ReleaseNotes9:Firmware: Difference between revisions
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Only happened if debug.xml leak check was used<br/><!--<br/>pbx_wait.h<br/>--> | Only happened if debug.xml leak check was used<br/><!--<br/>pbx_wait.h<br/>--> | ||
=== | === IP222 IP232 IP241: Adjust gains for better echo canceller performance === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="orange">To-decide</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83703 83703] | ||
|} | |} | ||
..<br/><!-- <br/>Use original EC mode by selecting mode 2.<br/>Max output gain reduced to avoid strong echo<br/>ADC input gain max<br/>Digital input gain set to 0db.<br/><br/>ac_phone3.cpp ac_codec.cpp --> | |||
=== | === myPBX: Remove unimportant notifications === | ||
{| | {| | ||
| Line 14,713: | Line 14,713: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83707 83707] | ||
|} | |} | ||
The following events are not so important that the user has to be notified using a windows bubble notification.<br/>- Visibility requests<br/>- Missed calls<br/><!--<br/>client.js<br/>--><br/> | |||
=== | === IP222 IP232 IP241: Enable noise reduction === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="orange">To-decide</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83715 83715] | ||
|} | |} | ||
Enable noise reduction on all handset/handsfree/DHSG headset micro.<br/>Use DSP code 660.27.pa.03<br/><br/>Noise reduction parameter can be tuned at<br/>http://addr/AC-DSP0/mod_cmd.xml?xsl=phone-dsp.xsl<br/><br/><!-- <br/>ac494e002ce4.660.27.pa.03.h ac494eac_494e.cpp ac_phone3.cpp ac_phone3.h phone-dsp.xsl <br/>--> | |||
=== | === Voicemail: Prevent <prompt>, <record> on incoming control calls === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83826 83826] | ||
|} | |} | ||
Turned out to cause idle-reset requests never being processed.<br/><br/>An administrative solution also exists: Watch out for the URL variable "$_noctl" in this article:<br/>http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables<br/><br/><!--<br/>pbx_vm.cpp<br/>--> | |||
=== | === HTTPCLIENT: Allow configuration of a http authenticated URL even if the server needs no authentication === | ||
{| | {| | ||
| Line 14,746: | Line 14,746: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83900 83900] | ||
|} | |} | ||
If a password was configured and the server didnt need it, a HTTP put created a file of zero length.<br/><br/><!-- httpclient_i.cpp httpclient_.h --><br/><br/> | |||
=== PBX: | === PBX Waiting: Trap on collision of operator connect and two-stage dialing === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83926 83926] | ||
|} | |} | ||
If two-stage dialing (Maps) is used to call a Trunk or Gateway object, the call is sent after a blockdial timeout. If an operator connected the call before this timeout happened, a trap occured.<br/><br/>For this to happen DTMF maps and operators have to be configured on the same Waiting Queue object, with is kind of unusual<br/><!--<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>--> | |||
=== PBX: | === PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received === | ||
{| | {| | ||
| Line 14,768: | Line 14,768: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83940 83940] | ||
|} | |} | ||
Calls from a Trunk must be explicitly marked as internal with respective connected number, otherwise they should be treated as external<br/><!--<br/>pbx.cpp<br/>pbx_mobility.cpp<br/>--> | |||
=== | === HTTP: Chunked transfer fails if the last 2 bytes of the chunk header are in the next tcp packet === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="orange">To-decide</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83986 83986] | ||
|} | |} | ||
Seen with the application platform as broken pipe<br/><br/><!-- http.cpp --> | |||
=== | === PBX: Registration with Name/Number did not work correctly if default device not first === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="orange">To-decide</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83989 83989] | ||
|} | |} | ||
For a registration with name or number, the information if the PBX password shall be used was always taken from the first device regardless if this was the default device (hw-id identical to name) or not<br/><!--<br/>pbx.cpp<br/>--> | |||
=== | === PBX: Partial Rerouting was prohibited in Alerting State (CFNR) === | ||
{| | {| | ||
| Line 14,801: | Line 14,801: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=83993 83993] | ||
|} | |} | ||
This was done under the assumption the partial rerouting is not supported as CFNR by public networks.<br/><!--<br/>pbx.cpp<br/>--> | |||
=== | === PBX Waiting: Call to operator with Twin Phone Checkmark did not work anymore === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84084 84084] | ||
|} | |} | ||
Collateral damage of<br/><br/>fix: #82858: PBX Waiting: Evaluate Busy on ... Calls for calls to an operator<br/><!--<br/>pbx_wait.cpp<br/>--> | |||
=== | === SoftwarePhone: Product string === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84119 84119] | ||
|} | |} | ||
The | The product string is changed, used e.g with the PBX registration.<br/><br/><!--<br/>Changed file:<br/>portkernel.cpp<br/>--> | ||
=== | === phone: provide complete dialog info to a phoneapp === | ||
{| | {| | ||
| Line 14,834: | Line 14,834: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84192 84192] | ||
|} | |} | ||
Both group indications and dialog infos are signaled via a group indication facility. For dialog infos the parked_to_alerting member was overloaded to provide the info as expected by the existing phoneapp.<br/>Now the parked_to_alerting member is passed to a phoneapp as received.<br/><br/><!--<br/>phonesig.cpp phonesig_if.h app_reg.cpp app_gui.cpp app_fkey.cpp app_disp.cpp<br/>--> | |||
=== | === IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added) === | ||
{| | {| | ||
| Line 14,845: | Line 14,845: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84200 84200] | ||
|} | |} | ||
In scenarios with frequent transistions of the attached PC to sleep renegotiating the link speed may be undesired. For this case the force same speed mechanism can be disabled.<br/><br/>Other changes:<br/>1000M is only changed to 100M if the other port runs at 100M. The previous version changed from 1000M to 100M if the other port runs at 100M or 10M.<br/><br/>The statistics can be collected from the PC port or from the LAN prot or from both.<br/><br/>Packet forwarding on the PC port is disabled if the port is down to avoid misleading collision counter behaviour.<br/><br/><!-- orchid_drv.cpp orchid_drv.h orchid_drv.xsl orchid_drv_s.xsl --><br/><br/> | |||
=== | === CDR fixes for external call detection === | ||
{| | {| | ||
| Line 14,856: | Line 14,856: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84211 84211] | ||
|} | |} | ||
The type="ext" attribute was not set reliably. Additionaly an attribute pseudo was added to the <user/> tag to indicate the type of object the CDRis created for.<br/><!--<br/>pbx_api.h<br/>pbx.cpp<br/>pbx_exec.cpp<br/>pbx_wait.cpp<br/>pbx_bc.cpp<br/>pbx_mobility.cpp<br/>--> | |||
=== | === http client : authentication was not retried after a failure when the offending request was repeated in the same session === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="orange">To-decide</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84217 84217] | ||
|} | |} | ||
When a httpclient user repeated a failing request in the same session the authentication was not tried again. Thus a change of the client side URL password or a change of the server side password had no effect until a new session was started.<br/><br/><!-- httpclient_i.cpp --> | |||
=== | === IP6000: Prevent blinking error LED on old IP6000 with HW-Build <110 === | ||
{| | {| | ||
| Line 14,878: | Line 14,878: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84227 84227] | ||
|} | |} | ||
the | Conference DSP driver was started on old hardware that doesnt support the conference DSP<br/><!-- adsp2191_drv.cpp --> | ||
=== phone | === IP222 IP232 IP241: pressing speaker key when phone is in handset mode switches to handsfree mode instead to toggle monitor mode === | ||
{| | {| | ||
| Line 14,889: | Line 14,889: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84297 84297] | ||
|} | |} | ||
handset/headset plus speaker is not supported<br/><br/><!-- codec.h ac_codec3.h ac_codec3.cpp phonesig_if.h phonesig.h phonesig.cpp app_ctl.cpp --> | |||
=== | === myPBX: Inconsistent display of group monitoring rights === | ||
{| | {| | ||
| Line 14,900: | Line 14,900: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84350 84350] | ||
|} | |} | ||
In the visibility settings groups were not displayed if the membership is "dynamic out". But members of that group still have monitoring rights.<br/><!--<br/>pbx_client.cpp<br/>--> | |||
=== | === phone: when scrolling directory search results sometimes one of the numbers of a contact was not displayed === | ||
{| | {| | ||
| Line 14,911: | Line 14,911: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84362 84362] | ||
|} | |} | ||
the tag characters assigned to the different numbers were not included in sort order. | |||
=== | === phone: status messages for outbound external calls were sometimes garbled on display === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="orange">To-decide</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84365 84365] | ||
|} | |} | ||
happened when for a preceeding outbound internal call in alerting state presence info was displayed<br/><br/><!-- forms.h forms_gen.cpp forms_color.cpp app_ctl.h app_call.cpp app_disp.cpp --> | |||
=== | === IP0010: DSP didnt start with build 9061044 === | ||
{| | {| | ||
| Line 14,933: | Line 14,933: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84388 84388] | ||
|} | |} | ||
<!--ac_491.cpp adsp2191:drv.cpp--> | |||
=== | === Page Ldap/Replicator/Status didn't display in WebKit === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84400 84400] | ||
|} | |} | ||
Page Ldap/Replicator/Status didn't display in WebKit<br/><!--<br/>ldaprep_status.xsl<br/>--> | |||
=== | === phone_orchid: displaying both dialed and connected number may be misleading, it's better to omit dialed number === | ||
{| | {| | ||
| Line 14,955: | Line 14,955: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84422 84422] | ||
|} | |} | ||
when for example 022222222 was dialed and the network reported a connected number 03022222222 the display info "022222222 -> 03022222222" looked like a transfer.<br/><br/><!-- forms_color.cpp --><br/> | |||
=== | === Hide LDAP Server Password For Viewer Accounts === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84557 84557] | ||
|} | |} | ||
Was accessible for viewers<br/><!--ldapsrv.cpp<br/>--> | |||
=== | === phone: enable directory search function key also in connected state === | ||
{| | {| | ||
| Line 14,977: | Line 14,977: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84559 84559] | ||
|} | |} | ||
it's sometimes useful to browse the directories for a number while talking.<br/><br/><!-- app_fkey.cpp --> | |||
=== | === myPBX: Only send one command at a time === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84568 84568] | ||
|} | |} | ||
Queue commands instead of sending overlapping commands. This limits the number of open AJAX connections to two.<br/><!--<br/>client.js<br/>--> | |||
=== IP232,IP222,IP241: Rendering errors when trying to use backround image with indexed colors === | === H.323: Accidential fallback to slowstart if faststart response received in PROGRESS === | ||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84601 84601] | |||
|} | |||
Only happened in some H.323/SIP interop scenarios<br/><!--<br/>h323sig.cpp<br/>--> | |||
=== IP222 IP232 IP241: Updated gain and equalizer setting === | |||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84605 84605] | |||
|} | |||
Tuned for high MOS values according to ETSI ES202737 ES202738 ES202739 ES202740<br/><!-- ac_codec3.* platform/filter/* --> | |||
=== phone: ip222, ip232: status stage added to all USB control transfers === | |||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84617 84617] | |||
|} | |||
the status stage is mandatory as well for IN as for OUT control tranfers<br/><br/><!-- ohci.cpp ohci.h ohci_defs.h --> | |||
=== H.323: More information on "Unexpected Message" event === | |||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84699 84699] | |||
|} | |||
The message type and the state for which this message was unexpected is needed to find out what the problem is.<br/><!--<br/>h323sig.cpp<br/>--> | |||
=== SIP: Trap during channel handling === | |||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84800 84800] | |||
|} | |||
Rare trap when re-assigning channels.<br/><br/><!--<br/>sip.cpp/h<br/>medialib.h<br/>--> | |||
=== IP232,IP222,IP241: Rendering errors when trying to use backround image with indexed colors === | |||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84814 84814] | |||
|} | |||
PNG mode "indexed colors" is not supported.<br/>Trying to use an indexed color png leaves phone screen in bad shape.<br/><br/><!--<br/>read_png.cpp<br/>--> | |||
=== NAT: Don't forward DNS requests from public network === | |||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84842 84842] | |||
|} | |||
As kind of denial of service attack, bursts of incoming DNS requests were seen. The nat process was forwarding these requests to the public DNS. This is a useful function for DNS requests from the private network, but not for requests from the public network.<br/><br/>These DNS requests are now discarded<br/><!--<br/>socket.h<br/>udp4.cpp<br/>udp.cpp<br/>nat.cpp<br/>--> | |||
=== H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC === | |||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84848 84848] | |||
