Closed
Id 58041

The tone inferface is added to the IP1200.

product_id 153,154 added

Status Closed
Id 58122

these new IDs are needed for IP152 based phone versions

PBX dtmf group feature marks dynamic in groups

Status Closed
Id 58536

As the PBX dtmf group feature shows all dynamic in and out groups, the displayed name of dynamic in groups will be preceeded with '* ' now.

SIP: Mapping of "403 Forbidden" into "Q.931 Requested circuit/channel not available"

Status Closed
Id 58635

Previously mapped into "Q.931 Call rejected"
Better mapped into "Q.931 Requested circuit/channel not available" in order to trigger re-routing at the Gateway

SIP: Support of P-Called-Party-ID

Status Closed
Id 58748

Get CDPN of incoming SIP calls from P-Called-Party-ID if present.

30s Timeout for dialing too short

Status Closed
Id 58783

When putting someone on hold with 'R' there was a timeout of 30s until the consultation call was terminated. This could be too short to find the one to whom to transfer the call.

The protocol timeout in H.323 (TO302) was increased from 30s to 120s Status: h323sig.cpp

PBX: Don't apply Send Number to Recording calls

Status Closed
Id 58878

For recording it is usually needed to know the real number Status: pbx.cpp

MWI key with configurable DTMF signaling type for message center calls

Status Closed
Id 58980

Some users must force inband DTMF for certain SIP providers but our Voice Mail requires out of band DTMF signaling.
Now the type of DTMF signaling to be used for calls to the message center can be configured at the MWI key.

phone: disable call intrusion via partner key when recording is active

Status Closed
Id 65918

Call intrusion cannot be performed while recording is active:
- recording establishes a 3party conference between local party, remote party and recorder.
- call intrusion establishes a 3party conference between local party and the two remote parties
- recording and call intrusion at the same time would require a 4party conference which cannot be set up because the phone has only 2 DSP coder channels.

Now if any kind of recording is configured call intrusion is neither offered in 'recall' menu nor performed via partner key.

Bug Fixes

Disabling local authentication also turned off module authentication

Status Closed
Id 57863

When Kerberos was configured on a box and the local admin accounts were disabled, logging and PBX administration using PBX users did not work anymore. Status: files: command.cpp

SIP: Transfer handling at Gateway may cause on-way-audio

Status Closed
Id 57906

I some scenarios where REFER is handled at the Gateway to transfer a local media call leg (e.g. ISDN) to any other call leg.

IP-DECT: no digits en-bloc timeout

Status Closed
Id 57925

The timeout of the en-bloc timer is changed for the case that no digits are dialed. This fixes the Aastra PBX block bug.

Resuming TLS sessions did not work correctly

Status Closed
Id 58013

The server now ensures that session IDs are unique by adding a timestamp and a serial number. This increases the size of session IDs from 16 bytes to 24 bytes.

Also IP addresses were not handled correctly by the session cache. Status: tls.cpp

QSIG Call Complettion to MD110 failed

Status Closed
Id 58372

QSIG Call Complettion to MD110 failed

phone directory collating sort order unexpected

Status Closed
Id 58386

The ordinal of the space character was higher than that of any alphameric character, thus for example "Smith Eric" was displayed behind "Smithson Eric".
The ordinal of the space character is now 0.

SIP: Don't send empty P-Asserted-Identity in provisional response

Status Closed
Id 58493

SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.64.32.2:14937;branch=z9hG4bK6728a259
From: ""<sip:850@10exchange.wschneider.com;user=phone>;epid=123A3A4D16;tag=c755636afc
To: <sip:00763773033@10.64.64.1;user=phone>;tag=3908677425
Call-ID: d0248a8c-a324-454b-807a-923c30c1e24b
CSeq: 34 INVITE
Contact: <sip:00763773033@10.64.64.1:5060;user=phone;transport=TCP>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH
Content-Length: 230
Content-Type: application/sdp
Server: (innovaphone IP800/8.00 dvl [tac-1.11108:/8050028/400])
Supported: replaces,privacy,answermode,from-change,100rel,timer,histinfo
P-Asserted-Identity:
P-Sig-Options: Sending-Complete

Invalid duplicate DTMF object caused the PBX to trap

Status Closed
Id 58514

A false config with an invalid DTMF object (name like DTMF#pickup_group) caused the PBX to trap.
Such an object will be ignored now.

Pickup function key display discards leading letter on transferred call

Status Closed
Id 58520

problem: Pickup function key display discards leading letter on transferred call, so the first letter or number of the calling party is always missing

solution: fixed in code

files: phone/app_disp.cpp

products: all telephones

risks: none


trap on late CHANNEL_INIT to relay_media_relay::serial_event()

Status Closed
Id 58524

A null pointer was referenced when a CHANNEL_INIT was passed to an object in closing state

AD-replicator: xml-show-namingcontexts leaks memory

Status Closed
Id 58564

a memory leak occurred every time when clicked on Configuration/LDAP/Replicator(AD)/DN/"Show Options"

Do not disconnect calls to directory search object from master/slave user

Status Closed
Id 58587

Calls from a master/slave user where disconnected by the directory search object.
These calls are allowed now.

Phone: Light up partner fkey even on active state

Status Closed
Id 58589

While the phone itself is in active state (non-idle) a partner fkey lamp did not light up when partner's presence indicate 'on-the-phone' activity.
Only in idle state the lamp indicated that partner is 'on-the-phone'.

SIP: Dialog-Info did not show "confirmed" state

Status Closed
Id 58594

"proceeding" was indicated instead.
Caused Problems on snom phones.

Soap::UserPickup() sometimes didn't work

Status Closed
Id 58665

Soap::UserPickup() sometimes didn't work

Call Intrusion across PBXs did not work (intrude call at slave from master)

Status Closed
Id 58710

There was a fix already for this, but this covered only intrude at master from slave. Status: pbx.cpp
pbx.h

Gateway Routes with CDPN map to number containing '#' did not work

Status Closed
Id 58737

The number starting with the '#' was omitted.

Collateral damage of fix: #56006: Gateway: Overlap Dialing routes did not work as expected Status: gk.cpp

PBX Trunk Object: Incomplete destination did not work for incoming incomplete enblock calls

Status Closed
Id 58755

collateral damage of fix: #54357: PBX Node 'incomplete Number' destination did not work for block dial calls Status: pbx.cpp

DRAM /Firmware upload stops sometimes

Status Closed
Id 58769

Depending on the timing the upload hangs.
Seen with the innovaphone test program and minifirmware Status: servlet_post_file.cpp

Gateway: Trap on early RELEASE from calling side

Status Closed
Id 58780

If the caller stops calling at an early stage, a trap may occur:

0:0806:591:0 - LOG CALL 15 Alloc
0:0806:591:3 - LOG CALL 15 A:Call -> / PRI2::->*::
0:0806:597:0 - LOG CALL 15 B:Call 100->226 / PRI2:5336100:->RP2:226:
0:0806:701:3 - LOG CALL 15 A:Rel 100->226 / PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry
0:0806:712:3 - LOG CALL 15 Media 100->226 G711A,20(0,0,0)/G711A,20(0,0,0) PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry
0:0806:713:7 - LOG CALL 15 B:Alert 100->226 G711A,20(0,0,0)/G711A,20(0,0,0) PRI2:5336100:->RP2:226: Cause: Recovery on timer expiry
0:0806:714:0 - TRAP: 0x10

PBX: Name Identification was not forwarded with forked call

Status Closed
Id 58786

With call forking the original calling name id was not forwarded Status: pbx.cpp

PBX: Trap if 'Escape dialtone from' is configured to a non-existent object

Status Closed
Id 58789

Check implemented to use internal TONE interface in this case Status: pbx.cpp

SIP: re-INVITE without SDP offer was rejected with 504 Server Timeout in 'held' state

Status Closed
Id 58822

re-INVITE without SDP offer was rejected with 504 Server Timeout if received on an inactive session.

SIP: Handling of reject of re-INVITE without SDP offer was incomplete

Status Closed
Id 58824

Handling of reject of re-INVITE without SDP offer was incomplete.
Need to generate dummy offer for app.

send PROGRESS after CALL-PROC to stop 10s T310

Status Closed
Id 58839

sometimes too short to forward a call

IP-DECT: potential trap

Status Closed
Id 58920

Potential trap in DECT devices fixed.
Trap identification:
XCPT: no 2 (TLB load) pc 94273278 ra 94273254 va 0000000c

Gateway: A call counter with name containing blank or other special character created problems

Status Closed
Id 58944

It could be configured, but if another map was added to the same route the config was corrupted Status: gk.cpp

Trap on CF remove while files are deleted

Status Closed
Id 58984

When files are deleted from the CF card and the card is removed or has an error, the box could trap.

Potential trap if routes with DTMF output combined with pause chars (',') are used for calls without channel or out-of-channels

Status Closed
Id 59012

In this situation pause digits are passed to a channel, which does not exits. This causes the trap.
Could also be dialed pause characters on a call-independent signaling. Status: relay.cpp

V8 Hotfix10 (80500.33)

Changes included in Version 8 hotfix10 Definition

New Features

Call Forwarding Function Key with "Apply 'Always' Setting Only" checkmark (CFU Only)

Status Closed
Id 59077

If "Apply 'Always' Setting Only" is checked the Function key toggles onls over the 'Always' (i.e. CFU) entries and keeps other existing diversions untouched.
Thus CFB or CFNR diversions set at the phone or at the PBX are not changed when toggling this key.

SIP: Registration lookup by attribute 'username' of Authorization header

Status Closed
Id 59078

Registration lookup by attribute 'username' of Authorization header (not only on anonymized calls)

x509: Support for DNS names in SubjectAltName extension of certificates

Status Closed
Id 59171

Create self-signed certificates and certificate requests that contain a DNS name in the SubjectAltName extension. Display the DNS name in the certificate details. Status: Files: x509.cpp, x509.h, x509asn1.h, request.xsl, certificate_create.xsl, certificate.xsl, oids_asn1.h

SIP: Support for another Contact-URI parameter in REGISTER

Status Closed
Id 59174

+u.sip!model.ccm.cisco.com

SIP: Interop feature "X-cisco-srtp-fallback"

Status Closed
Id 59198

Required for SRTP sessions

H.323-Q.931-Interworking - display text provided in the Display Information Element of an ISDN Information Message on phone

Status Closed
Id 59506

The text provided in the Display Information Element of an ISDN Information Message was silently discarded. Now it is displayed in the phone status line.

