Reference8:Release Notes Firmware
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
V8 Hotfix 6 (80500.20)
Changes included in Version 8 hotfix6 Definition
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.
some Compactflash cards not working with innovaphone cardslots
| Status | Closed |
| Id | 55903 |
There are compactflash cards that don't respond 848Ah as General Configuration word to Identify Device Command.
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
V8 Hotfix 7 (80500.27)
Changes included in Version 8 hotfix7 Definition
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
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"
V8 Hotfix 8 (80500.28)
Changes included in Version 8 hotfix8 Definition
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
IP24, IP28: Trap if doing a tranfer for a pickup call
| Status | Closed |
| Id | 56780 |
If a pickup was done (*0#) from an analog phone connected to a IP24, IP28 and this call was transfered afterward (R), a trap happened as soon as the transfer was executed either by R-4 or by hanging up Status: relay.cpp
PBX: Trap if calling a Boolean object from a mobile endpoint
| Status | Closed |
| Id | 56825 |
This could be used to set boolean state from a mobile phone Status: pbx_mobility.cpp, pbx_bool.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.
H.323 registration using a PPTP connection failed
| Status | Closed |
| Id | 56910 |
on PPTP server side wrong IP address was chosen as local address if interface was vonfigured without local address and authentication was used on H.323 Status: h323.h, h323ras.cpp
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
PBX: Diverting number sent in group indications was not adjusted
| Status | Closed |
| Id | 57213 |
The number was sent with all node prefixes
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.
PBX did not send REMOTE_HOLD/RETRIEVE_RESULT
| Status | Closed |
| Id | 57599 |
This causes problems with some third party equipment, which expects these messages Status: pbx.cpp
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.
PBX: IP Filter to restrict Registration do not work for UNKNOWN
| Status | Closed |
| Id | 57653 |
These Filter only work for registrations to configured objects
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.
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: < > " ' & 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
V8 Hotfix 9 (80500.32)
Changes included in Version 8 hotfix9 Definition
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
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.
V8 Hotfix10 (80500.33)
Changes included in Version 8 hotfix10 Definition
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: Do not take P-Called-Party-ID as CDPN (rollback of 58748)
| Status | Closed |
| Id | 59539 |
Rollback of #58748
Do not take P-Called-Party-ID as CDPN.
P-Called-Party-ID contains nothing but the AOR of the receiving interface.
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, Beschftigt, Mittagessen, Besprechung, Urlaub
Away, Busy, Lunch, Meeting, Vacation
Parti, Occup, Djeuner, Runion, Vacances
Assente, Occupato, Pranzo, Riunione, Ferie
Ausente, Ocupado, Comida, Reuni¢n, Vacaciones
Fravr, Opptatt, Lunsj, Mte, 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
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.
V8 Hotfix11 (80500.34)
Changes included in Version 8 hotfix11 Definition
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.
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
| 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)
V8 Hotfix12 (80500.36)
Changes included in Version 8 hotfix12 Definition
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
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.
V8 Hotfix13 (80500.37)
Changes included in Version 8 hotfix13 Definition
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
VM, <pbx-forward>, two display names were sent
| Status | Closed |
| Id | 62505 |
VM, <pbx-forward>, two display names were sent
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.
ISDN, QSIG, NT: No Disc Option can be used to send PROGRESS instead of DISC
| Status | Closed |
| Id | 62879 |
DISC message causes some PBX to release a call right away, so that no in-band busy tone can be played. In this case just to send PROGRESS is better. Status: nt_tbl.h
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
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
V8 Hotfix14 (80500.47)
Changes included in Version 8 hotfix14 Definition
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.
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.
V8 Hotfix15 (80500.49)
Changes included in Version 8 hotfix15 Definition
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
| Status | Closed |
| Id | 63419 |
'PPP connection port' dropdown should contain TEL and PRI1-4 Status: ip_config.cpp
V9 und V8 PPP port configuration geht nicht richtig auf ip6010
| Status | Closed |
| Id | 63427 |
Da erscheint in V9 bei Auswahl von PPPOE das ISDN Menu, dafr ist bei <none> jetzt kein weiterfhrendes Men sichtbar. Das sollte auch bei V8 so sein.
Bei V8 erscheint derzeit auch bei <none> das ISDN Menu, dafr ist bei V8 das PPPOE Men korrekt vorhanden wenn man PPPOE0/1 als Schnittstelle auswhlt.
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
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)
V8 Hotfix17 (09-80500.55)
Changes included in Version 8 hotfix17 Definition
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
| 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.
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
V8 Hotfix18 (80500.57)
Changes included in Version 8 hotfix18 Definition
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.
SIP: Always add payload type 13 (CN) to SDP
| Status | Closed |
| Id | 66735 |
Config file option /add-cn-capability is obsolete and replaced by /rem-cn-capability now.
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
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.
V8 Hotfix19 (80500.58)
Changes included in Version 8 hotfix19 Definition
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.
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
V8 Hotfix20 (80500.59)
Changes included in Version 8 hotfix20 Definition
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.
| 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.
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
V8 Hotfix21 (80500.60)
Changes included in Version 8 hotfix21 Definition
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
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
V8 Hotfix22 (80500.61)
Changes included in Version 8 hotfix22 Definition
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)
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.