|} | |||
Caused SIP interop problems with CUCM<br/><!--<br/>h323sig.cpp<br/>--> | |||
=== Maximum LDAP PDU Size Too Small === | |||
{| | |||
|Status | |||
|<font><font color="green">Closed</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84851 84851] | |||
|} | |||
Maximum LDAP PDU Size Too Small. Now internal maximum allocation unit plus a bit for encoding overhead.<br/><br/><!--<br/>ldapasn1.cpp<br/>--> | |||
=== PBX Broadcast: Call to group members was not cleared when CFNR was executed === | |||
{| | |||
|Status | |||
|<font><font color="orange">To-decide</font></font> | |||
|- | |||
|Id | |||
|[http://mantis.innovaphone.com/view.php?id=84857 84857] | |||
|} | |||
This was a collateral damage from<br/>fix: #79549: PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object<br/><!--<br/>pbx.cpp<br/>--> | |||
=== IPVA: V10 Code Merge Aiming To Stabilize FW Upload To DRAM === | |||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84862 84862] | ||
|} | |} | ||
Addresses a problem in conjunction with innovaphone's automated test environment<br/><!--<br/>ipva.cpp<br/>vlance_drv.cpp<br/>ide_drv.cpp<br/>--> | |||
=== | === PBX: Filters should only be applied to calls with media channels === | ||
{| | {| | ||
| Line 15,010: | Line 15,120: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84892 84892] | ||
|} | |} | ||
A join group operation could be prohibited with filters. This was unexpected.<br/><!--<br/>pbx.cpp<br/>--> | |||
=== | === phone: ip222, ip232: raise alarm if an USB Headset does not respond on USB bus anymore === | ||
{| | {| | ||
| Line 15,021: | Line 15,131: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84893 84893] | ||
|} | |} | ||
<!-- keys_ip222.cpp, ohci.cpp, ohci.h --><br/> | |||
=== | === SIP: Fix for media negotiation on calls re-routed from TONE interface to outbound SIP === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="orange">To-decide</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84932 84932] | ||
|} | |} | ||
Helps on gateway interfaces configured for media-relay with exclusive codec.<br/><br/><!--<br/>sip.cpp<br/>--> | |||
=== | === IP-DECT: Call transfer timer === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="green">Closed</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84951 84951] | ||
|} | |} | ||
This | The call transfer timer is stopped with the call proceeding event now. This fixes a call transfer to e.g. a mobile user with a delayed alert.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--> | ||
=== | === IP-DECT: Reverse phone book === | ||
{| | {| | ||
| Line 15,054: | Line 15,164: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84953 84953] | ||
|} | |} | ||
The reverse phone book does not work till hotfix 11. This is fixed again now.<br/><br/><!-- Changed files:<br/>dectradio.cpp<br/>dectradio_mastercall.cpp<br/>--> | |||
=== | === SIP: Loosing remote IP addresses when DNS becomes temorarily unavailable === | ||
{| | {| | ||
| Line 15,065: | Line 15,175: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=84991 84991] | ||
|} | |} | ||
Loosing remote IP addresses when DNS becomes temorarily unavailable.<br/>Results in interfaces without remote ip addresses.<br/><br/><!--<br/>sip.cpp/h<br/>--><br/> | |||
=== | === PBX Routing: A CFNR on a slave gateway object redirecting the call to the master failed === | ||
{| | {| | ||
| Line 15,076: | Line 15,186: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=85017 85017] | ||
|} | |} | ||
<!-- | This was a collateral damage of<br/><br/>fix: #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX<br/><!--<br/>pbx.cpp<br/>--> | ||
=== SIP: | === SIP: Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI) === | ||
{| | {| | ||
| Line 15,087: | Line 15,197: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=85047 85047] | ||
|} | |} | ||
Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI).<br/><br/><!--<br/>sip.cpp<br/>--> | |||
=== | === IP222 IP232 IP241: Headet volume adjust added, re-enable noise reduction in headset mode === | ||
{| | {| | ||
|Status | |Status | ||
|<font><font color=" | |<font><font color="orange">To-decide</font></font> | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=85058 85058] | ||
|} | |} | ||
Headset volume adjust can be done with the webinterface at<br/> http://addr/dsp.xml<br/><br/>This feature can be used to adapt to different DHSG headsets.<br/><br/>The noise reduction in headset mode is re-enabled to avoid sporadic noise with <br/><br/><!-- ac_phone3.* ac_codec3.* ac_dsp3.mak --> | |||
=== | === SIP: Memory leak === | ||
{| | {| | ||
| Line 15,109: | Line 15,219: | ||
|- | |- | ||
|Id | |Id | ||
|[http://mantis.innovaphone.com/view.php?id= | |[http://mantis.innovaphone.com/view.php?id=85083 85083] | ||
|} | |} | ||
Memory leak.<br/><br/><!--<br/>sip.cpp<br/>--> | |||
Revision as of 23:31, 8 August 2012
This is the Firmware V9 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!
V9 Release
This is the Version 9 Release. It is expected to be released 2011, Week 17. Definition
New Features
myPBX
| Status | Closed |
| Id | 26947 |
Web client for the end user allowing access to
* Call Control
* Contacts (Partners)
* Call Forwarding
* Directoty (LDAP)
IPv6
| Status | Closed |
| Id | 36001 |
IPv6 as additional network protocol
- will coexist with IPv4
- many network components and appications need to be adapted (DHCP, NAT, VOIP, ...)
Reporting
| Status | Closed |
| Id | 50332 |
Application to collect CDRs in a database, which can be used for call-lists
Use of SIP Dialog Info for Group Functions across PBXs
| Status | Closed |
| Id | 52926 |
Allow an endpoint to access dialog info from endpoints registered to other PBXs. This can be used for Partner Keys and similar functionality.
With the innovaphone PBXs the SIP Dialog Info is mapped to H.450 Group Indications
Other new Features
| 57648 | dde | My.innovaphone: interface for downgrade/upgrade licenses |
| 14925 | gd | LDAP over TLS |
| 37992 | gd | Boolean to check for day/month/year also |
| 59130 | msc | HTTP-Client: Support for proxy server |
| 61702 | msc | X.509: Certificates with domain names and IP addresses |
| 61790 | tac | G.722 negotiation in SIP signaling |
| 61837 | gd | SIP: Send Alert-Info in INVITE |
| 8495 | cmu | ip21/22/24/302 TFE |
| 52956 | queue | PBX: Password for unknown regs |
| 45803 | jfr | DHCP POSIX timezone option id is 100 officially (since 2007), let server provide it if explicitely requested |
| 59131 | msc | HTTP-Client: DNS resolution |
| 59132 | msc | HTTP-Client: Support for HTTP redirect |
| 54937 | tac | Support for multiple presences on phones |
| 60088 | tac | SIP: Support for Event "talk" |
| 60092 | tac | SIP: Support for Event "conference" |
| 60537 | tac | SIP: Put "alias" parameter into Via header of REGISTER request (RFC-5923) |
| 62741 | tac | SIP: Get display information from Call-Info header in connect response |
| 63186 | tac | SIP: Get display information from Call-Info header in register response |
Bug Fixes
V9 Hotfix 1 (90600.01)
Changes included in Version 9 hotfix1 Definition
New Features
SoftwarePhone: default configuration
| Status | Closed |
| Id | 63632 |
The default configuration is changed to coder G.711a and license type PBX.
SoftwarePhone: license information
| Status | Closed |
| Id | 63633 |
If no valid license is received from the PBX, a information is printed directly now.
| Status | Closed |
| Id | 63679 |
The DECT menu is visible in the IP6000/IP6010/IP0010/IP1060/IP3010 devices, if the required DECT multi-cell license is available.
SoftwarePhone: de-registration at application end
| Status | Closed |
| Id | 63781 |
The application sends a de-registration to the PBX, if the user closes it.
DECT: OEM device user configuration
| Status | Closed |
| Id | 63883 |
For an OEM device the user data handling must be different. This does not concern the IP1200.
myPBX: Redial
| Status | Closed |
| Id | 63930 |
Add a button to the web application that writes the last dialled number into the number input field.
Files:
client.js
client.htm
action_redial.png
pbx.mak
PBX Root Node Extern on Slave for calls from Master
| Status | Closed |
| Id | 63995 |
If a Root Node Extern interface was configured on a slave this was currently only used in case this was a "License Only" slave.
Now this interface is also used for calls from the master with no destination found. This can only happen in non-replicated environments.
Status:
pbx.cpp
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)
SoftwarePhone: Tray Icon
| Status | Closed |
| Id | 64198 |
Now the SoftwarePhone runs as Windows application with an tray icon.
PBX: Send 'Connected Name' in Connect
| Status | Closed |
| Id | 64199 |
This is for example needed to display E-Mail Icon in myPBX for contacts in remote PBX
Status:
signal.cpp
signal.h
h323sig.cpp
h323sig.h
pbx.cpp
DECT: forward info message to radio
| Status | Closed |
| Id | 64245 |
The info message is forwarded from the gatekeeper to the radio in an early call state now. The message is used for display updates by a OEM PBX between call setup and alerting response from DECT.
DECT: User import file convertion
| Status | Closed |
| Id | 64297 |
Support for latin1 to UTF-8 conversion for user import added. The import feature is only used in an OEM device.
DECT: Internal endpoint information 'assigned-by'
| Status | Closed |
| Id | 64316 |
The DECT system saves the information with which method an endpoint was assigned to an user - by the administrator or with the feature code 'user login'. It is used by OEM modules.
DECT: Location update option for UTF-8 support
| Status | Closed |
| Id | 64371 |
A new option is added for the location update events to distinguish between latin1 and UTF-8 handsets. This feature is only used by an OEM device and OEM handsets for display updates to support UTF-8.
Truncating UTF-8 aware string copy function
| Status | Closed |
| Id | 64380 |
New string library function to correctly truncate an UTF-8 string to fit in a buffer.
SoftwarePhone: Setup about logo
| Status | Closed |
| Id | 64414 |
The setup about innovaphone logo was changed.
SIP: Handling on INFO inside instant message session
| Status | Closed |
| Id | 64535 |
INFO sip:xxx@172.16.0.200:5060;transport=TCP SIP/2.0
Via: SIP/2.0/tcp 172.16.1.61:5060;branch=z9hG4bK.DF25F93ADB524A283550ECBD871BC73B
From: <***>;tag=902D6B1B5622453B000172A9724270D4;epid=00000003
To: <***>;tag=1440372429
Call-ID: 5D4FA4260D42445F4071AEA032D4321B
CSeq: 3 INFO
Contact: ***
Content-Type: application/xml
Max-Forwards: 70
Supported: timer
Content-Length: 82
<?xml version="1.0"?>
<KeyboardActivity><status status="type" /></KeyboardActivity>
SIP: Locating remote federation server with another SRV query
| Status | Closed |
| Id | 64549 |
There are different service name decorations in use
_sips._tcp.<domain>
and also
_sipfederationtls._tcp.<domain>
SIP: Treat "Privacy:off" like "Privacy:none"
| Status | Closed |
| Id | 64692 |
Treat "Privacy:off" like "Privacy:none" when receiving INVITE
DECT: Increase number of anonymous RAS registrations
| Status | Closed |
| Id | 64761 |
Now the number of the allowed anonymous RAS registrations is increased to 30.
IP241: Multi line text ctrl must support line break
| Status | Closed |
| Id | 64938 |
Multi line text ctrl must support line break when displaying received text messages or when composing new text messages.
DECT: Option not to use TLS
| Status | Closed |
| Id | 65053 |
for easier debugging
Status:
dectradio.cpp
dectradio.h
SIP: New config file option /no-alert-info
| Status | Closed |
| Id | 65084 |
Supress Alert-Info header for interoperability with Thomson ST2030 and Thomson TB30
IP241: Show active CFU destination on idle screen
| Status | Closed |
| Id | 65176 |
Show active CFU destination in header bar of idle screen.
Not only a symbol on status bar.
H.323: Display call state in "Signaling Timeout" error log
| Status | Closed |
| Id | 65194 |
To provide better indication about the nature of the problem Status: h323sig.cpp
SIP: add an "alias" header field parameter to the Via header field
| Status | Closed |
| Id | 65235 |
According to RFC-5923 to allow TLS connection re-use Status: siptrans.cpp
SIP: New config file option /add-cn-capability
| Status | Closed |
| Id | 65313 |
Required for mediation server (lync) interoperability.
Otherwise mediation server complains:
"The Gateway peer does not support comfort noise"
new License Mechanism <type>@<num>=n
| Status | Closed |
| Id | 65339 |
For hosting provider
Status:
inno_lic.cpp
inno_lic.h
PBX: BC Conference instant messaging support
| Status | Closed |
| Id | 65422 |
Instant messaging support with myPBX for the "Bc Conference" PBX object added.
Gateway: Allow sending of Date/Time in Connect on ISDN interfaces
| Status | Closed |
| Id | 65445 |
Was missing in the User Interface, so it could not be configured
Status:
relay_edit_phys.xsl
config.h of ip800, ip24, ip3000, ip6000, ip6010
Web-UI: Restyled Config Wizard
| Status | Closed |
| Id | 65489 |
Config wizard got new look
PBX CDRs: Transfered call generated 2 cdrs. This was confusing
| Status | Closed |
| Id | 65521 |
For an incoming call which was transfered, one CDR for the incoming call and one for the outgoing transfer was generated. Now one CDR for the incoming call including the information for the outgoing transfer is generated
Status:
pbx.cpp
pbx_api.h
pbx_wait.cpp
IP232: Implemented touch keyboard
| Status | Closed |
| Id | 65680 |
Implemented touch keyboard
IP222: Alpha input using the num block
| Status | Closed |
| Id | 65681 |
Alpha input using the num block
DECT: Option short busy tone
| Status | Closed |
| Id | 65693 |
New option 'Short busy tone' for the DECT system added. If this is enabled, the busy tone after remote hangs up is shortened.