SIP: Interop feature "X-cisco-sis-3.0.0"

Status Closed
Id 59533

Required for SRTP sessions

Debug: Support to identify bad objects

Status Closed
Id 59714

Only mem-clients are allowed be deleted dynamically.

Bug Fixes

H.323 Remote address was not checked for calls coming in on special trunks with non-standard ports

Status Closed
Id 58958

This is no problem which affects innovaphone standard products. It is only for H.323 trunks configured with fixed remote and local address and port. Status: h323sig.cpp

Interworked Control-Calls without Facilities Shall Stop in Relay

Status Closed
Id 59009

Interworked Control-Calls without Facilities Shall Stop in Relay

PBX Exec Object: A number Map object to be used to call exec directly

Status Closed
Id 59066

A number map can be put in exec secretary or direct call groups to call the exec thru this Map Object directly. This did not work for calls from IP Phones, which sent a source name with the call. Status: pbx_exec.cpp

PBX: Trap if using SOAP Version Method if PBX not started

Status Closed
Id 59071

null pointer access happens in this case Status: pbx_xml.cpp

DHCP client: "Wait for selected Server" timeout was not applied after a DHCP restart

Status Closed
Id 59076

When the DHCP client receives a DHCP restart request a timer is setup to trigger the restart. The failure happens when an offer arrives before this timer fires.

SIP: Media negotiation problem when processing INVITE without SDP

Status Closed
Id 59082

Media negotiation problem when processing INVITE without SDP

H.323: Don't send a call-independent-signaling call without facilities

Status Closed
Id 59088

This could happen if a QSIG call was interworked, with facilities we do not support Status: h323_tbl.tbl
h323sig.cpp
h323sig.h
phonesig.cpp

send PROGRESS after CALL-PROC to stop 10s T310 - in ISDN Stack not PBX

Status Closed
Id 59195

sending PROGRESS in the PBX could have some unwanted side effects, like a Cisco Callmanager believing that there is actual in-band media available Status: pbx.cpp
q931.cpp
q931.h
te_tbl.tbl
nt_tbl.tbl
isdn_interop.xsl

H.323 slowstart avoid sending duplicate TerminalCapabilitySet messages

Status Closed
Id 59203

If a media re-negotiation happened on a remote system at a time the local H.245 channel was not even established, it could happen that a sequence of TCS, TCS0 and TCS again was sent to a calling system. This irritated especially a Cisco Call Manager.

This happened for example, if a call was received from the call manager on one PBX, which was routed to another PBX on which a CFNR was configured. Status: h323ch.cpp

H.323 Slowstart media renegotiation did not work if TCS was not yet received

Status Closed
Id 59248

This caused a CFNR not being executed (call was cleared on the original called endpoint, but was not sent to new destination) for calls from Cisco Call Manager Status: h323ch.cpp

PBX Mobility: Filters were not evaluated for mobility calls

Status Closed
Id 59398

Calls from mobile phones thru the mobility object were not affected by filter configurations for the user Status: pbx_mobility.cpp

SNMP, ifSpeed wrong

Status Closed
Id 59504

SNMP, ifSpeed wrong

SIP: Media negotiation problem in some early media scenarios

Status Closed
Id 59711

SIP/H323 interworking problem.
Call was terminated with "504 Server Time-out" and "Recovery on timer expiry (102)" Status: sip.cpp

Phone IP150 - dialling numbers containing asterisks '*' does not work

Status Closed
Id 59768

if in offhook mode the asterisk key is pressed for a short time the key is ignored, if it is pressed longer it is evaluated as mute key.

SIP: Registration refresh interval not parsed from REGISTER response if behind NAT

Status Closed
Id 59826

Registration refresh interval not parsed from REGISTER response if behind NAT.
Wrong handling of 'received' and 'rport' parameters in Via header (RFC-3581). Status: REGISTER sip:talk.arcstel.netpbx5.net SIP/2.0
Proxy-Authorization: Digest username="1295_1",realm="talk.arcstel.netpbx5.net",nonce="12935856813:4d1dfa2cd75027df50e51d433f90d3a6",response="09e7e72f21d1772b12b73dffb5b51e3c",uri="sip:talk.arcstel.netpbx5.net",qop=auth,cnonce="b35c9f24e909d311",nc=00000001,algorithm=MD5
Via: SIP/2.0/UDP 192.168.0.34:2057;branch=z9hG4bK-E9764661;rport
From: <sip:1295_1@talk.arcstel.netpbx5.net>;epid=00013e01b12b;tag=847121008
To: <sip:1295_1@talk.arcstel.netpbx5.net>
Call-ID: fc72cde0e909d3119b2500013e01b12b@192.168.0.34
CSeq: 1001 REGISTER
Contact: <sip:1295_1@192.168.0.34:2057;transport=UDP>;expires=3600
Content-Length: 0
Expires: 3600
Max-Forwards: 70
User-Agent: (Ascom IP-DECT Base Station/ [4.1.24/4.1.24/IPBS1-A3/4C])
Allow-Events: reg,dialog,message-summary,presence

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.34:2057;received=89.233.254.81;branch=z9hG4bK-E9764661;rport=58537
From: <sip:1295_1@talk.arcstel.netpbx5.net>;epid=00013e01b12b;tag=847121008
To: <sip:1295_1@talk.arcstel.netpbx5.net>;tag=5439c50a
Call-ID: fc72cde0e909d3119b2500013e01b12b@192.168.0.34
CSeq: 1001 REGISTER
Contact: <sip:1295_1@192.168.0.34:2057;transport=UDP>;expires=54
User-Agent: Advoco/5.0.3046
Content-Length: 0

PBX: Call was possible from registration as standby PBX

Status Closed
Id 59844

A standby PBX registers at the active PBX to check if it is alive. This registration could be misused for calls. It could be done with H.323 and SIP. This fix prohibits calls from this registration and allows registration with H.323 only Status: pbx.cpp

Phone - switch off microphone while sending DTMF as voice data, increase volume of DTMF tones sent as voice data

Status Closed
Id 59846

When "Registration x/General/No DTMF Detection" is checked DTMF tones are sent as voice data. Detection of such tones at the receiving side may fail when mixed with microphone input.

PBX CDR records with a size near 1kB or larger were garbled when sent via HTTP

Status Closed
Id 59966

PBX CDR records with a size near 1kB or larger were garbled when sent via HTTP because of an encoding bug. Locally logging worked correct.

Phone: Translation for presence activities

Status Closed
Id 60119 
 Abwesend, Beschäftigt, Mittagessen, Besprechung, Urlaub
  Away, Busy, Lunch, Meeting, Vacation
  Parti, Occupé, Déjeuner, Réunion, Vacances
  Assente, Occupato, Pranzo, Riunione, Ferie
  Ausente, Ocupado, Comida, Reunión, Vacaciones 
  Fravær, Opptatt, Lunsj, Møte, Ferie

PBX: Trap if a call from mobile endpoint was diverted to a waiting queue, with altert Timeout

Status Closed
Id 60161

A NULL pointer access happend in this case while sending the ALERT message Status: pbx_wait.cpp

V8 Hotfix11 (80500.34)

Changes included in Version 8 hotfix11 Definition

New Features

SIP: Interop flag for Avaya: /no-t38-in-initial-offer

Status Closed
Id 59176

config change SIP /no-t38-in-initial-offer
Can be used to suppress T.38 capability indication in initial SDP offer.
A switch to T.38 fax mode may follow, if T.38 is enabled at the interface.

SIP: Add PAI/PPI header to 200/Ok for INVITE

Status Closed
Id 60249

Some SIP servers wants us to send P-Asserted-Identity/P-Preferred-Identity header in final INVITE response.

IP-DECT: number map for incoming calls (OEM)

Status Closed
Id 60294

Number map for incoming calls added for OEM devices.

SIP: Add PAI/PPI header to 181 response for INVITE

Status Closed
Id 60438

To get full identity information of the new remote partner

SIP: Module option /share-local-port

Status Closed
Id 60542

This option forces outbound TCP signaling connection to be bound to the same local port as the signaling interface is listening on.
(In order to make the remote peer do connection reuse)

Bug Fixes

SIP: Handling of re-INVITE w/o SDP offer while in held (inactive) state

Status Closed
Id 60296

A re-INVITE w/o SDP offer while in held (inactive) state must be answered with 200/Ok containing an sendrecv offer (not inactive).

SIP: SRTP re-negotiation failed sometimes

Status Closed
Id 60387

After switching to non-encrypted media (MOH) the re-negotiation for encrypted media failed (on CCM).

PBX: Slave license update period too short

Status Closed
Id 60390

was 100s (v8) or 10s (v7) should be 10min Status: pbx.h

Gateway: Trap on early RELEASE from calling side

Status Closed
Id 60400

Trap when Notification Indicator is received with ALERT while peer call is released already.

IP-DECT: potential trap

Status Closed
Id 60406

Some pointer checks are added to prevent traps.

PBX Waiting object: Problem with announcements from Boolean Object

Status Closed
Id 60421

The announcement worked, but if DTMF dialing to another Waiting object was done, DTMF dialing on this second Waiting object did not work anymore. Status: pbx.cpp
pbx_api.h
pbx_wait.cpp

PBX CF Loop detection indicated loop with CFNR even if there was no loop

Status Closed
Id 60427

A CFNR loop is only detected if the CFNRs are executed because of no registration. The loop was detected with a single Object without registration instead of only detecting the loop if all objects are without registration Status: pbx.cpp

H.323: If INFO was sent with cdpn and kp it could happen that it was forwarded with cdpn in SETUP and kp in INFO

Status Closed
Id 60443

If a call was established by the application (or incoming signaling) without dialing information and then before the TCP connection was established a INFO message was sent with keypad and called-party-number, the call (SETUP) was sent with the called-party-number followed by an INFO with keypad.