V8 Hotfix25 (80500.65)
Changes included in Version 8 hotfix25 Definition
IP6010: Wrong timer under high load
| Status | Closed |
| Id | 71001 |
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQïs 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
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
V8 Hotfix26 (8079900)
Changes included in Version 8 hotfix26 Definition
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
Phones: Switch for phoneapp to disable auto-answer
| Status | Closed |
| Id | 80233 |
Disable/enable auto-answer support on phoneapp level.
V8 Hotfix 28 (80804)
Changes included in Version 8 hotfix28 Definition
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.
Debug information on assertion
| Status | Closed |
| Id | 81973 |
More debug information on default event handler.
V8 Hotfix 29 (80807)
Changes included in Version 8 hotfix29 Definition
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
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
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
PBX: Potential trap when receiving unknown presence activity
| Status | Closed |
| Id | 98043 |
In the respective version unknown activities are mapped to "busy"
V8 Hotfix33 (80819.00)
Changes included in Version 8 hotfix33 Definition
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
V8 Hotfix34 (80820)
Changes included in Version 8 hotfix34 Definition
I6000 IP2000 Allow changing SRTP key while data is queued for encryption
| Status | Closed |
| Id | 100549 |
Bug in the crypto crypto driver. When the SRTP key is changed while a packet is being encrypted the SRTP socket hung up.
NTP-Server: use destination address from client request as source address in response to client
| Status | Closed |
| Id | 100940 |
a response to a client request received via ETH1 was sent with the ETH0 address as source address when routed through default gateway on ETH0.
a response with a source address not matching the adressed server is discarded on client side.
V8 Hotfix35 (80821)
Changes included in Version 8 hotfix35 Definition
PBX: Boot-Loop when replicating objects with wrong password
| Status | Closed |
| Id | 101879 |
When a replication was configured, replicating all objects from a different PBX, with different PBX password then configured already on the box, the box entered a boot loop, which only could be stopped with a long reset.
V8 Hotfix36 (80822)
Changes included in Version 8 hotfix36 Definition
IP230/IP240: Sporadic traps during manufacturing programming
| Status | Closed |
| Id | 71778 |
Inefficient loop during mac address programming
Status:
flash_firmware.cpp
V8 Hotfix37 (80857)
Changes included in Version 8 hotfix37 Definition
Phones: Only one diverting party displayed on incoming calls
| Status | Closed |
| Id | 74406 |
Only original called party displayed, not the last diverting party.
Phones with non-color display only (IP240,IP230,IP200,IP11,IP150)
phone: In Recording Mode 'transparent' or'optional' a 2nd call started by a dialing application could terminate the 1st call
| Status | Closed |
| Id | 114789 |
This happened when a 2nd call was started by a dialing application and then terminated again while the call was in alerting state.
V8 Hotfix38 (80866)
Changes included in Version 8 hotfix38 Definition
phone: In Recording Mode 'transparent' or'optional' internal calls were recorded although 'External Calls Only' was checked
| Status | Closed |
| Id | 114516 |
happened only to outbound calls initiated by some dialing application.
outbound calls initiated directly at the phone and inbound calls were recorded correctly.
V8 Hotfix39 (80868)
Changes included in Version 8 hotfix39 Definition
Kerberos: Admin UI trap when having too many Kerberos hosts
| Status | Closed |
| Id | 116957 |
The problem occured if many Kerberos hosts (~1000) were registered on the server. In this case the box trapped due to an XML encoding problem when opening the page General/Kerberos or PBX/Config/Security.
Was already fixed in v10 and v9hotfix18.
V8 Hotfix40 (80869)
Changes included in Version 8 hotfix40 Definition
FXS: Trap on very rare race collision of retrieve with call release
| Status | Closed |
| Id | 122980 |
If a retrieve happens at the same time as a call release of the held call, a trap could happen. The propabilty of this to happen was very low.
V8 Hotfix41 (80871)
Changes included in Version 8 hotfix41 Definition
wrong activation of non-existent spread-spectrum clock
| Status | Closed |
| Id | 108014 |
happens for all non-ip28 (ip22/24/302/305) gateways if hardware build >= 402, causes the gateways to stall due lack of clocking
V8 Hotfix42 (80872)
Changes included in Version 8 hotfix42 (80872) Definition
Incomplete HTTP responses from HTTP server in certain circumstances
| Status | Closed |
| Id | 138895 |
It might have happened, that the HTTP server closed the underlying TCP connection before all data could be sent.
SIP: Switch from Media-Relay to No-Media-Relay when handling INVITE with Replaces
| Status | Closed |
| Id | 139046 |
Switch from Media-Relay to No-Media-Relay when handling INVITE with Replaces.
May result into no media after INVITE with Replaces.
IP-DECT: Master trap
| Status | Closed |
| Id | 143206 |
There is a Master trap because of an uninitialized variable within a facility call. This is fixed now.
V8 Hotfix43 (81020)
Changes included in Version 8 hotfix43 Definition
SHA-2 hash algorithms
| Status | Closed |
| Id | 113239 |
Port the hash algorithm to our platform.
Support for SHA2 certificates
| Status | Closed |
| Id | 113352 |
- encoding and decoding
* verification
* create such certificates on boxes (except sha224)
Signature algorithms:
* sha224WithRSAEncryption { pkcs-1 14 }
* sha256WithRSAEncryption { pkcs-1 11 }
* sha384WithRSAEncryption { pkcs-1 12 }
* sha512WithRSAEncryption { pkcs-1 13 }
V8 Hotfix44
Changes included in Version 8 hotfix44 Definition