DECT: License check for OEM
| Status | Closed |
| Id | 65727 |
The DECT Master checks the DECT Radio license for an OEM device and PBX.
myPBX: Show allows that are configured using templates
| Status | Closed |
| Id | 65746 |
So that the user can see all access rights
files:
pbx_client.cpp, client_allows.js
SoftwarePhone: Jabra BIZ 2400 support
| Status | Closed |
| Id | 65755 |
Now the Jabra BIZ 2400 is supported by the SoftwarePhone.
SIP: New config file option /no-app-rerouting
| Status | Closed |
| Id | 65801 |
Interoperability with Ingate Session Border Controller.
Required when handling redirect responses.
SoftwarePhone: Jabra SPEAK 410 USB support
| Status | Closed |
| Id | 65832 |
Now the Jabra SPEAK 410 USB is supported by the SoftwarePhone.
SoftwarePhone: Jabra LINK 320 USB support
| Status | Closed |
| Id | 65833 |
Now the Jabra LINK 320 USB is supported by the SoftwarePhone.
DECT: Device state info for OEM GUI
| Status | Closed |
| Id | 65864 |
Device state info for OEM GUI added.
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.
IP241: Do not hide all fkeys in active state
| Status | Closed |
| Id | 65923 |
Keep these fkeys on display that do not interfere with call controls.
SoftwarePhone: Jabra PRO 94xx support
| Status | Closed |
| Id | 65933 |
Now the Jabra PRO 9400 serie is supported by the SoftwarePhone.
SIP: Handling of NOTIFY(message-summary) on gateway interfaces without registration
| Status | Closed |
| Id | 65946 |
Required for LYNC interaction
SIP: Improved CCM interoperability
| Status | Closed |
| Id | 66002 |
Required for IP-DECT
PBX Admin UI: When editing allowed access, show configuration inherited from templates
| Status | Closed |
| Id | 66059 |
Result was sometimes confusing, because it was very hard to find templates from which the current object inherited config
Status:
pbx.cpp
pbx_api.h
pbx_edit_allow.xsl
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
Gateway: New interface option "No Early Media" for SIP interoperability
| Status | Closed |
| Id | 66092 |
New config option "No Early Media" in section "SIP Interop Tweaks".
Required for LYNC interoperability to ignore SDP answer in 183 Session Progress,
since no inband progress information is provided by LYNC.
PBX Mobility: Min-Alert/Max-Alert timer on configured mobility
| Status | Closed |
| Id | 66152 |
This way it can be prohibited that an announcement from the mobile provider accepts a mobility call
Status:
pbx.cpp
pbx.h
pbx_api.h
pbx_mobility.cpp
pbx_mobility.h
pbx_admin.cpp
pbx_edit_fork.xsl
work.css
PBX: Name not case sensitive anymore
| Status | Closed |
| Id | 66172 |
An object Name in the PBX can now be used as user part of a SIP URI and this is not case sensitive. This could be a compatibility issue if users are configured with names which are the same except the case. Status: pbx.cpp
SIP: Decode application/rlmi+xml for presence from external UC server
| Status | Closed |
| Id | 66175 |
Decode application/rlmi+xml for presence from external UC server
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
DECT: Configuration option for OEM device
| Status | Closed |
| Id | 66425 |
New configuration option 'Output Power' in DECT Radio for an OEM device.
Bug Fixes
SIP: SRTP re-keying when retrieving held call
| Status | Closed |
| Id | 62620 |
Required for Cisco interoperability
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
SIP: Media negotiation problem during transfer to early media source
| Status | Closed |
| Id | 63422 |
test\\9.00\\relay\\early-media failed
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.
SIP: Not accepting calls from alternative proxy
| Status | Closed |
| Id | 63600 |
When being registered at a proxy with 2 ip addresses the gateway does not accept calls from the alternative ip address.
SoftwarePhone: reloading current configuration
| Status | Closed |
| Id | 63607 |
The device configuration is correctly loaded in setup.
SoftwarePhone: ringing through device
| Status | Closed |
| Id | 63608 |
Ringing through device fixed, if there are more output devices than input devices.
Web-UI: PBX password length is limited to 15 chars
| Status | Closed |
| Id | 63640 |
Added tooltip and fixed maxlength attribute on input elements.
SoftwarePhone: trap with busy external call
| Status | Closed |
| Id | 63642 |
If a call to external is busy and is hooked off with myPBX, a trap occurs. This is fixed now.
License: Character encoding problem
| Status | Closed |
| Id | 63645 |
Character encoding problem
IP241 DSP may freeze under heavy load
| Status | Closed |
| Id | 63648 |
A function key test script which creates/releases a big number of calls (single calls, conference calls, intrusion calls) in a very short time resulted in a frozen DSP.
myPBX launcher: Remove standard URL
| Status | Closed |
| Id | 63652 |
Currently the launcher uses a standard URL (http://172.16.0.10/PBX0/MY/client.htm). This is obsolete and should be removed.
files: ctiwin.cs
calling tone missing in some cases
| Status | Closed |
| Id | 63667 |
the calling tone was switched off when RTP data was received from remote party after the alerting indication even if the data did not contain in-band info.
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.
Trap with conference (ADSP driver)
| Status | Closed |
| Id | 63706 |
A trap in ADSP hardware driver occurs if a new member joins an existing conference in exclusive listen mode. All devices with ADSP hardware: IP6000/IP6010/IP0010/IP1060/IP3010.
myPBX launcher: Hotkey should always bring myPBX to front
| Status | Closed |
| Id | 63718 |
Not only if a number is marked
Files:
ctiwin.cs
client.js
myPBX: Script error on line 1351
| Status | Closed |
| Id | 63744 |
A script error is thrown when there is an unexpected missed-call-auth:
client.js, Line 1351, this.manager is null
files: client.js
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.
myPBX: Improvements of Finnish translation
| Status | Closed |
| Id | 63747 |
files: pbx_client_localisation.cpp
Soita => Puhelu
Keskustele => Keskustelu
phone ip241 - monitor mode (handset/headset + speaker) cannot be disabled once enabled
| Status | Closed |
| Id | 63750 |
if monitor mode is set by pressing the speaker key while in handset or headset mode the handsfree speaker gets the same data as the handset or headset speaker. pressing the speaker key again must mute the handsfree speaker.
Trap when starting from flash_stick
| Status | Closed |
| Id | 63752 |
and flash memory not yet programmed with bootcode Status: ip6010.cpp
myPBX launcher: Map all variants of the Norwegian language to "no"
| Status | Closed |
| Id | 63771 |
There are three ISO 639-1 language codes for the variants of the Norwegian languages.
Norsk (Bokmål) (nb)
Norsk (Nynorsk) (nn)
Norsk (no)
myPBX only provides a translation for "no". To avoid that users with the "nb" or "nn" locale get the English translation, all language codes have to be mapped to "no" by the myPBX launcher.
Files: ctiwin.cs, start.js
no dialtone and no audible connection on ip24, ip22 and ip28 with V9
| Status | Closed |
| Id | 63780 |
even in both directions Status: ac_dsp3.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
...
PBX: CDRs from Waiting Queue in case of call thru DTMF map wrong
| Status | Closed |
| Id | 63882 |
The CDR just ended without the transfer and following events Status: pbx_wait.cpp
GW Route Configuration: XML Content Not Well Formed When Inserting A Map
| Status | Closed |
| Id | 63884 |
Could happen with certain types of routing maps
myPBX: Handling of calls with no name and no number
| Status | Closed |
| Id | 63885 |
myPBX updated calls only if a number or a name was set in the call-update or contact-call-update message.
Files: client.js
SoftwarePhone: conference no speech
| Status | Closed |
| Id | 63902 |
Sometimes the SoftwarePhone 3pty conference is not working. It is fixed now.
Preserve number of calls on SIG_CONN without connected number
| Status | Closed |
| Id | 63906 |
On calls across PBXes the number is discarded when the SIG_CONN does not contain a connected number.
files: pbx.cpp (pbx_connector::receive_connected_number)
Calling Tone garbled when using G722 with a framesize > 40 milliseconds
| Status | Closed |
| Id | 64015 |
The maximum G722 framesize is 40 milliseconds (320 byte packets). A configured size > 40 was not adjusted correctly and triggered the calling tones too fast.
One-way audio when using G711u
| Status | Closed |
| Id | 64036 |
Playback of G711u RTP audio may fail and error log "Wrong Payload Type received" is set in some cases. Status: channel.cpp
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
ip6010 family : CFLASH overcrowds trace buffer if trace flag is turned on
| Status | Closed |
| Id | 64061 |
dto Status: mv78x00_cfdrv.cpp
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
PBX: Group not configurable for PBX object
| Status | Closed |
| Id | 64075 |
needed by SOAP applicatons (e.g. TAPI) to find slave PBXs Status: pbx.h
PBX: Rej.External Calls not possible for all object types
| Status | Closed |
| Id | 64111 |
when creating the objects it was sometimes possible to set this checkmark but later when editing the object the checkmark was not displayed anymore Status: pbx_edit_object.xsl
propagating comma separated option values to multiple device dependent config lines did not work
| Status | Closed |
| Id | 64114 |
Some devices (for example ip110, ip200a, ip150, ip230, ip240) use an identical firmware which is configured at boot time according to the device it's running on. This configuration is kept on device specific config command lines. A "config change" was not handled correctly.
H.323 limit registration redirects to avoid high load on misconfiguration
| Status | Closed |
| Id | 64116 |
In a non-replicated PBX environment it could happen that a loop of registration redirections is configured. In this case the endpoints were redirected as fast as the resources allowed between the different destinations. This could produce high load on the PBXs.
Status:
h323.h
h323sig.cpp
myPBX: Save button for contacts stays disabled when using autocomplete
| Status | Closed |
| Id | 64117 |
This can happen using Firefox. Turn autocomplete off and use the oninput event additionally.
files: client_contact.htm
Media negotiation problem H.323 to SIP call, with reverse Media and media relay
| Status | Closed |
| Id | 64124 |
a call without Media in both directions could be the result
Status:
h323ch.cpp
sip.cpp
SIP: L16 codec handling was wrong
| Status | Closed |
| Id | 64126 |
Dynamic payload type for L16/16000 and L16/8000 got lost.
SDP-Offer:
m=audio 5000 RTP/AVP 107 105 0 8 9 101
a=rtcp:5001 IN IP4 192.168.10.247
a=rtpmap:107 L16/16000
a=rtpmap:105 L16/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Decoded into:
1:L16/16000(10) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)
1:L16/8000(10) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)
1:G711u/8000(0) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)
1:G711A/8000(8) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)
1:G722/16000(9) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)
1:DTMF/8000(101) . xmit(0) recv(0) 192.168.10.247:5004 (0.0.0.0:0)
HTTP server: Query string is lost on redirect for "Force HTTPS"
| Status | Closed |
| Id | 64128 |
The location header of the redirect does not include the initial query string.
files: http.cpp, http.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.
SIP: Memory leak when receiving small datagrams on SIP port
| Status | Closed |
| Id | 64134 |
Some SIP clients send small UDP datagrams in order to refresh mapping on NAT devices.
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
phone orchid: called party microphone muted after a G722 call has been connected
| Status | Closed |
| Id | 64139 |
This is an effect of the sampling rate change when switching the channel from G711 (ringing) to G722. The microphone was unmuted when the speaker volume was changed thereafter. Now the volume settings are restored automatically after the sampling rate has been changed.
myPBX launcher: Hotkey does not work for same number two times in a row
| Status | Closed |
| Id | 64169 |
files: TextPeeker.cs
SDP: Wrong clock rate for G.722 codec
| Status | Closed |
| Id | 64182 |
RFC-3551
Even though the actual sampling rate for G.722 audio is 16,000 Hz,
the RTP clock rate for the G722 payload format is 8,000 Hz because
that value was erroneously assigned in RFC 1890 and must remain
unchanged for backward compatibility. The octet rate or sample-pair
rate is 8,000 Hz.
myPBX: Use different icon for logout
| Status | Closed |
| Id | 64184 |
So that users can better see the difference between logging out and closing the window.
files: window_action_logout.png
Phone: trap upon deletion of user reg on pbx if registration information being viewed on the phone
| Status | Closed |
| Id | 64186 |
problem: trap upon deletion of user reg on pbx if registration information being viewed on the phone
solution: fixed in forms code (unbind screen)
files: box/forms/lcd/forms_impl.cpp
products: phones
risks: none
PPPOE: specific configuration not reachable from config web page
| Status | Closed |
| Id | 64192 |
problem: PPPOE: specific configuration not reachable from config web page, so no new PPPOE can be configured (already present ones run though), also ISDN part always visible
solution: fixed
files: ip_pppif.xsl (now check for PPPOE0, PPPOE1 and PPPOE2 types)
products: all (gateways effected)
risks: none
myPBX: Show login screen after logout instead of closing window
| Status | Closed |
| Id | 64194 |
files: client.js
DHCP configuration with secondary but no primary gatekeeper not possible
| Status | Closed |
| Id | 64232 |
This kind of configuration is required in some redundancy scenarios. It could be configured manually on the phones but not distributed via the innovaphone DHCP server. Keep in mind that only new telephone firmware will understand a DHCP configuration providing the secondary gatekeeper only.