This could result in a duplication of the dialed digits.

Only in special OEM scenarios, because keypad is usually not used. Status: h323_tbl.h

editing function keys via WEB interface broken after invalid characters have been entered in an e164 number field

Status Closed
Id 60468

xml syntax characters like < > & entered in a number field were not encoded on output and thus garbled the xml structure

Memory leak when configuring H.323 NAT

Status Closed
Id 60474

Memory leak when configuring H.323 NAT

possible trap with enabled trace flag on CF checkdisc

Status Closed
Id 60513

The box could trap while checking the card, if the trace flag for CF0 was enabled.

PBX/SOAP: Potential trap when disconnecting a mobility call

Status Closed
Id 60538

If a SOAP application (e.g. TAPI) disconnects a call to/from a mobile user, a trap could happen Status: pbx_xml.cpp

PBX DECT System object: DECT parameters got lost, when changing critical flag

Status Closed
Id 60565

The object was written back to flash without the parameters stored by the DECT system Status: pbx.cpp
pbx.h
pbx_api.h
pbx_dect.cpp
pbx_dect.h

PBX SOAP Admin: Critical flag could not be set in object

Status Closed
Id 60568

The attribute "critical" was not allowed Status: pbx.cpp

Ldap Replication, Problems with Percent-Char in Password

Status Closed
Id 60611

Ldap Replication, Problems with Percent-Char in Password

Optional display of text provided in the Display Information Element of an ISDN Information Message

Status Closed
Id 60612

The text provided in the Display Information Element of an ISDN Information Message is displayed at the phone status line.
This may be supressed now by checking "Phone/Preferences/Hide Display Info from ISDN Providers"

SIP: Authentication issue (AVAYA-SM interworking)

Status Closed
Id 60712

Another re-try with authentication required.

Group Indication with a diverting number of zero length caused a encoding error

Status Closed
Id 60715

The number should not be sent at all. This happend if a group indication was to be sent from a call which was diverted by an object without number Status: h450.cpp

PBX Waiting: Don't forward DTMF to announcement source

Status Closed
Id 60838

Announcement source could be a boolean object and DTMF could change the state of the boolean Status: pbx_wait.cpp

IP-DECT: cause code changed

Status Closed
Id 60958

The cause code is changed to "cause unassigned number" if the call is released because no radios are available.

Fix for SIP requests with 10+ header instances

Status Closed
Id 61014

Response to following INVITE request did not returned all Via headers:

INVITE sip:229@192.168.193.181:2058;transport=UDP SIP/2.0
Record-Route: <sip:145bf82@192.168.193.210;transport=udp;lr>
Record-Route: <sip:192.168.193.219:15060;lr;sap=433098584*1*016asm-callprocessing.sar-624908352~1296718381566~-535462628~1>
From: "H323-2" ;tag=8084387dbc40e01d7f4d42da8200
To: <sip:229@localdomain.com>
Call-ID: 8084387dbc40e01d8f4d42da8200
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.193.210;rport;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099903-AP;ft=192.168.193.210~13c4
Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099903
Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199901
Via: SIP/2.0/UDP 192.168.193.219:15070;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199900
Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bK8084387dbc40e01d7f4d42da8200-AP;ft=6565
Via: SIP/2.0/TLS 192.168.193.104;branch=z9hG4bK8084387dbc40e01d7f4d42da8200;avaya-cm-term-reaction=shortcut
Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099899-AP;ft=7355
Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9099899
Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199897
Via: SIP/2.0/TLS 192.168.193.219:15080;branch=z9hG4bKC0A8C1DBFFFFFFFFDEB7B1F9199896
Via: SIP/2.0/TLS 192.168.193.210;branch=z9hG4bK8084387dbc40e01d9f4d42da8200-AP;ft=6565
Via: SIP/2.0/TLS 192.168.193.104;branch=z9hG4bK8084387dbc40e01d9f4d42da8200
Supported: 100rel,histinfo,join,replaces,sdp-anat,timer
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH
User-Agent: Avaya CM/R016x.00.1.510.1 AVAYA-SM-6.1.0.0.610012
Contact: "H323-2" <sip:201@192.168.193.104:5061;transport=tls>
Accept-Language: en
Accept-Contact: *;+avaya-cm-line=1
Alert-Info: <cid:internal@localdomain.com>;avaya-cm-alert-type=internal
History-Info: <sip:229@localdomain.com>;index=1
History-Info: "229" <sip:229@localdomain.com>;index=1.1
Min-SE: 1200
P-Asserted-Identity: "H323-2" <sip:201@localdomain.com>
Record-Route: <sip:145bf82@192.168.193.210;transport=tls;lr>
Record-Route: <sip:192.168.193.219:15061;transport=tls;lr;sap=433098584*1*016asm-callprocessing.sar-624908352~1296718381477~-535462632~1>
Record-Route: <sip:145bf82@192.168.193.210;transport=tls;lr>
Record-Route: <sip:192.168.193.104:5061;transport=tls;lr>
Session-Expires: 1200;refresher=uac
Content-Type: application/sdp
Content-Length: 178
P-Location: SM;origlocname="Interoplab";termlocname="Interoplab"
Max-Forwards: 63

v=0
o=- 1296719515 1 IN IP4 192.168.193.104
s=-
c=IN IP4 192.168.193.105
b=AS:64
t=0 0
m=audio 2564 RTP/AVP 8 18 96
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000

SIP: Do not send INFO(dtmf) before call is connected

Status Closed
Id 61025

Do not send INFO(dtmf) before dialog is in confirmed state.

V8 Hotfix12 (80500.36)

Changes included in Version 8 hotfix12 Definition

New Features

Phone: New config option "Proxy" for SIP registrations

Status Closed
Id 59396

Now DNS names can be specified.
Replaces config option "Primary Server Address".

phone: " reject if busy" option for incoming announcement calls

Status Closed
Id 61412

In some scenarios it's required that announcement calls are not accepted when the phone is busy.

v8 Firmware for IP6010, IP3010, IP1060, IP0010

Status Closed
Id 61522

Version 8 Firmware will be released for the new IP6010 Gateway familiy as part of a hotfix release.

IP-DECT: Abnormal call release error event

Status Closed
Id 61705

Now the DECT Master sends an error event to the event logger every time if an abnormal call release occurs.

new: phonesig api method to restart registration process without deregistration

Status Closed
Id 62165

WLAN phones we need a way to restart a RAS registration when coming back from a out-of-coverage condition to syncronize the handsets and PBX's registration state.

Bug Fixes

IP2x/30x: T.38: Option for high speed data redundancy

Status Closed
Id 60866

to configure this option use
http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl

IP2x/30x: T.38: Calling tone (CNG) detect didnt work

Status Closed
Id 60879

to configure this option use
http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl

IP3xx: Trap if switching a PBX from Standy to Off

Status Closed
Id 60956

This happens because we try to unregister from a CONF interface, which does not exist on the IP3xx platform Status: pbx.cpp

SIP: Trap when receicing provisional response for obsolete INVITE

Status Closed
Id 61035

In overlap dialing scenarios overlapping INVITE client transactions are used.
Same Call-ID, different CSeq and different To-URI.

SIP: Read PAI/PPI header when receiving MESSAGE request

Status Closed
Id 61086

Read PAI/PPI header when receiving MESSAGE request in order to get calling party identity

Phone: Memory leak when deleting SIP registration

Status Closed
Id 61132

Failed to delete registration, but only if trying to delete during state "rgistration failed due to no response from server".

H.450 encoding problem with call-transfer and diverting facilities, if length of number was 0

Status Closed
Id 61222

A zero lenght number cannot be encoded, it must be omited from the message Status: h450.cpp

SIP: Bug in handling of re-direct responses

Status Closed
Id 61264

New remote port was not respected when maddr parameter was present in redirection URI.
E.g.

sip:662@10.0.77.46:4432;user=phone;transport=Tcp;maddr=10.0.77.46;x-mss-call-id=a515c882e909d311874700903306177f%4010.0.77.70

IP2x/IP30x: T38: Missing "no signal indications" on remote initiated T.38 session

Status Closed
Id 61273

This solves a problem with SIP-Provider behing a NAT router on outgoing fax calls.

Critical Flag at DECT System Object disappears

Status Closed
Id 61318

If the DECT system is replicated from the PBX and systems settings are changed on the DECT system, the critical flag on the DECT System object in the PBX is lost Status: dectusers.cpp
dectusers.h

Calls redialled from call list were not set up with CLIR although CLIR was active for the original call

Status Closed
Id 61321

The CLIR setting of the original call was saved in the call list but not applied when the call was redialled from list.

CFNR at PBX object, was executed on call to busy endpoint

Status Closed
Id 61323

should only be executed registration down or no respone at all Status: pbx.cpp

phone: function key Boolean Object with 'Toggle State' checked did not display the correct state sometimes

Status Closed
Id 61368

This happened when the state of the boolean object was toggled from 'manual-on' to 'automatic-off' state at the PBX or by another phone with such a key. It did not happen when with a key where the 'Toggle State' checkmark was not set.

SIP: No overlap sending if 'sending complete' was declared

Status Closed
Id 61472

Do not start overlapping INVITE transaction for new dialing digit if 'sending complete' was indicated for the call.

PBX phone config templates could overrun when a big number of function keys was configured

Status Closed
Id 61476

There was a general 4kB size limitation for attributes read from LDAP directory which was too small for the 'phone' attribute of a config template.

Webdav: Bad encoding of special characters in XML properties

Status Closed
Id 61505

Bad encoding of file/folder names containing special characters.

do not open multiple HTTP sessions when forwarding a big number of alarms in a short time

Status Closed
Id 61527

when alerm forwarding is active the fault handler passed new alarms immediately to the forwarding httpclient and httpclient opens a new session when there is no idle session.

PBX: Boolean Function Key was not updated when joining group

Status Closed
Id 61590

For the Boolean function key it is required to receive Group Indications from the Boolean object, which does not happen if the phone is not member of the group (dynamic out). When joining the group an update should be sent to the phone. Status: pbx.cpp
pbx.h
pbx_gi.cpp
pbx_gi.h
pbx_bool.cpp

Possible to configure use of Feature Codes on Basic Rate ISDN

Status Closed
Id 61620

This configuration option is not useful on ISDN BRIs. In fact it usually results in unexpected behaviour.

This option is removed from the user interface. Status: ip800/platform/config.h
ip24/platform/config.h

IP-DECT: OEM module update function

Status Closed
Id 61671

The update function for an OEM module was changed.

IP-DECT: trap with call transfer

Status Closed
Id 61676

Null pointer trap with call transfer and release event from the DECT side.
Trap identification, IP1200, V8 Hotfix 10:
XCPT: no 2 (TLB load) pc 943fd6d4 ra 94278e9c va 0000000c

PBX-SOAP: Admin function removed password if Object Long Name (cn) was changed

Status Closed
Id 61725

If the cn is changed the object must be identified by guid an the password of this old object is to be used Status: pbx.cpp

PBX-SOAP: Admin function could not be used to configure "phone-config"

Status Closed
Id 61726

"phone-config" was missing in the list of allowed attributes Status: pbx.cpp

SNMP, If Index sometimes missing in interfaces walk

Status Closed
Id 61985

SNMP, If Index sometimes missing in interfaces walk

SIP: Very large SIP request headers were rejected with 414 Request-URI Too Long

Status Closed
Id 62033

SIP request headers larger than 2000 bytes were rejected with 414 Request-URI Too Long

ISDN, QSIG, NT, Invalid Progress message was sent

Status Closed
Id 62190

The mandatory Progress Indicator was missing in Progress message when rejecting a call. This could cause that the inband busy tone could not be sent. Status: nt_tbl.h

V8 Hotfix13 (80500.37)

Changes included in Version 8 hotfix13 Definition

New Features

CAS E1 3bit pulse dialing

Status Closed
Id 62191

Support for CAS E1 3bit pulse dialing, which is sometimes used instead of DTMF addressing.

RPCAP uses system time instead of uptime now

Status Closed
Id 62745

A wireshark capture with RPCAP will now receive packet timestamps with the system time and not the uptime anymore.

Gateway Routing: Support of '?' wildcards in CGPN and CDPN output

Status Closed
Id 62809

In the routing table digits received at places marked with '?' are forwarded to the respective '?' in the output number. This works for CDPN and CGPN maps in routes. It does not work in interface maps Status: gk.cpp
gk.h

Bug Fixes

SIP: INVITE after redirect must not contain the old remote tag

Status Closed
Id 62263

INVITE after redirect did contain the old remote tag.
Now it is cleared before new INVITE is sent to new destination.

SIP: Expect early inband information if 180 with SDP answer is received

Status Closed
Id 62275

Expect early inband information if 180 with SDP answer is received

PBX Quickdial: Transferscenario leaves orphaned call

Status Closed
Id 62311

PBX Quickdial: Transferscerio leaves orphaned call
The orphaned call remains under PBX/Calls and cannot be cleared.

License: License upload shows error "No licenses available"

Status Closed
Id 62318

"No licenses available" when uploading license XML.

Do SRTP Re-keying when doing media renegotiation

Status Closed
Id 62325

Using the same SRTP key could be a security issue. When after a transfer the same SRTP keys are used, in theory the party doing the transfer could still decrypt the SRTP even if not in this call anymore Status: h323ch.cpp
media.cpp
channel.cpp
channel.h

phone: a call unparked by a phone with recording active was released instead of reconnected

Status Closed
Id 62367

When the phone receices the SETUP indicating the unparked call the call should be automatically connected and become the active call. This failed because the currently active call was not put on hold before and thus there was no free DSP cannel to connect the unparked call.

Polish Language could not be configured in the PBX Phone Config

Status Closed
Id 62410

The table entry for polish language was missing

General btree library problem: Potential Trap if many outgoing registrations need to be retried

Status Closed
Id 62428

Actually the problem is in the commonly used btree library, but there are not that many cases in which the libray is used in a way that create the problem Status: btree.cpp

PBX Waiting: Limited DTMF targets could be added using Internet Exporer

Status Closed
Id 62432

URL size limitiation of IE -> use POST instead Status: pbx_edit_waiting.xsl

PBX Waiting: Connected Number handling different from normal Connected Number Handling

Status Closed
Id 62437

This caused different behaviour whether the operator answered the call on a SIP or H.323 phone. In case of SIP the Connected Number was sent, in case of H.323 not Status: pbx.cpp
pbx.h

SIP: Media negotiation failed when interworking with H.323

Status Closed
Id 62439

When calling from H323 to a user with multiple registrations
and the called user accepts on one of its (SIP type) secondary registration,
the media negotiation can fail.

PBX: Progress Indicator in Alert not forwarded by PBX

Status Closed
Id 62483

This could result in in-band info not played at receiving phone in case no progress incator was sent in previous message of same call Status: pbx.cpp

Call Completion to MD110 didn't work

Status Closed
Id 62512

Call Completion to MD110 didn't work

VM, Smtp authentication sometimes in-place, although not required

Status Closed
Id 62571

VM, Smtp authentication sometimes in-place, although not required

SIP: Media negotiation issue

Status Closed
Id 62606

Handling of re-INVITE w/o SDP offer in 'held' state requires change.

PBX: Blind transfer with consultation to mobile endpoint -> Retrieve missing

Status Closed
Id 62638

The caller is put on hold for the consultation, but is not retrieved when the transfer happens. If the caller is SIP, this results in no media sent. Status: pbx.cpp

Possible trap on certain compact flash operations

Status Closed
Id 62703

There has been the possibility of a trap on certain compact flash file operations.
This trap has been fixed.

DHCP client: timeout for response to a REQUEST too small in some case

Status Closed
Id 62709

When the DHCP client REQUESTs an OFFERed address a variable timeout (min 2 seconds) is set up. In the case in question the server always responds to DISCOVERs and REQUESTs with a delay of a little bit more than 2 seconds and thus a new DISCOVER was triggered a short time before the ACK arrived.
To overcome this problem the minimum timeout is changed to 5 seconds which should be enough for any server.

ADSP driver: initialization changed

Status Closed
Id 62869

The ADSP2191 initialization is changed. This fixes some missed voice channels in conference calls.

Diagnostic/Tracing on IP6000: Trace flag on TEL could not be cleared

Status Closed
Id 62914

once set, it could only be cleared with a !config change command Status: tracing.xsl

V8 Hotfix14 (80500.47)

Changes included in Version 8 hotfix14 Definition

New Features

New flash S29GL256P90/S29GL128P90 on IP1200

Status Closed
Id 58643

This flash is used on new IP1200 devices.
Bootcode downgrade to older bootcode is disabled.
If the bootcode is downgraded the bootcode version is shown as 1013.

SNMP, innoColdStart Trap to be sent only after sw failure or button reset

Status Closed
Id 63160

Settlement of a feature request to have the innoColdStart SNMP trap indicate severe reboot reasons only.

DECT: GUI password input limit info

Status Closed
Id 63349

The user password is truncated to 15 signs. Now the input field is limited and an info is shown.

support for external ringer unit

Status Closed
Id 63358

some special purpose phones may be equipped with an external ringer unit. the information controlling the internal ringer is now passed to the module controlling the external ringer unit.

Bug Fixes

H.323: Don't send a call-independent-signaling call without facilities and user-user information

Status Closed
Id 62961

This fix is related to the fix #59088.
A call-independent-signaling call without facilities should not be sent, but if it has got a user-user information, it should be sent.
This fixes the DECT messaging problem on the IP1200. Status: h323sig.cpp

TCP: Ack was not sent under special conditions with re-transmissions

Status Closed
Id 62965

This could cause the breaking of a TCP connection in case of packet loss, even if the packet loss was not too bad Status: tcp.cpp

Trap when processing webdav requests

Status Closed
Id 62980

Trap when webdav request session were terminated irregularly.

SIP: Bad encoding of To-URI in INVITE when handling REFER with special chars in user part of Refer-To URI

Status Closed
Id 63030

Refer-To: <sip:+49231395710880_(399)@172.20.173.104>
received with REFER was mangled into
To: <sip:%2049231395710880_(399)@172.20.173.104>
and send in INVITE

HTTP-Server: Closing connection after transaction causes trouble with Webdav client

Status Closed
Id 63045

NetDrive client fails when uploading files Status: http.cpp

Webdav: Bug when handling GET with Range header

Status Closed
Id 63131

When applied on a zero length file this response was returned:

\tHTTP/1.1 206 Partial Content
\tDate: Tue, 12 Apr 2011 14:52:23 GMT
\tServer: innovaphone Virtual Appliance / 9.00 dvl [xxx/1000/0]
\tAccept-Ranges: bytes
\tContent-Type: application/octet-stream
\tContent-Length: 0
\tContent-Range: bytes 0-4294967295/0

Error response "416 Requested Range Not Satisfiable" must be returned instead.

Webdav: Don't keep zero-length files open on server side

Status Closed
Id 63133

In case of large files, NetDrive performes GET operation between PUT0 and PUT.
The actual PUT was rejected with 500 error resonse then.

62879: ISDN, QSIG, NT: No Disc Option can be used to send PROGRESS instead of DISC - fix for this fix

Status Closed
Id 63209

This fix from hotfix13 did only for calls on which a CALL-PROC was sent as well. For calls still in overlap dialing (only SETUP-ACK sent) it did not work Status: nt_tbl.tbl

SIP: Fix for dialog-info notification

Status Closed
Id 63249

NOTIFY for dialog state 'terminated' was missing sometimes.

SIP: Trap when session timer is used

Status Closed
Id 63271

Trap on collision of session timer and call release

SIP: Authentication passwords were truncated

Status Closed
Id 63321

Authentication failed because password was truncated.