Out Of Memory Trap when running VM without prompt files
| Status | Closed |
| Id | 64243 |
When calling into a Voice Mail object without prompt files
memory objects are allocated at high rate without being freed.
Memory is freed at disconnect.
This may cause a OOM trap when call stays connected for a longer time.
IP241: Number information interfere with status information on call ctrl
| Status | Closed |
| Id | 64250 |
Truncate number information to match screen metrics.
Show status information if space available only.
IP241 - VLAN frames > 1514 bytes were not accepted
| Status | Closed |
| Id | 64251 |
The default maximum receive length of 1518 does not permit for a 1516 byte frame.
With the maximum receive length set to 1536 such frames are received.
DECT: File upload for user import
| Status | Closed |
| Id | 64264 |
File upload fixed. Used for user import by an OEM device.
myPBX launcher: Open configuration dialog if no URL is configured
| Status | Closed |
| Id | 64265 |
On startup the config dialog should show up if no URL is configured, yet.
files: ctiwin.cs
PBX: Trap on security denial of 'unknown' registration
| Status | Closed |
| Id | 64269 |
If there is a 'unknown' registration attempt with wrong password (Unknown Registrations - With Admin Pwd only enabled) another attempt is rejected during security hold off period. If during this period another registration attempt with correct password happens, the PBX traps. Status: pbx.cpp
X.509: Use random serial numbers for self-signed certificates
| Status | Closed |
| Id | 64279 |
Currently the serial numbers start at 0 after long reset and are incremented by 1 for each new certificate. This causes collisions with old certificates that were created before the long reset.
Use a random number in order to avoid collisions after long reset.
Files: x509.cpp, x509.h
SIP: P-Asserted-Identity in UPDATE not working
| Status | Closed |
| Id | 64289 |
PAI with changed remote party identification was not handled
if also Session-Expires header was present in UPDATE request.
DNS/Hosts: SRV-Record, Port Default 5060 Confusing. Removed.
| Status | Closed |
| Id | 64312 |
Edit field is now going be empty
IP241: IP230X not working
| Status | Closed |
| Id | 64355 |
wrong UART used Status: ip230x.cpp phone_orchid.cpp config.h
supress "Send Number" for calls triggered by a 'Dial' function key with 'Send as Control Call' checked
| Status | Closed |
| Id | 64365 |
When using a 'Dial' function key with 'Send as Control Call' checked to control a call recording device the unique original calling party number must be passed to the recorder. The 'Send Number' configured in the the PBX user object may be the same for a group of phones and does not identify a certain phone.
Trunk Park/Pickup (line keys) did not work anymore
| Status | Closed |
| Id | 64373 |
Collateral damage from fix
fix: #61590: PBX: Boolean Function Key was not updated when joining group
Status:
pbx_gi.cpp
pbx_gi.h
pbx.cpp (v9 only)
pbx.h (v9 only)
pbx_api.h (v9 only)
IP200A IP230 IP240: manufacturing test not working
| Status | Closed |
| Id | 64378 |
write fixed bootheader fails Status: flash_firmware.cpp
IP241!: IP230X not working
| Status | Closed |
| Id | 64376 |
wrong UART used Status: ip230x.cpp phone_orchid.cpp config.h
VM, <prep-callinfo> doesn't apply with <pbx-xfer>
| Status | Closed |
| Id | 64384 |
VM, <prep-callinfo> doesn't apply with <pbx-xfer>
memory leak check missing for last parked call info
| Status | Closed |
| Id | 64445 |
when a call is parked using the 'Park' function key info about the parked call is kept for later checks when the call is unparked again. the leak check for this info was missing.
Phone trap when leaving User-Choose screen with Menu key
| Status | Closed |
| Id | 64448 |
problem: Phone trap when leaving User-Choose screen with Menu key. Missing "SaveChages?" screen.
solution: fixed in code. TXT_EMPTY==SvgChg Id. Worst Case occured.
files: phone_user_config.cpp
products: All Telephones
risks: none
PBX CDRs: Dummy call to endpoint because of SOAP UserCall generated CDR
| Status | Closed |
| Id | 64449 |
The PBX-SOAP UserCall function initiates a call to the endpoint and transfers this call to the destination given by UserCall. This initial call to the endpoint should not generate a CDR Status: pbx_xml.cpp
Phone: trap when redefining function key (long press access) from destination to partner
| Status | Closed |
| Id | 64498 |
problem: Phone traps when redefining function key (long press access) from e.g. destination number to partner
solution: fix in code
files: phone_user_func.cpp
products: All Telephones
risks: None
SIP: Trap when mypbx changes presence and external uc is configured
| Status | Closed |
| Id | 64504 |
Trap when mypbx changes presence and external uc is configured.
PBX: Failover to WAN for calls to Slave PBX did not work
| Status | Closed |
| Id | 64540 |
accidentally broken in v9
Status:
pbx.cpp
pbx.h
IP241: Directory entries did not show number tag
| Status | Closed |
| Id | 64555 |
Number tag indicates type of number (M = mobile number)
IPVA, Include NAT
| Status | Closed |
| Id | 64557 |
Include the NAT0 module as in other innovaphone products
PBX: Reporting licensing for standby operation did not work
| Status | Closed |
| Id | 64560 |
Reporting did not work for endpoints registered in standby mode, because of license check failed Status: pbx.cpp
Timeout when calling Mobile endpoint which does not send alert
| Status | Closed |
| Id | 64563 |
Some SIP carriers do not send correct alert but only something which can be translated to CALL-PROC. In this case the CALL-PROC was not forwarded to the caller and therefore the call timed out after 12s Status: pbx_mobility.cpp
User Interface: OK, Cancel, Apply buttons inconsistent
| Status | Closed |
| Id | 64567 |
The sequence is sometimes different. Should always be OK, Cancel, Apply
Status:
pbx_edit_allow.xsl
pbx_edit_cfs.xsl
pbx_edit_fork.xsl
pbx_edit_groups.xsl
pbx_edit_object.xsl
pbx_edit_presence.xsl
pbx_edit_profiles.xsl
pbx_edit_rights.xsl
HTTP-Client: Trap in minifirmware on requests using a domain name
| Status | Closed |
| Id | 64568 |
The HTTP client does not check if the DNS module exists before using it. In the minifirmware there is no DNS module.
Files: httpclient_i.cpp
Test: 9.00/unit-tests/httpclient/basic/150-get-dns-missing
A PBX user with "Full PBX Administration" Rights could not edit phone configuration
| Status | Closed |
| Id | 64572 |
The configuration pages could be opened once but after changing an item the input was disabled
PBX: IM sessions to external UC did not work
| Status | Closed |
| Id | 64583 |
The session was not initiated towards the external UC
Status:
pbx.cpp
pbx_uc.cpp
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
PBX: Publish to external UC, From header wrong
| Status | Closed |
| Id | 64591 |
A From header of anonymous was sent Status: pbx.cpp
myPBX: Detect standby cases in a different way
| Status | Closed |
| Id | 64611 |
Needed because the behaviour of api_other_location has changed (#64540)
files: pbx_client.cpp
A Bootcode Update could disrupt the Media stream for some seconds
| Status | Closed |
| Id | 64631 |
This was observed on phone devices with relatively slow flash memory when a bootcode update took place while a call was active.
Bootcode: Support for RGB LED added
| Status | Closed |
| Id | 64636 |
... Status: box.h boot.c boot.h flash.c flash_firmware.c soc_init.c platform_comcerto.c boot_comcerto.mak
phone: picking up a call failed sometimes
| Status | Closed |
| Id | 64679 |
Sometimes pressing the partner, pickup or park key to pick up an alerting or parked call had no effect.
phone: prevent the pc port of the ethernet switch from receiving frames directly from the phone firmware
| Status | Closed |
| Id | 64689 |
In some cases is not desired that frames sent by the phone firmware via the cpu port are recieved by the pc port. This may be prevented now by
config add ETH0 /isolate-pc
PBX User Interface did not work with Groups containing XML reserved characters (&,<,>,...) or non-ascii
| Status | Closed |
| Id | 64695 |
XML or URI encoding was missing in some paces. The browser could not display the page.
This happend when using the left PBX/Group tree for nvigation
Status:
pbx_admin.cpp
pbx_objs_left.xsl
pbx_objs_right.xsl
IP222/IP232 Bootcode: Support for hardware 300
| Status | Closed |
| Id | 64701 |
USB-Clock is generated from PRE-USB Clock, needs a 25 pre-divider Status: startup_orchid.s
TLS: Error on processing huge handshake messages
| Status | Closed |
| Id | 64702 |
The current implementation does not work with handshake messages that are bigger than 8 kilobytes. Especially the CertificateRequest message that is used for MTLS can be bigger.
files: tls.cpp
PBX v5 SoftwarePhones licenes did not work on v9 or v8 PBX
| Status | Closed |
| Id | 64709 |
An old v5 SoftwarePhone license installed on a v9 PBX did not work for v5 SoftwarePhones
Status:
inno_lic.cpp
inno_lic.h
DECT: Registration string for test registration
| Status | Closed |
| Id | 64766 |
The registration string for the test registration is fixed to work with V9.
PBX:OEM Voicemail license did not work
| Status | Closed |
| Id | 64769 |
collateral damage from supporting licenses from different versions on a PBX Status: inno_lic.cpp
SIP: Interworking of "Q.931 CALL PROCEEDING" into "183 Session Progress"
| Status | Closed |
| Id | 64770 |
Required if only CALL PROCEEDING and no ALERTING is received.
IP241: Diverting/Transfering party was not shown
| Status | Closed |
| Id | 64775 |
If name (not number) was called and call was diverted/transferred
the diverting/transferring party (dialed name) was invisible.
phone: Secretary/Executive function of Partner key does not work when "Subscribe for Dialog Info" is checked.
| Status | Closed |
| Id | 64783 |
When "Partner Type" is 'Secretary' or 'Executive' and "Subscribe for Dialog Info" is checked the key does not work correctly because this functionalities require group indications. To prevent configuration errors "Subscribe for Dialog Info" must be supressed in this case.
Linux: Reset required links
| Status | Closed |
| Id | 64813 |
Reset required links in the Linux XML pages fixed.
ip800 trace telling wrong information about power source
| Status | Closed |
| Id | 64826 |
PCBs since V300 cannot detect POE power and trace therefore told 'not powered'. Status: ip800.cpp
Phone: When activating a callback through function-key, the stored number behind the function-key is also called
| Status | Closed |
| Id | 64830 |
problem: phone: When activating a callback through function-key, the stored number behind the function-key is also called. press function key invokes the callback, release function key in idle mode invokes idle-function-key action
solution: fixed in code: now ignoring function key invocations through function-key-release if popup active
files: app_fkey.cpp forms.* forms_impl.*
products: all telephones
risks: none
Linux: Fix address from ETH1 network
| Status | Closed |
| Id | 64877 |
Linux network configuration is not working if Linux gets a fix ETH1 network address. The default gateway address is wrong from ETH0. Now the network mask and the default gateway can be configured.
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.
PBX CDRs: Wrong Display Name sometimes with Pickup
| Status | Closed |
| Id | 64917 |
when calls routed between locations were involved Status: pbx.cpp
PBX CDRs: Transfer not indicated if transfer initiated by myPBX
| Status | Closed |
| Id | 64924 |
in case of transfer with consultation Status: pbx_client.cpp
SIP: Interworking of calls with Q.931 Bearer Capability "Unrestricted digital information" rejected
| Status | Closed |
| Id | 64932 |
Calls with Q.931 Bearer Capability "Unrestricted digital information" were rejected.
PBX: Conference trap
| Status | Closed |
| Id | 64936 |
A zero pointer trap in the broadcast conference PBX object is fixed.
PBX CDRs: On Transfer to Waiting, the "connected" message did not contain peer
| Status | Closed |
| Id | 64955 |
Happend with transfer with consultation to a waiting queue Status: pbx_wait.cpp
IPVA, RAM-Dependent Flash Limitation
| Status | Closed |
| Id | 64961 |
Flash Utilization must be limited with RAM <= 64MB
To prevent loss of flash data when uploading IPVA v9hf1:
1)Perform Maintenance/Config/Download
2)Upload the new ipva.bin
3)Perform Maintenance/Config/Upload with the step 1) config.