SIP: Not accepting calls from alternative proxy

Status Closed
Id 63327

When being registered at a proxy with 2 ip addresses the gateway does not accept calls from the alternative ip address.

V8 Hotfix15 (80500.49)

Changes included in Version 8 hotfix15 Definition

New Features

DECT: Radio firmware for new handsets

Status Closed
Id 63577

The new radio firmware PCS05Ah accepts new handsets with the new IPEI number range.

phone: improved czech display texts

Status Closed
Id 63998

now all texts are translated to czech, previous errors were fixed (translations provided by zakharova@annexnet.cz)

Bug Fixes

PBX: License mechanism changed to allow easy migration to new version

Status Closed
Id 63381

- licences of different versions may be installed
- check for min version
- v8 master can act as license master for v9 licenses
- applications may run on older version Status: inno_lic.cpp
inno_lic.h
pbx.cpp
pbx_api.h
pbx_general.xsl
pbx_edit_loc.xsl

PBX: Trunk - don't retry call to next gateway if wrong number

Status Closed
Id 63386

all gateways registered to a trunk are by definition to the same network, so a rerouting is useless, if the cause indicates that the dialed number was wrong Status: q931lib.h

Command traps in minifirmware on joining or leaving Kerberos realms

Status Closed
Id 63415

Because command does not check if kerberos_client_provider::provider is null.

Files: command.cpp

TEL and PRI1-4 not contained in 'PPP connection port' dropdown menu on ip6010, ip3010 and ip1060

Status Closed
Id 63419

'PPP connection port' dropdown should contain TEL and PRI1-4 Status: ip_config.cpp

ip0010 wizard configures PRI1, gateway/interfaces shows PRI1

Status Closed
Id 63430

PRI1-L1 must be renamed into PRI1-CLK Status: config.h, ip6010.cpp

HTTP-Client: Bad encoding of uri parameter in digest authentication

Status Closed
Id 63469

Uri parameter in digest authentication was not URL encoded

Gateway: Outgoing Call Completion did not work when outgoing call was routed through TONE interface

Status Closed
Id 63517

Outgoing CC request did not went out to ISDN interface.

SIP: Message buffer too small for REGISTER request for re-try with authentication

Status Closed
Id 63539

On some installations a change-of-nonce at server side may cause volatile "Registration down error" on client side.

certain non latin-1 characters entered via WEB interface or provided by an external LDAP Server cause display errors

Status Closed
Id 63591

entering such characters via copy/paste as when editing a PBX object may result in an xml-error when showing PBX objects.
when such characters are provided by an external LDAP Server to a phone the display may get cleared.
Now such characters are transcribed to a single latin1 character or replaced by a '-' if no transscription is available.

Web-UI: PBX password length is limited to 15 chars

Status Closed
Id 63640

Added tooltip and fixed maxlength attribute on input elements.

License: Character encoding problem

Status Closed
Id 63645

Character encoding problem

config download may trap when malformed LDAP config data has been uploaded

Status Closed
Id 63678

a buffer overrun happens on config download when a "mod cmd FLASHDIR0 add-view nnn cn=..." line with a length > 63 characters has been uploaded.

Presence functionality is not available when registered via H323 at a non-innovaphone PBX

Status Closed
Id 63745

Presence operations via H323 are encoded in private facility elements which are unknown to a non-innovaphone PBX. Presence control calls sent to such a PBX may be misunderstood and routed back as normal voice call to the sending phone.
Thus no presence control calls must be sent to such a PBX.

Trap when starting from flash_stick

Status Closed
Id 63752

and flash memory not yet programmed with bootcode Status: ip6010.cpp

SIP: Allocated message size to small for INVITE redirect response (Avaya)

Status Closed
Id 63829

Memory allocation is a bit to tight to fit the message due to many Via headers.

INVITE sip:3003@192.168.150.140:2059;transport=UDP SIP/2.0
Record-Route: <sip:5793d7f@192.168.150.115;transport=udp;lr>
Record-Route: <sip:192.168.150.114:15060;lr;sap=315810451*1*016asm-callprocessing.sar1905633216~1304428214402~-1054885358~1>
Via: SIP/2.0/UDP 192.168.150.115;rport;branch=z9hG4bKC0A896726E7526620194612-AP;ft=192.168.150.115~13c4
Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526620194612
Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194610
Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194609
Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bK0e2106b7388e016424db9a29200-AP;ft=11786
Via: SIP/2.0/TCP 192.168.150.118;branch=z9hG4bK0e2106b7388e016424db9a29200;avaya-cm-term-reaction=shortcut
Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bKC0A896726E7526620194608-AP;ft=12651
Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526620194608
Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194606
Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194605
Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bK0e2106b7388e018424db9a29200-AP;ft=11786
Via: SIP/2.0/TCP 192.168.150.118;branch=z9hG4bK0e2106b7388e018424db9a29200
Via: SIP/2.0/TCP 192.168.150.84;branch=z9hG4bK200_f1774512c29cc2e5cd78966_I2371
User-Agent: Avaya one-X Deskphone AVAYA-SM-6.1.1.0.611023 Avaya CM/R016x.00.1.510.1
Record-Route: <sip:5793d7f@192.168.150.115;transport=tcp;lr>
Record-Route: <sip:192.168.150.114:15060;transport=tcp;lr;sap=315810451*1*016asm-callprocessing.sar1905633216~1304428214355~-1054885362~1>
Record-Route: <sip:5793d7f@192.168.150.115;transport=tcp;lr>
Record-Route: <sip:192.168.150.118;transport=tcp;lr>
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 215
...

IP152: Flash access not working with version 8050047

Status Closed
Id 64009

With fix #58643 16 bit access to spansion flash doesnt work Status: boot_coldfire.mak common.mak flash_coldfire.c

No received cause code should be treated as 'normal clearing'

Status Closed
Id 64043

Was sometimes treated as cause code to do re-routing. This happened esspecially with multiple registrations to v8 gateway object. A call sent successfully to the gateway on the first regsitration was sent again on the second registration after call clearing. Status: q931lib.cpp
relay.cpp

missing response 'reset required' when changing PRIx-Lx config options

Status Closed
Id 64055

changing i.e. the ,NT-Mode' config option didn't show the 'reset required' link button after pressing 'OK'. Status: falc56_drv.cpp, config.h ipac_drv.cpp V9:falc56_drv.xsl

PBX: Transfer Recall timer was not started if destination was ringing after blind transfer

Status Closed
Id 64064

After a blind transfer without consultation to a busy destination the recall timer should be started as soon as the destination is not busy anymore and the call is delivered Status: pbx.cpp

Gateway: Allow interface maps for analog interfaces as well

Status Closed
Id 64068

Was prohibited in the past, but there are uses for this. Status: ip24/config.h

Conference on IP6000 Hardware 200 and lower not working with v8hf14 and v9

Status Closed
Id 64132

The ADSP serial port has been changed from SPORT1 to SPORT0 for the IP6010.
Old IP6000 hardware has the SPORT0 not connected, so now SPORT1 is again used on IP6000.

PBX: Potential Trap on calls to exec, map or waiting object

Status Closed
Id 64135

under some rare circimstances, which are unfortunatly not known, there could be a NULL pointer access Status: pbx_exec.cpp
pbx_wait.cpp
pbx_map.cpp

V8 Hotfix17 (09-80500.55)

Changes included in Version 8 hotfix17 Definition

New Features

QSIG: Avaya expect Progress Indicator with external calls

Status Closed
Id 66074

Avaya uses the Progress indicator 'Interworking with a public network' to identify a call as external. This Progress Indicator is now added for calls from a Number NOT with private numbering plan (which is our way to identify internal calls) Status: q931.cpp

ISDN: New interop flag to forward network provided or checked cli only

Status Closed
Id 66183

Useful if the real calling number is needed and not a number provided by CLIP no screening Status: q931.cpp
q931.h
isdn_interop.xsl

Bug Fixes

SIP: Session refresh was taken as session modification

Status Closed
Id 63310

Local SRTP key was re-calculated after re-INVITE for session refreh was received.
Causes SRTP decode error at remote side.
CUCM scenario

IP6010, IP6000: Use optimized memcpy

Status Closed
Id 64587

Use of load/store multiple and shifts for 32 bit alignment speeds up memcpy by a factor of approx 2

Orginal memcpy
<info product="IP6010" mips="800Mips">
<memcpy bytes="1000000" time="2ms" speed="347.826Mbyte/s"/>
<read bytes="1000000" time="2ms" speed="347.826Mbyte/s"/>
<write bytes="1000000" time="2ms" speed="470.588Mbyte/s"/>
<stack_memcpy bytes="1000000" time="7ms" speed="133.333Mbyte/s"/>
<uncached_memcpy bytes="1000000" time="41ms" speed="24.169Mbyte/s"/>
<aes bytes="1000000" time="135ms" speed="7.373Mbyte/s"/>
<sha bytes="1000000" time="70ms" speed="14.260Mbyte/s"/>
</info>

Optimized memcpy:
<info product="IP6010" mips="800Mips">
<memcpy bytes="1000000" time="1ms" speed="888.888Mbyte/s"/>
<read bytes="1000000" time="2ms" speed="347.826Mbyte/s"/>
<write bytes="1000000" time="2ms" speed="421.052Mbyte/s"/>
<stack_memcpy bytes="1000000" time="7ms" speed="142.857Mbyte/s"/>
<uncached_memcpy bytes="1000000" time="15ms" speed="64.000Mbyte/s"/>
<aes bytes="1000000" time="138ms" speed="7.200Mbyte/s"/>
<sha bytes="1000000" time="70ms" speed="14.285Mbyte/s"/>
</info>

CPU load with the test test/9.00/box/dsp/ip6010 shows approx 1% lower CPU load.
Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate, compared to 9708Kbyte/s with the old memcpy.

With ECC enabled the CPU load was 19% / 21% without SRTP and 31% / 33% with SRTP
With ECC Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate10309 Status: ip6010.mak ip6000.mak arm.mak box/arm/memcpy.S

v8: ip6010.mak, box/box.mak, box/memcpy.S

Incorrect rpcap timestamp after TRACE LOST messages

Status Closed
Id 64915

The RPCAP timestamp (Wireshark) after a TRACE LOST message was incorrect, as the TRACE LOST message contained an incorrect timestamp.

VM, Project script didn't run for endpoints having "Send Number" configured

Status Closed
Id 65456

VM, Project script didn't run for endpoints having "Send Number" configured

Kerberos: Do not allow registration of multiple databases for one realm name

Status Closed
Id 65589

This happened when a box hosted multiple PBXes with the same system name.

files:
kerberos_if.cpp
kerberos_kdc.h (v9 only)
kerberos_kdc.cpp
kerberos_db.cpp

DECT: Trap during registration up handling

Status Closed
Id 65698

Trap in DECT Master fixed. It occurs if the master endpoint is in delete state and a RAS registration up event is received.

MWI does not work in various Node/Pbx combination

Status Closed
Id 65750

MWI does not work in various Node/Pbx combination

Trap: When Dectmaster registers user at PBX using SIP protocol

Status Closed
Id 65798

Occurred on IPBL[4.1.22]

SIP: Fix for SDP answer to SDP offer with "a:inactive"

Status Closed
Id 65863

Interop with CUCM.
Should return RTP/AVP(inactive) if offer was RTP/AVP(inactive).
Not not RTP/SAVP(inactive).

Message Waiting Interrogation: Result message coding wrong

Status Closed
Id 65912

a malformed message was displayed in wireshark Status: h450.cpp
h450asn1.h

SIP: Set CLIR if display string of From-URI contains "Anonymous"

Status Closed
Id 65925

Not only if userpart of From-URI contains "anonymous".

ip6010 - same MAC address was assigned to ETH0 and ETH1

Status Closed
Id 65939

this results in problems when both interfaces are connected to the same LAN segment

PBX-SOAP: Don't provide caller number if CLIR was used on call to monitored endpoint

Status Closed
Id 65944

If this was an internal call, the PBX knows the calling number anyway, but it should not be sent on SOAP Status: pbx_xml.cpp

PBX-SOAP: UserDTMF did not send DTMF to Voicemail or Waiting Objects

Status Closed
Id 65958

It only sent DTMFs to a VOIP connection Status: pbx_xml.cpp

Gateway SIP Interfaces: Could not configure internal registration for a disabled interface

Status Closed
Id 65975

and if a interface was disabled afterwards, the config for the internal registration was lost Status: gk.cpp

SIP: Trap when receicing provisional response with RSeq header

Status Closed
Id 65986

Trap when trying to send PRACK

ip6010 - frame loss on ethernet ports running in a VLAN

Status Closed
Id 66028

receiving of VLAN tagged frames did not work stable, when running ping -t over a longer time a frame loss from 5 to 10 percent was reported

PBX Broadcast: CFNR was executed only after No Response Timeout even if no member

Status Closed
Id 66032

If there is no member in the broadcast group, a CFNR configured at the Broadcast object should be executet immediatelly.

This was a collateral damage from hotfix

65261: PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions" Status: pbx_bc.cpp

IP3010/6010: fax problems

Status Closed
Id 66110
  • CED is not transfered
    * Wrong T38 encoding in V8

Status: ac_dsp3.cpp ( AC491 doesnt want the V21/V22... relay bits set )
config.h ( config.h, X missing, on V9 this parameter is not needed )

PBX: Missing Group Indications when SIP phone is monitoring

Status Closed
Id 66148

If a SIP phone is monitored by another SIP phone,
there are GI's missing if the monitored SIP phone is calling.

DECT: Delete duplicate LDAP 'pbx' <gw> items

Status Closed
Id 66174

Now duplicate LDAP 'pbx' <gw> items are deleted by the DECT users module.

PBX Trunk: Prefix was added to connected number even if no connected number present

Status Closed
Id 66213

The PBX then displayed just the Trunk prefix as remote number on the calls page when the call was connected. Status: pbx_trunk.cpp

PBX-SOAP: FindUser should not show hidden objects

Status Closed
Id 66216

Could be confusing Status: pbx_xml.cpp

IP6010-CF: Kingston compact flash was not recognized

Status Closed
Id 66269

the card was not recognized because a register was wrongly initialized.

SIP: Bug in SDP handling

Status Closed
Id 66274

If value of the session id and version in the o line are zero.

phone: Hexadecimal values instead of descriptive texts were displayed for some rare disconnect causes

Status Closed
Id 66343

"0x57 - unknow cause" was displayed instead of "user not a CUG member". Mainly german descriptive texts were missing.

V8 Hotfix18 (80500.57)

Changes included in Version 8 hotfix18 Definition

New Features

X.509: Add key usage to certificate requests

Status Closed
Id 66413

The Microsoft CA (standard) does not write the key usage into the certificate if it is not specified in the request.

DHCP-client monitors ethernet link down/up events and revalidates current lease after link up

Status Closed
Id 67006

This prevents problems when a device is hot plugged to another network.
Further this helps to overcvome a problem with certain cable modems.

Bug Fixes

SOAP, Send leg2Info.originalCalled Info

Status Closed
Id 66422

As CallInfo.No with type="leg2orig" Status: pbx_xml.cpp

PBX CF Filter for external calls did not work as expected in case of chained CFs

Status Closed
Id 66599

A filter for external calls did not match if the external call was forwarded already by an internal user Status: pbx.cpp

Gateway: Trap in case of collision of hold and clearing from remote

Status Closed
Id 66642

This could happen on gateways with analog interfaces if the R-Key was pressed right when the other side hung up

H.323 potential trap if AlertingNumber is received

Status Closed
Id 66710

is no problem with existing equipment, because we don't know of any sending an AkertingNumber. Could become an problem if we do this sometimes in the future

H.323 Coding error, when forwarding tunneled SDP in some cases

Status Closed
Id 66727

This could happen if during call setup a media negotiation happened on a call with a SIP and a H.323 leg.

This happened for example if a call was received from a SIP Trunk to a Quickdial object in the PBX. The outgoing call from Quickdial could fail because of this.

Release not forwarded in quick dial object

Status Closed
Id 66728

If the called party released the call, the remote party didn't get the release.

possible noise in PRI connections with ip6010 ip3010 ip1060

Status Closed
Id 67302

some few gateways may produce noise when using the PRI ports. This can be fixed with a new CPLD code contained in future firmware. Status: cpld.h

V8 Hotfix19 (80500.58)

Changes included in Version 8 hotfix19 Definition

New Features

ip200a/230/240: handset conversations can be monitored in a directly connected headset

Status Closed
Id 67666

This feature is required for a special application and is supported only for ip200a/230/240 phones with a directly connected headset (non DHSG).
It is enabled via
config add INCA_DSP /handset-spy <volume>
whith <volume> in the range from 1..8

Bug Fixes

IPxx10: error handling in sata driver

Status Closed
Id 67229

Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.

DECT: IP6000/IP6010/... default config Master mode off

Status Closed
Id 67479

Now the Dect Master is in mode off by default for the IP6000/IP6010/...

VM: Trap while processing self-forwarded call

Status Closed
Id 67570

VM: Trap while processing self-forwarded call

SIP: Uninitialized data in SDP offer/answer

Status Closed
Id 67617

Applies to G.726 exclusive calls only.

SIP: Interoperability with Lync and media-bypass

Status Closed
Id 67645

Ack contained wrong To-Tag when calling a lync client in media-bypass scenario.
Results into call drop after 30 seconds.

PBX: Don't forward original diverting_leg2 info if divertion is executed

Status Closed
Id 67686

The leg2 information which is generated when executing an diversion already contains theoriginal called number from previous diversions, so the old leg2 info is not needed anymore. In fact it is harmfull if the call is received by an application only looking at the first leg2 info (e.g. Voxtron)

PBX: License accounting in centralized licensing scenario wrong if master not available

Status Closed
Id 67698

When the master is available the slave stores the licenses from the master including the usage. This stored usage included the licenses used by the slave itself, so if after a reset the master was not available the local usage just added to this.

Now from the stored usage the local usage is subtracted.

PBX Trunk: Problem with Forking to trunk if multiple GWs are registered to Trunk

Status Closed
Id 67720

If one of the gateways rejected the call (no channel, not connected, ...), the original call from which was forked was disconnected

SIP: Fix for early media from Waitng Queue

Status Closed
Id 67775

PROGRESS after ALERT was not handled by SIP stack.
Now 183 Session Progress with SDP is send after 180 Ringing w/o SDP.

H.323: A name_id of length 0 resulted in invalid H.450 coding

Status Closed
Id 67796

An empty name identification received was forwarded in H.323 as invalid H.450. Such a name is now forwarded as 'name not available'.

H.323 Malformed packet

Status Closed
Id 67803

The ASN.1 encoder had a bug under one special condition: For a constrained character string with a maximum length of more or equal to 16bits, with an effective length of zero, the padding for octett alignment was missing for the zero length bitfield containing the string.

In H.323 this only happens for the CallIdentity used for H.450 call transfer message in case of blind transfer without consultation.

This fix breaks compatibility with earlier versions, for this reason this fix is available for version 9,8,7 and 6.

If phones and PBX with versions containing and not containing this fix are mixed the following problems will occur:
- A blind transfer without consultation (initiated with the redial key) is not possible
- A call which was transfered without consultation is not displayed at the transfered-to phone as transfered

SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg

Status Closed
Id 67819

If in incoming SIP was routed to multiple destinations
the final session could be media-relay although not configured.

SIP: DNS problem when SRV response provides no additional records

Status Closed
Id 67907

If 2-step resolving is required (SRV and A) the service port
of the SRV response got lost and default SI Pport 5060 was used.

SIP: Trap when configuring STUN server on a SIP/TCP or SIP/TLS interface

Status Closed
Id 67923

STUN is for SIP/UDP only.

PBX: Master/Slave compatibility problem with version 9 and version 8 and non-ascii characters in PBX name

Status Closed
Id 67956

In version 8 only latin1 characters were allowed, which means in unicode the high byte was always 0. So it could be ignored and when sending location information between master and slave sometimes the high byte contained 0xff.