Note:
-The flash directory space is 13MB with 64MB Ram
-The max. flash directory space is available from on 192MB Ram
IP222: changes for new hardware
| Status | Closed |
| Id | 64967 |
- Reduce average current through IR-LED
* Re-init LCD orientation (fight against resets of the display controller)
Status: orchid_lcd.cpp
Setup wizard used wrong default password on some products
| Status | Closed |
| Id | 65024 |
If no password was entered on
- ip6010
- ip2010
- ip1060
- ip302
- ip305
- ip6000
- ip2000
A wrong password was configured
Status:
ip6000/config_wizard.txt
ip24/config_wizard.txt
ip6010/config_wizard.txt
myPBX: Usability improvements for call transfers
| Status | Closed |
| Id | 65032 |
Blind transfer of incoming alerting calls.
Transfer of outgoing alerting calls.
Autoselect new outgoing calls.
Hide transfer buttons on the selected call.
files: client.js
PBX Mobility: Use of RTP-DTMF was prohibited even if media terminated in PBX itself
| Status | Closed |
| Id | 65036 |
For DTMF within a call (e.g. ** for R-Key) RTP-DTMF cannot be used end-to-end, because the Mobility object would not see these DTMFs.
If media is terminated in the PBX (because of media-relay or whenever tones are played by the PBX itself) RTP-DTMF can and should be used
Status:
channel.cpp
PBX CDRs: Wrong information if forward on other PBX
| Status | Closed |
| Id | 65041 |
The leg1 info from the other slave was not evaluated for CDRs
Status:
pbx.cpp
ep_lib.cpp
ep_lib.h
fty.cpp
fty.h
h450.cpp
Sockets: misleading trace: connect & bind events carry uninitialized IP6 addr
| Status | Closed |
| Id | 65045 |
problem: sockets: misleading trace: connect & bind events carry uninitialized IP6 addr
solution: now set to 0, fixed in code
files: types.h socket.h
products: all products
risks: none
ip241 - G722 3party conference did not work
| Status | Closed |
| Id | 65078 |
When a 3party conference was setup with at least one G722 connection involved the microphone data of one party was terminated at the conference point
DECT: Wrong packetsize sometimes used after renegotiation
| Status | Closed |
| Id | 65086 |
This could cause interop problems with some PBX, which was not able to receive media packets above a certian size. Status: h323ch.cpp
VM, Session Guard Timer Not Stopped For <pbx-record>
| Status | Closed |
| Id | 65097 |
The following script didn't work correctly with v9: http://wiki.innovaphone.com/index.php?title=Howto:A_Simple_Recording_Solution_On-Top_of_The_innovaphone_Voicemail
myPBX: Improvements of French translation
| Status | Closed |
| Id | 65115 |
Transfer direct -> Transférer
Transférer -> Mettre en communication
files: pbx_client_localisation.cpp
SIP: Use DNS query without recursion on closed federation interfaces
| Status | Closed |
| Id | 65126 |
Keep DNS server from doing automatic recursion when resolving domain names on closed federation interfaces.
H.323: Don't forward G.729B capability if silience compreession not enabled
| Status | Closed |
| Id | 65133 |
This solves quality issues some SIP provider have with G.729B. Status: h323ch.cpp
DECT: Master Call GUI close/clear for OEM
| Status | Closed |
| Id | 65137 |
The link 'close' and 'clear' of the DECT Master call page did not work for an OEM device and is fixed now.
Phone UI: Truncate h323id on header of idle screen to reserve space for e164 number
| Status | Closed |
| Id | 65140 |
Long h323id's may take all available space and keep e164 number from being displayed. Status: forms_color.cpp
Network Link GUI
| Status | Closed |
| Id | 65142 |
The network (ETH0/ETH1) Link page was not reloaded after pressing 'ok' or ' clear' in IP800 and IP1200 and is fixed now.
myPBX: Sessions sometimes get stuck
| Status | Closed |
| Id | 65143 |
Session IDs used the full dword scope but btree_compare can only handle values in the positive integer scope.
files: pbx_client.cpp
firmware of a certain image size did not boot from scattered flash segments
| Status | Closed |
| Id | 65154 |
when booting from scattered flash segments the firware is decompressed in smaller chunks. A false alarm was raised after successful decompress for a last chunk smaller than 5 byte.
PBX CDRs: Missing transfer-to on operator connect, when calling waiting from other PBX
| Status | Closed |
| Id | 65178 |
This was a general problem with transfer on another PBX. Happens with normal transfer on other PBX as well
Status:
pbx.cpp
pbx_wait.cpp
Phone: Broken "Display Name on Pickup/Partner Key" Option
| Status | Closed |
| Id | 65179 |
problem: Broken "Display Name on Pickup/Partner Key" : pickup function key always displaying numbers, never H.323 alias (name)
solution: fixed in code
files: box/forms/lcd/phone_forms_impl.cpp phone/app/app_disp.cpp app_ctl.h
products: all inca telephones
risks: none
SIP-H323 calls with SRTP: No media after multiple Hold/Retrieve
| Status | Closed |
| Id | 65185 |
After first Hold/Retrieve there was no SRTP, after the next Hold/Retrieve very often no media Status: h323ch.cpp
Phone UI: Truncate h323id on header of idle screen to reserve space for e164 number
| Status | Closed |
| Id | 65258 |
problem: Long h323id's may take all available space and keep e164 number from being displayed.
solution: new user pretty print algorithm now tries to display the whole e164 number and fills the remaining space left aligned with h323 id. ellipsis is only one char on the phone lcd and notes that the text ahead of it has been shortened. following cases are possible:
1) |123...| number too long
2) |123456| number just right
3a)| 12345| no name found or less than 2chars left for name
3b)| 1234|
4) |n... 1| name shortened with ellipsis, number complete
5) |nam 23| enough room for name and number
6) |name | no number - enough space for name
7) |name01| no number - enough space for name (just right)
8) |nam...| no number - name shortend with ellipsis
files: forms_impl.cpp
products: all inca telephones
risks: none
PBX: Busy on ... Calls should be executed on calls from on object to itself
| Status | Closed |
| Id | 65260 |
was disabled in 2008, no idea what this should be good for. Can be very confusing Status: pbx.cpp
PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions"
| Status | Closed |
| Id | 65261 |
If "Execute Member Diversions" was checked a call to Broadcast was also sent to CFB destination Status: pbx_bc.cpp
IP241: Call list does not show display name if call was diverted
| Status | Closed |
| Id | 65269 |
Display name of the diverted-to party was not shown in call list.
Also truncate names and numbers to avoid interference with date and time.
Gateway: Not possible to enter wildcards ('.') in interface maps
| Status | Closed |
| Id | 65280 |
wrong check for correct value Status: gk.cpp
Gateway: Configured signaling port got lost, when ediiting interface maps
| Status | Closed |
| Id | 65303 |
The signaling port was reset to the standard port when saving interface mappings Status: gk.cpp
PBX: Name Identification was missing when initaiting IM Session
| Status | Closed |
| Id | 65304 |
No display name of initiating party was sent Status: pbx.cpp
Trap in rarely used OS function bufman::remove
| Status | Closed |
| Id | 65338 |
could result in negative length of buffer Status: os.cpp
phone: blinking of central MWI-LED could not be disabled for a MWI function key without an own LED
| Status | Closed |
| Id | 65354 |
When the MWI function key was assigned to a key without an own LED the central MWI-LED did blink although "Active State/LED" was set to "off".
Associating a call from a Gateway object with Gateway object Node did not work if CGPN represented an incomplete Number
| Status | Closed |
| Id | 65359 |
This can result in not finding the correct Node Extern interface Status: pbx.cpp
HTTP /home did not work for URLs containing more then one '/'
| Status | Closed |
| Id | 65364 |
/home could be used for PBX0/MY/start.htm, but this did nor work because of multiple '/' Status: http.cpp
PBX UI: Registration with Admin Pwd misleading, PBX Pwd is better
| Status | Closed |
| Id | 65370 |
The Admin Pwd could be understood as the General/Admin password. PBX Pwd should be pretty clear what this is.
Status:
pbx_edit_object.xsl
pbx_general.xsl
phone_orchid: handsfree mode does not work with DSP Fimware 494E003ce4.640.11
| Status | Closed |
| Id | 65379 |
the calling party was fully echoed when the called party was in handsfree mode.
| Status | Closed |
| Id | 65394 |
Refresh staus bar in menu screen
PBX-SOAP: Trap if initiating multiple outgoing calls from a Waiting object at the same time
| Status | Closed |
| Id | 65418 |
Some applications do this to deliver voice messages
Status:
pbx_wait.cpp
pbx_wait.h
SIP: Gateway interfaces without registration need another config option
| Status | Closed |
| Id | 65436 |
New config option "Local Domain" added.
Required for URI processing on incoming calls.
If domain part of destination URI matches the local domain,
the domain part is stripped from the destination information.
Gateway: Record URL at SIP interface was lost when Internal registration was configured
| Status | Closed |
| Id | 65443 |
UI problem Status: gk.cpp
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
VM, email attachments weren't sent for https URLs
| Status | Closed |
| Id | 65458 |
VM, email attachments weren't sent for https URLs
PBX: Call-Intrusion could result in wrong name display
| Status | Closed |
| Id | 65462 |
esspecially for silent intrusion
Status:
signal.cpp
h450asn1.h
pbx.cpp
pbx.h
AD Replication, Alphanumeric symbols were rejected in maps
| Status | Closed |
| Id | 65499 |
A bug introduced by code-adaption from v8 to v9. I.e. this problem is new with v9 and does not occur under v8.
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
myPBX: Default values for configuration of external LDAP search
| Status | Closed |
| Id | 65593 |
Use the same default values as on the phone.
Name Attributes: "sn,givenName,company"
Number Attributes: "telephoneNumber:D,homePhone:P,mobile:M,:@"
files: pbx_client.cpp
Phone: Bei "Nachricht antworten" wird von der gerade aktiven Reg gesendet
| Status | Closed |
| Id | 65630 |
Es sollte aber von der Reg gesendet werden, auf der die Msgs empfangen wurde.
Wird ja bei Voice-Calls auch so gemacht.
IP241: Connected call was displayed as transferred call
| Status | Closed |
| Id | 65641 |
When name was dialed instead of number, the connected number (received with connect message) caused display to show transfer condition. Status: phonesig.cpp
myPBX: Outgoing IM session did not reach state "up" on instant connect
| Status | Closed |
| Id | 65651 |
The following message was missing when the remote entpoint connected the call right away: <update-im id="0" state="up"/>
files: pbx_client.cpp, pbx_client.h
myPBX: Performance improvements
| Status | Closed |
| Id | 65654 |
Searching in lists was suboptimal
Removed unused method
files: pbx_client.cpp, pbx_client.h
Phone: Croatian translation completed (some missing entries)
| Status | Closed |
| Id | 65661 |
problem: Croatian translation completed (some missing entries)
solution: translated
files: phone/text/*
products: all phones
risks: none
IP241: Fix for display rendering
| Status | Closed |
| Id | 65678 |
Zeroize png object when image loading fails.
IP241: Fix for multi line editor
| Status | Closed |
| Id | 65686 |
Fix for multi line editor
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.
myPBX: Forward instant messages only once in order to prevent loops
| Status | Closed |
| Id | 65699 |
Forwarded messages are detected by comparing the remote endpoint and the sender name.
files: pbx_client.cpp
test: unit-tests\\client\\iteration07\\402-multisession-loop2
myPBX: Use displayname as sender name of instant messages
| Status | Closed |
| Id | 65704 |
The cn was used before. Now one of the following attributes is used: dn, cn, h323
files: pbx_client.h, pbx_client.cpp
test: unit-tests\\client\\iteration07\\250-singlecall-dn
IP241: Trap on display rendering
| Status | Closed |
| Id | 65726 |
When entering too many digits while preparing block dialing.
Phone: Message reply uses current registration instead of registration which received the incoming message
| Status | Closed |
| Id | 65729 |
problem: Message reply uses current registration instead of registration which received the incoming message. Now message sending functions also take message source as parameter.
solution: fixed in code.
files: phone/app/* phone/list/* phone/dir/*
products: All Telephones
risks: none
IP241: Fix for XML screen dump
| Status | Closed |
| Id | 65737 |
XML screen dump is only used for automated testing.