In version 9 this non-ascii location information was not correct unicode at all.

The problem happened only if non-ascii characters were used when naming a PBX.

PBX: End of call intrusion was not signaled to the phone

Status Closed
Id 68007

The call intrusion tone was generated even if the intrusion was terminated

phone_inca: "ETH0/Isolate PC Link" checkmark could not be cleared via WEB UI once set

Status Closed
Id 68098

Only a WEB UI problem, a "config rem ETH0 /isolate-pc" did help.

SIP: Interoperability with LinkSys SPA3102

Status Closed
Id 68174

LinkSys SPA3102 gives "g729a" as RTP payload type mapping:

v=0
o=- 510843041 510843041 IN IP4 192.168.10.20
s=-
c=IN IP4 192.168.10.20
t=0 0
m=audio 16404 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
...

Needs to be handled.

Gerneral/Admin page was broken if too many authentication servers were configured

Status Closed
Id 68231

The number of authentication servers is now restricted to 10.

phone: intrusion call started in handset mode is not terminated when going on hook when TAPI or operator run on PBX

Status Closed
Id 68249

With TAPI or operator running on the PBX the the signaling of a busy condition is changed such that a disconnect instead of a release is sent. The disconnect was not handled correctly, the hookswitch state was lost and the next on-hook signal was ignored. TThus teh call could be terminated with the disc-key only.

IP-DECT: Adding OEM radios to Kerberos realm did not work with passwords containing special characters

Status Closed
Id 68377

The password was not URL-decoded when reading it from the UI.

DTMF user configuration with invalid checkbox check for presence setting

Status Closed
Id 68383

The check of the checkmark of the presence setting was wrong.

X509: Fix for reading innovaphone info from flash

Status Closed
Id 68435

Parsing the innovaphone info text was incorrect

License: Be safe against factory reset during license invalidation

Status Closed
Id 68447

If factory reset is done before license invalidation procedure is complete,
will keep you from completing the license invalidation.
Now the procedure can be completed even after factory reset.

phone: DHSG headset not reset to idle after a hookswitch signal in idle state

Status Closed
Id 68567

most DHSG headsets generate a hookswich signal and enter voice mode when taken out of basestation. This hookswitch signal was simply ignored.
Now the voice mode is cleared after one second if there is no other DHSG event before.

V8 Hotfix20 (80500.59)

Changes included in Version 8 hotfix20 Definition

New Features

ISDN interop issue with SecuGATE LI 30 from Sirrix

Status Closed
Id 69168

The SecuGATE LI30 is sending/receiving ISDN INFO messages in Call Proceeding State (State 3 and state 9), which was not supported

Allow multiple HTTP IP address filters (allowed stations)

Status Closed
Id 69645

synced from V9 Status: http.cpp
http.h
http.xsl

Bug Fixes

Gateway: Allow configuration of username and password for ENUM/SIP interfaces

Status Closed
Id 68147

For rare where remote destination server asks for authentication.
(And all remote destination servers ask for same auth or remote destination server s always the same.)

SIP/TCP: Transport error when connection is closed by client

Status Closed
Id 68578

If transaction client closes connection before final response has been sent,
the server tries to open a new connection toward ephemeral port of closed connection.

SIP: Fix for Dialog-Info notification

Status Closed
Id 68581

Send an empty dialig-info XML after inbound subscription.
Required for interop with Grandstream GXP2010.

SIP: Problem decoding INFO(application/dtmf-relay)

Status Closed
Id 68667

DTMF digit was not decoded from message body if whitespace between EQUAL and DIGIT.
E.g. Signal= 5

Phone: Changing config option /sip-hold does not call for reset

Status Closed
Id 68691

Reset is required and 'reset required" must be displayed.

Kerberos: Protect against ping pong attacks

Status Closed
Id 68822

Do not answer with an error message to unexpected or malformed messages.

This protects against the "Kerberos Server Spoofed Packet Amplification DoS" attack. The attack causes two Kerberos servers to send each other error messages in a ping pong style.

Potential Trap because of recursive loop, if "incomplete" deastination used at a Node to invalid name/number

Status Closed
Id 68862

Check for loop implemented (merge from v10, v9)

H.450: Bad encoding of DivertingLegInformation4 arguments

Status Closed
Id 68868

DivertingLegInformation4 content coding was wrong.
Wireshark displayed it as malformed.

Note:
This fix causes interoperability problem with phones with older (non-fixed) firmware versions!
Phones also require an updated firmware if PBX is updated.

PBX: Phone config was not sent to phone, if phone was power cycled shorty after registration

Status Closed
Id 69280

The new registration after the power cycle was not detected as new registration but as re-transmission of the previous registration, so it was not reported to the PBX and no phone config was sent

SIP: NOTIFY sent after 302 moved temporarily

Status Closed
Id 69282

After processing "302 moved temporarily" on an outbound call a NOTIFY (sipfrag) was sent.

IP-DECT: New radio BMC firmware PCS05Ak

Status Closed
Id 69468

The new radio BMC firmware PCS05Ak for the IP1200 fixes a trap by the DECT system if more than 255 DECT users without an endpoint subscription are sent to it.

PBX: Reject calls without media, if no known facility

Status Closed
Id 69477

Fixes compatibility issues between versions. For example presence subscription sessions from v8 phones being forwarded to voicemail

PBX: Filter for internal or external calls at CFs did not work CFB or CFNR if call already diverted

Status Closed
Id 69483

Problem:

User A has CFU to User B
User B has CFNR for ext. Calls only to User C

An internal call to A was diverted to B (ok) and after no response diverted to C (nok)

PBX Waiting: No ringback when doing two-stage dialing to a Gateway/Trunk object

Status Closed
Id 69531

A local ringback is now switched on, when receiving ALERT from called party

phone: assume an outbound call to be an external call if connected number info is missing in connect event

Status Closed
Id 69581

In certain ISDN configurations the PBX can not provide the connected number info in the connect event for an outbound call. In this case the the call was assumed to be an internal call and consequently was not recorded when transparent recording of external calls was configured.
Now an external call is assumed in this case.

phone: VLAN signaling priority could not be configured via phone menu

Status Closed
Id 69633

Under "Menu/Administration/IP Settings/VLAN" there was only a "VLAN Priority" menu item. This menu item did override the 'Priority RTP Data' value but not the 'Priority Signaling' value as entered via WEB configuration.
Now the items "Prio. RTP Data" and "Prio. Signaling" replace the "VLAN Priority" item.

IPxx10-sata: trap after config /trace /track activation

Status Closed
Id 69642

Instruccion was accessing uninitialized pointer.

IP6010: RSTP did not work

Status Closed
Id 69731

When connecting ETH0 in RSTP mode to an HP Pro Curve switch the switch changed the port state to blocked after negotiation phase Status: files: mv78x00_drv.cpp, mv78x00_drv.h

SIP: Trap when handling NOTIFY(application/qsig)

Status Closed
Id 69771

Traps if no progress indicator present in tunneled DISCONNECT message.

IP6010: SRTP using AES-192 and AES-256 did not work

Status Closed
Id 69828

Due to a bug in the encryption driver of the IP6010, only AES-128 worked on this platform.

V8 Hotfix21 (80500.60)

Changes included in Version 8 hotfix21 Definition

New Features

Gateway: Forward Display Info received from ISDN Setup to H.323

Status Closed
Id 70562

needed for compatibility with SecuGATE LI30

phone: LED mode of Join Group function key can be set both for idle and for active state

Status Closed
Id 71247

sometimes the "not in group" state must be signaled as the exception

phone: Mic Off/On controllable via Soap:UserRc(<call>,14/15)

Status Closed
Id 71721

To allow Soap app's control of the mute key

Other new Features

71747 jfr phone_coldfire(OEM device): keypad light and display can be switched off


Bug Fixes

VM, email attachments weren't sent for https URLs

Status Closed
Id 69965

i.e. voicemail wave attachments

SIP: Reject unsupported method types with "SIP/2.0 405 Method Not Allowed"

Status Closed
Id 70526

Not ignoring them.

PING sip:tel3@PBX0 SIP/2.0
Via: SIP/2.0/UDP 172.16.77.14:5060;branch=z9hG4bK937906956;rport
From: ;tag=3520474
To: <sip:tel3@PBX0>
Call-ID: 193626070
CSeq: 20 PING
Contact: <sip:tel3@172.16.77.14>
Max-Forwards: 70
Content-Length: 0

SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 172.16.77.14:5060;branch=z9hG4bK937906956;rport
From: <sip:tel3@PBX0>;tag=3520474
To: <sip:tel3@PBX0>
Call-ID: 193626070
CSeq: 20 PING
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH
Content-Length: 0

Trap: When Dectmaster registers user at PBX using SIP protocol

Status Closed
Id 70675

After closing regstration Dectmaster starts another call.
Call is rejected, but signaling enity is deleted before call object.

SIP: No route processing if neither Record-Route header nor Contact header is present

Status Closed
Id 70971

Misleading trace message:
sip_call::process_routing(0xA8) Unsupported transport protocol: sip:user@domain.com;user=phone

when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat

Status Closed
Id 71246

after a template has been edited unchanged information units inherited from predecessor templates must be removed from the edited template. this did not work for the dialing location and thus a later change in a predecessor template had no effect.

SIP: No media after accepting a waiting call

Status Closed
Id 71288

Call waiting on a phone.
Going onhock while another call is waiting starts ringer.
After going offhook again the waiting call is accepted, but no media in both directions.

phone: send config to PBX only when the config was edited on phone

Status Closed
Id 71387

A config from an older PBX may contain duplicate elements which are stripped by the phone. I such a stripped config is sent back to the PBX the PBX will return the old config again.

SIP: Interop with Nortel CS1000 SIPLine GW

Status Closed
Id 71426

Nortel sends 183/Progress with 'sendrecv' answer
followed by UPDATE with 'inactive' offer
followed by UPDATE with 'sendrecv' offer.

Innovaphone SIP stack remains in 'inactive' state.

SIP: Interoperability with MX-ONE

Status Closed
Id 71480

A semi-attended transfer fails if MX-ONE sends INVITE(Replaces)
instead of 200/OK when connecting a call.

SIP: Trap on timer expiration during call release

Status Closed
Id 71699

Media negotiation watchdog timer expired after final SIG_REL went to app.
But before app deleted the call object.

phone: display info provided by SETUP or CONNECT was ignored

Status Closed
Id 71727

only the display info provided by an INFO event was handled

V8 Hotfix22 (80500.61)

Changes included in Version 8 hotfix22 Definition

New Features

Debug information on assertion

Status Closed
Id 71961

More debug information on default event handler.

SIP: Get display information from Call-Info header in register response

Status Closed
Id 72448

Get display information from Call-Info header in 200/OK

PBX: Forward original received ISDN display element to picking up or forwarded call

Status Closed
Id 73278

In the display element from ISDN there could be vital information from equipment like crypto gateways. This should be available also if the call was picked or forwarded.

Bug Fixes

TCP: Roundtrip measurement wrong in case of packet loss

Status Closed
Id 71985

In case of packet loss, way to high round trip values were measured. If the packet-loss was to high, this could result in a constantly increasing re-transmission timeout value.

SIP: Trap on IP-DECT when re-configuring PBX link

Status Closed
Id 72190

85:2195:425:7 - REG_PRI.4 default(8102be48): serial_timeout
85:2195:425:7 - Assertion failed line 748 in common/os/os.cpp, object deleted
Status: Merged to 09-80500

Scheduling improved to avoid processes not being scheduled during long flashman operations

Status Closed
Id 72243

In version 7 it could happen, that IP and other processes were not scheduled any more during periods of long flashman operations (e.g. bootcode update or reorganizing flash).

In version 8 and higher there was already a fix for this problem, but this included special handling of the flashman priority level, which was not a good solution even if it worked.

SIP: Cleanup failed (resources leaking)

Status Closed
Id 72284

Call and channel objects were not freed sometimes
when INVITE was followed by CANCEL very fast.

PBX SOAP: Called Number presentation not correct for calls to 'local' objects

Status Closed
Id 72396

If an object is marked as local, the PBX prefix should not be included in the called number.

This is a fix, which is merged from v9 and higher back into v8

update - scfg command could hang when the HTTP session was broken or prematurely closed by the server

Status Closed
Id 72708

in consequence update script processing was stopped until reboot

Trap: When Dectmaster registers user at PBX using SIP protocol

Status Closed
Id 72729

When Dectmaster registers user at PBX using SIP protocol

PBX: Called Name displayed when calling an object with forking was wrong

Status Closed
Id 72735

The name of the forking destination was displayed instead of the name of the called object

PBX: No Audio if call thru Waiting Queue DTMF destination, was transfered to BC-Conf

Status Closed
Id 72746

Problem caused by call state management error in PBX for calls connected without alert if alert was received later

SIP: Memory leak during transfer

Status Closed
Id 73003

Occured on internal testing only (002-conf-with-bcast.xml)

V8 Hotfix25 (80500.65)

Changes included in Version 8 hotfix25 Definition

New Features

HTTP-Client: MD5-sess authentication

Status Closed
Id 77773

HTTP Digest Authentication with alogrithm=MD5-sess.
Choose the first supported "WWW-Authenticate" line from 401 response headers.

Needed for new versions of IIS.

Status: http://wiki.innovaphone.com/index.php?title=Support:DVL-Feature_Requests#HTTP_Client

Bug Fixes

IP6010: Wrong timer under high load

Status Closed
Id 71001

-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQ´s a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs.
-The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled.
Status: ip6010.cpp
ip6010.h

ip6010/3010/1060: Ethernet transmit packet length is sometimes wrong

Status Closed
Id 77774

Sometimes old content of the tx dma descriptor was used by the ethernet MAC.
Now the memory write buffers are drained before enabling the tx dma. Status: mv78x00_drv.cpp
mmu.S

ip6010/3010/1060: Ethernet receive packet sometimes delayed

Status Closed
Id 77781

Sometimes the rx descriptor are processed with the next tx event.
Now the rx queue is processed completely in on interrupt.
Status: mv78x00_drv.cpp
mv78x00_drv.h

Gateway: Trap when interworking Call Completion

Status Closed
Id 78228

Trap when interworking Call Completion.

LOG CALL 6 A:Call -> / PRI2::->*::
R_CALL free error c18a59b8

TLS flow control damaged in versions 7 and 8

Status Closed
Id 78377

The following fix was not good:
#75004: TLS: Flow control for incoming data

Therefore TLS did not work correctly in the following releases:
v7hotfix35 and v7hotfix36
v8hotfix23 and v8hotfix24

No problem in version 9.

SIP: Be save against sudden death of SIP caller

Status Closed
Id 78460

Lifetime of an INVITE trasnaction is not limited by any timeout
after provisional response has been send/received.
Sudden death of a caller make calls hang forever.
Now overall lifetime of an INVITE server transaction is limited to 3 minutes.
After expiration fimnal reject response is sent and call is released.

IP6000: Traps in DSP driver under high load

Status Closed
Id 78591

under high load timing may change. Checks in driver relaxed to take this into account.

SIP: Wrong number of waiting messages (MWI)

Status Closed
Id 78890

MWI: Number of voice messages not decoded from incoming NOTIFY(application/simple-message-summary).
Was either 1 or 0.

IP6010/3010/1060/0010: RSTP not working

Status Closed
Id 79251

RSTP packets were sent to but not received from switch port Status: checked in to 8.00,09-80500

V8 Hotfix26 (8079900)

Changes included in Version 8 hotfix26 Definition

New Features

Phones: Switch for phoneapp to disable auto-answer

Status Closed
Id 80233

Disable/enable auto-answer support on phoneapp level.

Bug Fixes

IP1060 IP3010 IP6000 IP6010: DSP packet debug didnt show some packets, version endian ,and dsp-trace port was wrong

Status Closed
Id 79754

cleanup Status: ac_491.cpp
debug.h
ac_dsp3.cpp
trace.xsl

PBX Waiting: Missing ringback on call forward after announcement

Status Closed
Id 87674

This was a collateral damage of

fix: #81370: PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2

PBX Waiting: DTMF overlap dialing or blind transfer to same Waiting object was rejected with busy

Status Closed
Id 87681

Even if this was caused by a CFB or CFU on the dialed destination

V8 Hotfix 28 (80804)

Changes included in Version 8 hotfix28 Definition

New Features

Debug information on assertion

Status Closed
Id 81973

More debug information on default event handler.

Bug Fixes

HTTP-Server: Configuration of "Public compact flash access" did not work for all cases

Status Closed
Id 82064

E.g. /DRIVE/CF0/Neuer Ordner/ does not work, because HTTP request contains escaped sequences.

Gateway CDR with '0. 0' charge amount

Status Closed
Id 82359

Should be '0.00' instead

H.323:No Media for calls with reverse media to a H.323/SIP exclusive Code Media Relay interface

Status Closed
Id 82408

The execlusive coder/media relay config is used to avoid media negotiation problems with carrier which do not support media renegotiations. In case of a call with reverse media to such an interface, this did not work. This happens for example if a CFNR is configured at a Waiting Queue which redirects a call, which received an announcement from the Queue to such interface.

Debug "HTTP_GET LOG_HTTP.1: retry, authentication failed" removed

Status Closed
Id 82499

SIP: Trap during call handling

Status Closed
Id 82544

Trap during call handling

SIP: SRTP key exchange failed

Status Closed
Id 82616

Bug in base64 decoding of SRTP key.

V8 Hotfix 29 (80807)

Changes included in Version 8 hotfix29 Definition

New Features

Bug Fixes

failure of analog ports of ip28

Status Closed
Id 82488

ip28 analogue ports do not react to incoming calls and hook-off. Problem could only be solved by reset.

phone: when scrolling directory search results sometimes one of the numbers of a contact was not displayed

Status Closed
Id 84362

the tag characters assigned to the different numbers were not included in sort order.

SIP: Trap during channel handling

Status Closed
Id 84800

Rare trap when re-assigning channels.

V8 Hotfix 30 (80811)

Changes included in Version 8 hotfix30 Definition

New Features

Bug Fixes

AD Replication: Configuration Buffer Increased

Status Closed
Id 86211

Was too small for many maps

V8 Hotfix 31 (80815 )

Changes included in Version 8 hotfix31 Definition

New Features

Bug Fixes

Gateway: #11 could not be dialed on analog interfaces with feature codes enabled

Status Closed
Id 86819

This is a featiure code used on DECT systems and it was not disabled on analog interfaces

PBX: Trap if a Hold was attempted for a call without media

Status Closed
Id 86874

Could be caused by a misbehaving application or voip device

(clone of #80623) SIP: Calls may remain in clearing state

Status Closed
Id 88134

SIP calls may remains undeleted.

V8 Hotfix32 (80816.00)

Changes included in Version 8 hotfix32 Definition

New Features

Bug Fixes

PBX: Potential trap when receiving unknown presence activity

Status Closed
Id 98043

In the respective version unknown activities are mapped to "busy"

V8 Hotfix33

Changes included in Version 8 hotfix33 Definition

New Features

Bug Fixes

SIP: Wrong encoding of proprietary response header

Status Closed
Id 98235

200/OK for REGISTER delivers endpoint's alias list.
Encoded in proprietary response header "P-Alias".
Encoding specifier was wrong.

Was:
P-Alias: 2,17,uranus%2Ck%FCmmel
Must be:
P-Alias: 1,17,uranus%2Ck%FCmmel

SIP: SDP version not increased when answering an offer where only media-mode has changed

Status Closed
Id 98739

If remote side changes from 'sendrecv' to 'inactive'
the SDP answer follows this change of media-mode,
but SDP version was not increased.

SIP: Do not add payload type 13 to media description for fax

Status Closed
Id 98757

Add payload type 13 only to media description for audio

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