MWI does not work in various Node/Pbx combination
| Status | Closed |
| Id | 65750 |
MWI does not work in various Node/Pbx combination
Appended called number to CF at gateway object needs to be adjusted acording to destination node
| Status | Closed |
| Id | 65754 |
Reroute from master to slave in an e164 scenario for IP down did not work
Status:
pbx.cpp
pbx.h
pbx_api.h
Trap: When Dectmaster registers user at PBX using SIP protocol
| Status | Closed |
| Id | 65798 |
Occurred on IPBL[4.1.22]
PBX: Routing did not work correct for calls between Nodes/PBXs for some Facility Calls
| Status | Closed |
| Id | 65825 |
Some Facility calls are routed without checking for call-forwards (e.g. MWI interrogation). When these calls where routed to a different PBX, routing to an object not within root node did not work, if caller was not in root node Status: pbx.cpp
PBX User Interface: Text refers to pages PBX/General or PBX/Password which do not exist anymore
| Status | Closed |
| Id | 65834 |
Should be pages PBX/Config/General and PBX/Config/Security
Status:
pbx_global.xsl
pbx_objs_left.xsl
pbx_regs.xsl
pbx_soap.xsl
pbx_import.xsl
pbx_export.xsl
pbx_dyn.xsl
pbx_calls_view.xsl
pbx_calls_classic.xsl
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).
SIP: Allow message bodies larger than 20kB
| Status | Closed |
| Id | 65880 |
Problems handling Event Notification Extension for Resource Lists.
Bodies can grow very large.
ip6010 - ethernet statistics counter values missing
| Status | Closed |
| Id | 65905 |
only the counter names but no values were displayed on the ethernet statistics page
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
SIP: Media negotiation problem during hold/retrieve
| Status | Closed |
| Id | 65995 |
Media negotiation problem during hold/retrieve on local media
debug.xml, Tracing Modified
| Status | Closed |
| Id | 66004 |
A few trace options were altered
-SIP
-LDAP
Web-UI: Removed config option "external style sheet"
| Status | Closed |
| Id | 66006 |
Does not work in version 9.
Phone: Changing current presence activity does not remove presence note
| Status | Closed |
| Id | 66017 |
Remove presence note when changing presence activity.
Phone: Pre-load more directory entries
| Status | Closed |
| Id | 66026 |
Until to now the number of pre-loaded directory entries was defined by number of lines on screen.
More entries have been loaded on demand when scrolling up or down.
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
IPVA, More Watchdog Info
| Status | Closed |
| Id | 66029 |
a bit more diagnostics info is written in case of a watchdog
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
H.323: Media Negotiation problem, with reverse media calls thru media-relay/exclusive media interface
| Status | Closed |
| Id | 66037 |
This could result in no media with mobility calls to sip provider Status: h323ch.cpp
| Status | Closed |
| Id | 66077 |
Now, the Gateway menu is added again in V9 for the IP1200.
Web-UI: Fixed links into wiki reference
| Status | Closed |
| Id | 66079 |
Fixed dead links into wiki reference
PBX/Access: Visibility defined for a domain, should just add to the other definitions
| Status | Closed |
| Id | 66104 |
Other then a definition based on a name, which should overwrite everything else Status: pbx.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 )
Kerberos: Wrong RC4 key usage type for authenticators
| Status | Closed |
| Id | 66113 |
The AP-REQ message uses key usage type 7 if it is encapsulated in an TGS-REQ.
files:
kerberos_client.cpp
kerberos_prot.cpp
kerberos_prot.h
Config upload error message changed to show all skipped lines
| Status | Closed |
| Id | 66133 |
Request from Ascom Status: servlet_post.cpp servlet_post.h
DECT: Call waiting without Feature Codes
| Status | Closed |
| Id | 66153 |
Call waiting is not working if the feature codes are disabled and the first call is in alerting state. Now the following call is not rejected.
HTTP-Server: Configuration page was broken
| Status | Closed |
| Id | 66166 |
The processing of the configuration form accidentially produced a too long command line.
files: http.cpp, http.h
HTTP-Server: Restrict number of filters and paths on configuration page
| Status | Closed |
| Id | 66167 |
The GUI looked like the user could define more than ten entries. Also there was a trap when the user defined one CF path too much.
files: http.cpp, http.xsl
PBX-CDR: Forwarded call representation improved
| Status | Closed |
| Id | 66169 |
CDR for incoming call, which shows the forwarding as well and another CDR for the outgoing call to be used for billing only Status: pbx.cpp
DECT: Delete duplicate LDAP 'pbx' <gw> items
| Status | Closed |
| Id | 66174 |
Now duplicate LDAP 'pbx' <gw> items are deleted by the DECT users module.
SoftwarePhone: GN 8120 USB volume keys
| Status | Closed |
| Id | 66204 |
The volume up and down key mapping for the GN 8120 USB is fixed.
| Status | Closed |
| Id | 66216 |
Could be confusing Status: pbx_xml.cpp
PBX External UC: Update presence of users with correct External UC set only
| Status | Closed |
| Id | 66223 |
If presence info was received on an External UC interface for users not configured for External UC, the presence was still updated. This was wrong Status: pbx_uc.cpp
Phone: Changing or resetting the presence activity leaves presence note as is - should delete it
| Status | Closed |
| Id | 66246 |
problem: phone: Changing or resetting the presence activity leaves presence note as is. Which usually does not reflect what the user wants; remove the (absence) note.
solution: fixed in code, remove note upon activity chnage from function key or in user-menu
files: phone/app/app_disp.cpp phone/app/app_gui.cpp
products: all telephones
risks: deletes the old user activity note on the phone.
Phone: Function key "Park" not displaying parked calls
| Status | Closed |
| Id | 66258 |
problem: phone; Function key "Park" not displaying parked calls
solution: fixed in code
files: forms/lcd/forms_phone_impl.cpp, forms/lcd_lib/forms_color.cpp
products: all telephones
risks: none
phone: DHCP provided configuration was not displayed under "Phone/User x/General/Options"
| Status | Closed |
| Id | 66268 |
DHCP provided configuration values always override locally configured values. A locally changed value is stored and displayed in the edit fields but has no effect as long as as DHCP provides another value. To make this clear the currently active values are displayed below or behind the corresponding edit field.
But instead of the active values the local values were displayed.
Thus when ticking for example "Enable Secure RTP", 'enabled' was displayed behind the checkmark even when Secure RTP was not enabled in the DHCP configuration.
IP6010-CF: Kingston compact flash was not recognized
| Status | Closed |
| Id | 66269 |
the card was not recognized because a register was wrongly initialized.
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.
Kerberos: Disable RC4 encryption by default
| Status | Closed |
| Id | 66352 |
RC4 encryption is now disabled by default. It can be turned on on the General/Admin page.
Status:
kerberos_if.h
kerberos_provider.h
kerberos_provider.cpp
command.h
command.cpp
command.xsl
IPVA, Double-Fault Handling Should Skip Trace History
| Status | Closed |
| Id | 66362 |
Otherwise a triple fault occurs in case of a corrupted trace buffer
PBX administration: Slave password is lost when configuring myPBX
| Status | Closed |
| Id | 66380 |
When the configuration of myPBX was submitted on the PBX/Config/myPBX page the slave password was reset.
Status:
pbx.cpp
pbx_client.xsl
SOAP, Send leg2Info.originalCalled Info
| Status | Closed |
| Id | 66407 |
As CallInfo.No with type="leg2orig" Status: pbx_xml.cpp
V9 Hotfix 2 (90600.02)
Changes included in Version 9 hotfix2 Definition
New Features
X.509: Add key usage to certificate requests
| Status | Closed |
| Id | 66413 |
The Microsoft CA (standard) does not write the key usage into the certificate if it is not specified in the request.
configurable DHCP client hostname to be sent in DHCP requests
| Status | Closed |
| Id | 66487 |
By default the hostname sent in DHCP client requests is the catenation of the short device name and the last 3 bytes of the mac address, for example IP241-2a-00-c6.
The MS DHCP server creates a DNS entry from this name and the assigned IP address.
A configurable name may better meet existing naming conventions.
PBX-SOAP: Support for Fax Server interface
| Status | Closed |
| Id | 66590 |
UserFindDestination function added
UUI Protocol Discriminator fixed
Status:
pbx_xml.cpp
pbx_xml.h
Relay: Forward facilities to local destinations
| Status | Closed |
| Id | 66629 |
The relay forwards all facility events if interworking is disabled and the destination is local. This is needed for some interfaces.
PBX-SOAP: Support receiving of UUI
| Status | Closed |
| Id | 66758 |
needed for Fax Server
| Status | Closed |
| Id | 66801 |
Now it is possible to mute the ringer with the context menu item 'Mute ringer' of the tray icon. This option is not saved for the next application start.
Allow configuration of SRTP crypto suite, to be used for media proposals
| Status | Closed |
| Id | 66810 |
This is needed, because some equipment (e.g. Microsoft Lync) is not able to adapt to whatever proposal it receives
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.
Fax: Progress informations
| Status | Closed |
| Id | 67228 |
The FAX interface provides progress and error informations, now.
SIP: Presence: Support for presence/person/activities/note
| Status | Closed |
| Id | 67363 |
Not only decode presence/person/note
additional choices for 'loop current' on ip22/24 ip302 and ip28 gateways
| Status | Closed |
| Id | 67396 |
for special applications higher loop currents above 26mA are requested. Implementation adds 30mA, 35mA and 40mA values. Status: si3210_drv.cpp si3241_drv.cpp
SIP: Web UI for cause code mapping
| Status | Closed |
| Id | 67440 |
Configuration of mappings between ISDN cause codes and SIP response codes.
http://x.x.x.x/SIP/mod_cmd.xml?xsl=sip.xsl
http://x.x.x.x/TSIP/mod_cmd.xml?xsl=sip.xsl
http://x.x.x.x/SIPS/mod_cmd.xml?xsl=sip.xsl
Bug Fixes
IP2x2x: Reset USB Hub, OHCI and LCD during softreset
| Status | Closed |
| Id | 66418 |
was missing on softreset.
Also enable/disable USB power added
H.323 Media Re-Negotiation Problem if reverse Media call to Media-Relay/exclusive Coder Interface
| Status | Closed |
| Id | 66547 |
This could happen if a call from an mobile endpoint was initiated to a SIP Trunk with media-relay/exclusive coder and re-negotiation happens, e.g. because the call was sent to a TONE interface before
PBX-SOAP: When monitoring a Gateway object, dialing information received with overlap receiving was missing
| Status | Closed |
| Id | 66559 |
Applications, which monitor a gateway object, and want to connect the call depending on the number dialed, need this information
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
SIP: SIP/TLS registration was not accepted
| Status | Closed |
| Id | 66625 |
Inbound SIP/TLS connection was not accepted since client certificate was checked.
Usually client certificates are self signed and not trusted by innovaphone PBX.
Mutual TLS was done.
Status:
siptrans.cpp
PBX-CDR: Generate CDRs for Broadcast objects
| Status | Closed |
| Id | 66628 |
No CDR was generated from Broadcast object.
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
phone: local/default coder config was overriden by unsupported or wrong DHCP config
| Status | Closed |
| Id | 66716 |
DHCP provided coder config was not checked before overriding the locally defined configuration or the default values.
Now a ';' separated list of coder configurations can be specified as well for the general as for the local coder, for example
G729A,60,x;G723-53,60,x/G722,30;G711,30
If no coder in the list is supported by the device the respective default coder and frame size is used together with the SRTP-flag(x) and the DTMF-flag(n) from the last entry in list.
PBX-CDR: In CDRs generate for a Gateway object with prefix, the prefix was removed from the number
| Status | Closed |
| Id | 66719 |
For a call to a Gateway object with Prefix checkmark set, the local number was provided with out the prefix (the number of the Gateway object itself). This was different from the behaviour of a Trunk object and the number could not be used without the knowledge of the object.
SIP: DNS resolving failed on no-reg interfaces if DHCP takes longer than 10sec from boot
| Status | Closed |
| Id | 66720 |
DNS resolving does not work as long as local IP config is not complete.
DNS resolving is part of initialization process of SIP interfaces.
SIP interfaces give up on DNS after 10 secs of retrying.
Local IP config may take longer if DHCP is used.
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.
PBX-SOAP: Don't show dummy call, which is generated by UserCall
| Status | Closed |
| Id | 66748 |
The UserCall (make outgoing call) function initiates a call to the user phone first and then transfers this call to the outgoing call. This initial dummy call should not show up as call, when monitoring the user from a different application
SIP: Lync interoperability
| Status | Closed |
| Id | 66750 |
Lync uses same Rseq number on several provisional responses.
Must send PRACK for each of it.
phone orchid: DTMF digits typed in handsfree mode sometimes duplicated when speaker volume is high
| Status | Closed |
| Id | 67092 |
an audible feedback for the typed digit is sent to the speaker. when the speaker volume is high this feedback is received by the microphone and detected as if it was generated for example by a DTMF tone generator.
Now this detection is disabled while typing DTMF digits.
phone orchid: Audible Signal after alerting for partner/pickup key did not work
| Status | Closed |
| Id | 67094 |
audible signal was triggered but not played
myPBX launcher: Accept language code from configured URLs
| Status | Closed |
| Id | 67137 |
Use the language code from the URL (lang=it), if configured. Otherwise use the language of the current windows login, like in prior versions.
myPBX: Workaround for redrawing problems in IE9
| Status | Closed |
| Id | 67138 |
IE9 does not redraw the presence note when the activity image stays the same.
This can be fixed by avoiding to replace an image or text by the same image or text.
IP241: embedded test: Function key LED test didnt work
| Status | Closed |
| Id | 67142 |
init was missing Status: forms_color.cpp
IP110 enters boot loop after power cycle with v9hf1
| Status | Closed |
| Id | 67158 |
... other phone_incas like 200a 230 240 are not affected.
To recover from this gwload is needed.
Status:
phone_inca.cpp
PBX CF Filter worked different if previous divertion with CFU or CFNR
| Status | Closed |
| Id | 67212 |
This was confusing
PBX: Syslog for registration attempt to unknown user wrong
| Status | Closed |
| Id | 67224 |
The reason for the reject was provided as "Missing Authentication". It is now changed to "User unknown".
IP241: Disable LINK LEDs not implemented
| Status | Closed |
| Id | 67225 |
on IP240 this function is available Status: orchid_drv.cpp .h .xsl
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
ip22/24/28/302/305: Reduce memory footprint
| Status | Closed |
| Id | 67304 |
The new DSP code increases the needed memory, on boxes PBX (IP302/IP305)
the memory may be already low. So we split the binary with the extra IP28 DSP code to save some bytes.
Status:
ac_494.cpp
ip24.mak
ac_dsp3.mak
Web-UI: Vertical alignment of ui elements
| Status | Closed |
| Id | 67313 |
Diagnostics/Counters and others
Status:
work.css
log_counters.xsl
reg_edit_fkey.xsl
IPxx10 traps when clicking on Linux stop
| Status | Closed |
| Id | 67314 |
wenn Linux gestopped wird, trapt das Gerät
RTCP: Interval reduced to 5 secs (from 10 secs)
| Status | Closed |
| Id | 67322 |
For interop with Lync.
SIP: Trap when receiving unsolicited NOTFIY with dialog-info
| Status | Closed |
| Id | 67368 |
Trap when not subscribed for dialog-info
SIP: Handling if transport errors on connection oriented transport types
| Status | Closed |
| Id | 67398 |
Handling of transport error was added for SIP/TCP and SIP/TLS.
Transaction timeout (32secs) is no longer required to elaps.
SIP: NOTIFY sent after 302 moved temporarily
| Status | Closed |
| Id | 67450 |
After processing "302 moved temporarily" on an outbound call a NOTIFY (sipfrag) was sent.
Compact Flash: possible trap when deleting directories
| Status | Closed |
| Id | 67456 |
There has been a chance for a trap when deleting directories.
V9 Hotfix 3 (90600.03)
Changes included in Version 9 hotfix3 Definition
New Features
SIP: Send OPTIONS at configurable interval
| Status | Closed |
| Id | 67519 |
For keep alive pruposes on interfaces without registration.
Required for Lync interoperability.
(config change TSIP /options-interval 30)
Status:
sip.cpp/h
siptrans.cpp/h
SIP: Fast re-routing on gateway interface w/o registration
| Status | Closed |
| Id | 67593 |
OPTIONS is used to check availability of remote peer.
If OPTIONS fails, not trying to send INVITE.
SIP: New config file option /no-ms-acceptedby
| Status | Closed |
| Id | 67665 |
Some IP phones get confused by this Microsoft extension.
Cisco SPA 303 and Cisco SPA 962 don't stop ringing anymore
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
PBX: Translation of Cause "Call Rejected" to Cause "User Busy" for endpoint objects only
| Status | Closed |
| Id | 67668 |
If the cause is received from a gateway, it is forwarded transparently now
myPBX: Show active sessions on the admin interface
| Status | Closed |
| Id | 67708 |
The active myPBX sessions are shown on the page PBX/myPBX.
Phone: Added new function keys to Phone-UI (Toggle & Prepare Override)
| Status | Closed |
| Id | 67729 |
Description: Phone: Added new function keys to Phone-UI (Toggle & Prepare Override)
PBX: New presence activity "do not disturb"
| Status | Closed |
| Id | 67777 |
If a users sets this presence activity no calls are passed.
PBX: Allow CFB on Gateway Type Objects
| Status | Closed |
| Id | 67827 |
A CFB is triggered by a User Busy. If a CFB is used for example at a Trunk, the CFB is executed when the called remote user returns busy. Because this may be unexpected the CFB was not executed at a Gateway Type Object.
It is now enabled again, because it is useful when connecting external systems which return busy to indicate an out of channels situation
PBX Trunk/Gateway: Round robin within registrations to same device, different devices sequentially
| Status | Closed |
| Id | 67835 |
This way both round-robin or sequential usage of gateways can be configured
improved test for Timeslot Switch Chip of ip6010 ip3010 ip0010 ip1060 and ip6000
| Status | Closed |
| Id | 67840 |
intension is better analysis of hardware problems Status: idt72_drv.cpp, idt72_drv.h
ip22/24/28/302/305/6010/3010/1060/241/222/232: False DTMF detects
| Status | Closed |
| Id | 67844 |
Seen on IP222 and IP6010 Status: ac_dsp3.cpp
phone_orchid: builtin color display test
| Status | Closed |
| Id | 67845 |
After the builtin test function has been started the display test mode is entered when the 'Esc' key is pressed. Numeric keys trigger a full screen test display, all other keys stop the display test mode.
To the keys 0..9 the following patterns are assigned:
DarkGray, White, Grey, Black, Red, Green, Blue, Yellow, Cyan, Magenta
IP-DECT: Configuration XML data for OEM device
| Status | Closed |
| Id | 67858 |
A new configuration XML attribute is added for changing a OEM GUI.
Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type
| Status | Closed |
| Id | 67905 |
Description: Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type. Currently, only modification of preset function keys is disabled, but the creation of new ones enabled and possible.
Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely
| Status | Closed |
| Id | 67943 |
Description: Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely. See /hide-mask option to PHONE ADMIN-UI in wiki for more information.
IP-DECT: OEM registration string
| Status | Closed |
| Id | 68058 |
The Radio registration string of an OEM device is changed to check a OEM license in the DECT Master.
SIP: Diagnostics of transport error "Remote server certificate mismatch"
| Status | Closed |
| Id | 68142 |
If remote sverer certificate does not match the destination domain name
signaling connection is refused by client.
myPBX: Support for WebEx meeting passwords
| Status | Closed |
| Id | 68172 |
In WebEx a meeting can have a password that must be entered by the attendees when they join. Some WebEx accounts can only create meetings with passwords.
The possibility to configure a global meeting password is added to the PBX/Config/myPBX page.
Phone: Message function key
| Status | Closed |
| Id | 68208 |
Description: Phone: Message function key. Multifunctional depending on number of unread messages. Stores one prepared message (with destination and message text) and presents the new message screen when invoked. If incoming messages pending, display the letter/message icon and jump to incoming-messages subscreen upon invocation.
phone: Finnish translations updated
| Status | Closed |
| Id | 68395 |
no english placeholders anymore, Texts begin with upper case letter
IP150: dimming of key LEDs and LCD backlight
| Status | Closed |
| Id | 68414 |
to save power in special environments the key LEDs can be dimmed by
config add KEYS0 /light-off
The lcd backlight can be configured the usual menu way on the phone.
HTTP-Client: Allow user names longer than 16 characters
| Status | Closed |
| Id | 68499 |
Now user names with up to 64 characters are allowed.
| Status | Closed |
| Id | 68504 |
Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu. Usage: enter number or search for phonebook entry, press menu-key, scroll down to choose call-forwarding (always, busy or no-reply) and acknowledge choice in CF-screen.
IP-DECT: Allow setting empty text for idle display
| Status | Closed |
| Id | 68553 |
Some handsets will not show signal strength and battery symbols if idle display is defined, they must be set to empty string.
ac_dsp3: support echo canceller trace
| Status | Closed |
| Id | 68649 |
required to analyze echo canceller problems
Phone: On IP240, make OK key a headset activation key (along with Space Key)
| Status | Closed |
| Id | 68774 |
Phone: On IP240, make OK key a headset activation key (along with Space key).
Enabled from Phone/Preferences : Use Newline/OK Key as Headset Key.
Bug Fixes
IPxx10: error handling in sata driver
| Status | Closed |
| Id | 67229 |
Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.
DECT: IP6000/IP6010/... default config Master mode off
| Status | Closed |
| Id | 67479 |
Now the Dect Master is in mode off by default for the IP6000/IP6010/...
SoftwarePhone: DTMF to voice mail
| Status | Closed |
| Id | 67563 |
Now, sending DTMF with myPBX is correctly working. This fixes sending DTMF to voice mail.
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.
| Status | Closed |
| Id | 67618 |
Phone: Main menu scrolling below last item broken.
1st item hould be activated upon down arrow press (done) and screen focus moved up (not being done - bug).
cpld update not working on ip241
| Status | Closed |
| Id | 67629 |
-
Status:
phone_orchid.cpp phone_orchid:config.h xilinx.cpp
and as consequence also ip6010:config.h
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.
Phone: Automatic key repeat did not work on IP222/IP232
| Status | Closed |
| Id | 67661 |
Automatic key repeat did not work on phone w/o alpha keyboard
Call to Voicemail did not work anymore with SRTP
| Status | Closed |
| Id | 67672 |
This was a collateral damage from fix
http://wiki.innovaphone.com/index.php?title=Support:DVL-Roadmap_Firmware_V9#Allow_configuration_of_SRTP_crypto_suite.2C_to_be_used_for_media_proposals
Phone: long function key titles hide idle screen information
| Status | Closed |
| Id | 67681 |
Description: Phone: long function key titles hide idle screen information.
Fixed: important idle screen information now shortens the amount of displayed function key name. Following information is now displayed over the function key text: a) crossed bell icon on do-not-disturb (lines 2+3) b) CFU + CFU-destination (lines 2+3) c) missed calls, unread messages and waiting callbacks (line 4)
PBX Waiting: Call forwarded with DTMF mapping was shown in myPBX for each registration
| Status | Closed |
| Id | 67682 |
The call was not sent with the original conferenceID, so myPBX could not detect that it was in fact the same call
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.
phone_orchid: microphone is not mute on a call intrusion in silent monitoring mode / microphone cannot be muted in a conference
| Status | Closed |
| Id | 67704 |
On a call intrusion in silent monitoring mode the microphone of the intruding party must be mute. In a conference the micro should be muted when the micro key is pressed and unmuted when the micro key is pressed again.
Muting the microphone did work when only one call was active but not when two calls were active as in a intrusion/conference.
Phone: CLIR on text messaging did not work
| Status | Closed |
| Id | 67710 |
CLIR on text messaging did not work
SIP: Interoperability with Lync: Handling of REFER from Lync
| Status | Closed |
| Id | 67713 |
REFER for blind transfer was rejected with "406 Not Acceptable",
due to absence of user part in refer-to URI.
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
Phone: Calls received with CLIR appear in call list with an empty entry
| Status | Closed |
| Id | 67722 |
Description: Phone: Calls received with CLIR appear in call list with an empty entry; now fixed to display the CLIR text: anonymous/unbekannt/...
PBX: License accounting did not work with Unknown Registrations under some special conditions
| Status | Closed |
| Id | 67731 |
It could happen that a registration to a user was not accounted for if the endpoint used for this already had an unknown registration at the time the user was created
User Interface: Alignment of tables fixed
| Status | Closed |
| Id | 67734 |
Some strange alignments were introduced with the last hotfix
Gateway: Interface Name with multiple "Umlaute" did not work
| Status | Closed |
| Id | 67737 |
There was a length limitation of the URL encoded output, which was already exceeded if three "Umlaute" (or any character which is encoded in more the one byte with utf-8), were used
IP-DECT: Packetization could change after handover
| Status | Closed |
| Id | 67738 |
On the new radio the RTP should be sent using the same packetization as was negotiated with the original call
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.
Leak checking improved
| Status | Closed |
| Id | 67783 |
sometimes leaks were falsely detected. Problem if objects are about to be deleted, which were not owned by any module anymore. This happend esspecially with httpclient.
SIP: Generate/add SRTP key on media-relay interfaces
| Status | Closed |
| Id | 67789 |
This enables even partial SRTP (SRTP on one side of media-relay).
Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set
| Status | Closed |
| Id | 67791 |
Description: Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set. Activating a registration is a state change, and not a configuration modification, so allow this option.
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.
ip22/24/28/302/305: DSP affected by certain fax tones
| Status | Closed |
| Id | 67821 |
Brother fax problem, the first fax is transfered, the next fax transfers fail.
Switch to fax from remote is now done without reopening the channel.
Closing the channel waits until t38 is switched off.
Status:
ac_dsp3.cpp ac_dsp3.h
IP30x, IP1060, IP2010, IP6010: Fax did not work if rerouted from ISDN interface to a Voip destination
| Status | Closed |
| Id | 67823 |
DSP was not configured correctly
IP2x2: Handset microphone gain increased
| Status | Closed |
| Id | 67843 |
changed from 22.5db to 30db Status: ac_codec3.cpp
IP241: Remote party information truncated more than necessary
| Status | Closed |
| Id | 67872 |
Line 3 of call ctrl
IP-DECT: Changed channel option SRTP crypto suite
| Status | Closed |
| Id | 67904 |
Now DECT system channel configuration option 'Secure RTP' is a drop down box. The DECT Master correctly transmits the changed option. This feature was changed in V9 Hotfix 2, related case #66810.
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.
myPBX: Always send an ID with CT_INITIATE
| Status | Closed |
| Id | 67932 |
This is needed for interoperability with phones using v9hotfix2 or older. (See #67803)
SIP: Must answer every request - even unknown/unsupported methods
| Status | Closed |
| Id | 67935 |
Lync sends proprietary NEGOTIATE request and waits for response.
Otherwise transport connection is blocked for any upcoming request.
PBX Waiting: A call rejected on a primary operator was never sent to all operators
| Status | Closed |
| Id | 67945 |
if again and again rejected on all primary operators. Problem happend also if rejected by "Do not disturb" on operator phone.
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.
SIP: Generate new SRTP key on every incoming re-negotiation
| Status | Closed |
| Id | 67958 |
Required to have different encryption after transfer.
PBX-CDR: Mobility calls to Trunk, external number was missing in CDR
| Status | Closed |
| Id | 67978 |
only the number of the trunk itself was available
IP150: OEM specific WEB GUI modifications did not work
| Status | Closed |
| Id | 67992 |
manufacturer specific stylesheets have to be be adjusted to the new GUI style
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
PBX: User Interface Inconsistency. Same thing was sometimes call 'Response Timeout'/' and 'No Response Time'
| Status | Closed |
| Id | 68008 |
Only Response Timeout used anymore
PBX-SOAP: Support UserClear for pending outgoing calls on Waiting Queues
| Status | Closed |
| Id | 68009 |
UserClear was ignored in this state
SIP: Fix for media negotiation with SRTP
| Status | Closed |
| Id | 68067 |
Exchange of SRTP key may fail after hold/retrieve
phone_orchid: DTMF Tones detected in voice data from microphone were propagated to remote side
| Status | Closed |
| Id | 68075 |
DTMF tones may be detected from audible feedback on pressing a dial key in connected state and also from some other source. It's better to propagate only tones requested explicitely via a dial key and not from some external source.
SIP: Trap handling 491 response on reliable transport
| Status | Closed |
| Id | 68093 |
Trap with MAX_BUSY_TICKS
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: Trap when outgoing SIP call is cancelled while DNS resolving is ongoing
| Status | Closed |
| Id | 68099 |
Trap when outgoing SIP is cancelled while DNS resolving is ongoing
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: 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.
IP241: Fix for display rendering
| Status | Closed |
| Id | 68181 |
Display of diversion destination was corrupt.
Pickup fkey labeling was wrong when party information needed truncation.
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.
SIP: Interoperability with Lync
| Status | Closed |
| Id | 68232 |
Media negotiation problem on calls coming from on Lync client
and getting forwarded to another Lync client.
IP241: Updated some display text information
| Status | Closed |
| Id | 68234 |
No need to strongly abbreviate disconnect cause text.
Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number
| Status | Closed |
| Id | 68240 |
Description: Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number
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.
Phone: Switch presence fkey light on if presence activity is set
| Status | Closed |
| Id | 68256 |
Switch presence fkey light on if presence activity is set
myPBX: Wrong charset was used when authenticating against reporting or changing user password
| Status | Closed |
| Id | 68258 |
Javascript uses Unicode. The PBX and the reporting use UTF8. Therefore strings have to be converted to UTF8 before encrypting them using RC4.
IP241: New fkey backgrounds
| Status | Closed |
| Id | 68304 |
Less transparency to avoid interference with personalized background images
myPBX: Chunked response from application sharing provider did not work
| Status | Closed |
| Id | 68311 |
After receiving HTTPCLIENT_RECV_RESULT, pbx_client did not send another HTTPCLIENT_RECV.
phone_orchid: call pickup via partner/pickup key fails when the key is pressed immediately after audible signal
| Status | Closed |
| Id | 68313 |
When "Audible Signal after alerting" was configured on a partner/pickup key and the key was pressed while or a short time after the audible signal was played then the picked call was mute. Status: files: ac_dsp3.cpp
| Status | Closed |
| Id | 68351 |
External background image source can be configured on web ui.
Background image can be selected on phone menu.
Now also external background image can be selected.
myPBX: Hide passwords for application sharing and reporting in config
| Status | Closed |
| Id | 68352 |
The passwords are moved from the config line to the VARS.
PBX0/MY-A - Application sharing password
PBX0/MY-R - Reporting password
SIP: Failed to register on dynamic PBX
| Status | Closed |
| Id | 68362 |
Attempt to register on a dyn PBX with SIP protocol was rejected with "301 Moved Permanently".
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.
SIP: Pending control calls on gateway
| Status | Closed |
| Id | 68378 |
Incoming unsolicited NOTIFY(message-summary) may cause pending control call on Gateway.
Control calls are calls (signaling connections) without media channel.
These calls are now released.
WEB GUI page cannot be scrolled completely when height of left hand logo is too big
| Status | Closed |
| Id | 68382 |
Height of FHF logo is bigger than that of the default logo, this must be considered when computing size for iframe below tab lines.
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.
IP222: Alpha input using the num block
| Status | Closed |
| Id | 68398 |
Automatic switching back to numeric mode when leaving input ctrl.
SIP: Trap when using TLS as transport
| Status | Closed |
| Id | 68410 |
Only if remote side closes transport connection while requests are pending.
phone_orchid: begin of voice mail prompt was cut off sometimes
| Status | Closed |
| Id | 68416 |
Connection of Voip to DSP channel was delayed and thus the first packets of the RTP stream were lost.
body onload attribute can be extended and tab_active method doesn't crash anymore
| Status | Closed |
| Id | 68430 |
Custom PBX object XSL had no method anymore to set the onload attribute of the body. This can be now extended with a XSL template parameter.
Additionally the tab_active method has been called by default and the default value caused the method to crash.
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.
IP241 - handsfree speaker volume to low
| Status | Closed |
| Id | 68451 |
The handsfree speaker volume was too low even when configured to maximum. Now the general output volume is increased by 3 dB. In case of problems the general output volume can be changed by
config add AC-DSP0 RINGER /VoiceOutputGain n
with n = 1..63 -> (-32 + n)db, n = 32 -> 0dB, n = 0 -> mute
wrong link to PPP Interface State Info Help
| Status | Closed |
| Id | 68463 |
Link out of V9 namespace
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.
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: Use secondary proxy address on interfaces without registration
| Status | Closed |
| Id | 68635 |
When secondary proxy address resolved by DNS it should be used to send INVITE to.
SIP: Handling of 491 response on reliable transport
| Status | Closed |
| Id | 68652 |
Re-try of re-INVITE was missing.
SIP: Treat domain part of SIP URI case-insensitive
| Status | Closed |
| Id | 68653 |
According to RFC3261(19.1.4)
SIP: Fix for REFER handling
| Status | Closed |
| Id | 68663 |
Support for attended transfer between two calls from/to different SIP endpoints.
As long as both calls belong to the same signaling interface.
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.
SIP: Web UI for cause code mapping (fix)
| Status | Closed |
| Id | 68772 |
Did not work as expected
SIP: Out-Of-Memory trap if Group Indications are enabled
| Status | Closed |
| Id | 68804 |
If 'Group Indications' are enabled on a PBX object where a SIP client registers,
an OOM trap may occur in case of heavy call activity on PBX.
Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed
| Status | Closed |
| Id | 68820 |
Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed
Phone: DnD Absence Message containing newline dumped to config-file breaks phone upon config-file upload
| Status | Closed |
| Id | 68898 |
Solution: All flashdir entries cotaining newline now dumped as binary.
IP30x V9 hf2 media problem: Echo canceller not reliable
| Status | Closed |
| Id | 68901 |
Sometimes voice from ISDN/Analog to the IP is muted after some time.
Seems to be a problem in the latetest echocanceller.
Change to old DSP code until fixed DSP code is available.
Status:
ip24.mak
Denial of Service filter in ethernet library did not work
| Status | Closed |
| Id | 68907 |
this filter can be useful to prevent DOS attacks on non routing devices
SIP: Trap when handling SUBSCRIBE on federation interfaces
| Status | Closed |
| Id | 68976 |
SIP client object may be deleted while DNS query is pending.
But only under critical timing conditions.
myPBX: Use display name for sending meeting urls and conference numbers
| Status | Closed |
| Id | 68992 |
For normal chat messages the DN is used as the sender name. For sending the links for application sharing or audio conferences the CN was used.
Phone: Hide calling party on Pickup key fixed
| Status | Closed |
| Id | 68995 |
Phone: Hide calling party on Pickup key fixed
SoftwarePhone: No CT setup with remote connected calls
| Status | Closed |
| Id | 68996 |
The call transfer setup facility is removed in the call setup if the call is a by remote control connected call used in case of outgoing calls with myPBX. This fixes an empty diverting party number information element in the PBX. Now it can be used with a trunk PBX object with the enabled option 'Set Calling=Diverting No', otherwise the calling party number was removed within this object.
PBX: Conference trap
| Status | Closed |
| Id | 68998 |
A zero pointer trap in the broadcast conference PBX object is fixed.
IP222/IP232: Hookswitch on some PCB not working
| Status | Closed |
| Id | 69001 |
Reduce LED frequency from 12kHz to 6 kHz Status: orchid_lcd.cpp
phone_orchid - builtin speaker test did not work
| Status | Closed |
| Id | 69026 |
works only over channel 0
myPBX: Login did not work with system names containing special characters
| Status | Closed |
| Id | 69033 |
Missing URL encoding when sending the connect-request message.
PBX Mobility: Mobility object hanging if call establishment with SOAP/myPBX canceled
| Status | Closed |
| Id | 69052 |
If a call for a mobile phone is initiated by SOAP or myPBX, a call is first sent to the mobile phone. If the mobile phone accepts the outgoing call to the destination is initiated. If the mobile phone did not accept the initial call, no other calls could be done from then on.
IP-DECT: Configuration of Media preferences did not work anymore
| Status | Closed |
| Id | 69056 |
Media preference were ignored. This was a collateral damage introduced with fixes from the last hotfix
V9 Hotfix 4 (90600.04)
Changes included in Version 9 hotfix4 Definition
New Features
Improved protection against Denial of Service attacks
| Status | Closed |
| Id | 69166 |
flooding a box with different kinds of packets may lead to out of memory conditions. The Denial of Service filter in the ethernet layer is activated where required. TCP listening sockets have a backlog limit now. The http service restricts the number of half-open sessions and limits the number of concurrent sessions according to the total memory available on a box.
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
PBX Mobility: Support of transfer on mobile side, not using mobility
| Status | Closed |
| Id | 69275 |
In case that the mobile phone transfers the call to another destination, this call must be removed from the mobility function, so that the mobility function is available for another call
USB Driver merge from v10 to v9
| Status | Closed |
| Id | 69288 |
First step to support USB headsets. This is not functional yet.
SIP: New config file option /hold-notify-as-inactive
| Status | Closed |
| Id | 69293 |
If set, holdNotific is interworked into "a=inactive".
If not, holdNotific is interworked into "a=sendonly".
SIP: New config file option /prefer-pai2
| Status | Closed |
| Id | 69459 |
Interoperability with Telepo:
When receiving INVITE, get calling party id from second P-Asserted-Id header.
new: DHCP manufacturer specific option 'boot-cfg' provides an URL to read config from after any reset except 'creset'
| Status | Closed |
| Id | 69472 |
This option permits to boot boxes with a fresh config provided via TFTP/HTTP without storing the config on the device.
It is intended to be used as follows:
1. the box is started with DHCP enabled (no initial configuration)
2. the box contacts the DHCP server and gets the ip-address and also the Vendor Specific Information in option 43.
Suboption 249 of the Vendor Specific Information specifies the URL of the boot config file.
3. the box polls the TFTP/HTTP server for the config file.
4. the box reads the config file and executes the commands provided in the file
The URL may contain the same meta-character strings an Update Server URL, for example #m (mac-address)
The length of the URL in the DCHCP suboption is restricted to 127 characters.
The URL is polled in 5 second intervals.
The config file is read and executed by the update process in the usual way.
A 'creset' commmand as last command of the file will restart the box with the new configuration without writing any 'config' command options to the flash.
After a restart by the 'creset' commmand the boot-cfg URL is ignored.
After a restart by any other of the 'reset' commands or by a power cycle the boot-cfg URL is processed again.
On an Innovaphone DHCP-Server configuration of a boot-cfg URL and providing it to clients via suboption 249 must be explicitely enabled by
config add DHCP0 /boot-cfg
config write
config activate
Once enabled the URL may be entered under "IP4/ETH0/DHCP-Server/Boot Config URL" and is provided to all clients then.
If an Innovaphone DHCP-Client receivess a boot-cfg URL it is displayed under "IP4/ETH0/DHCP/Boot Config URL".
auto complete dtmf feature codes with '#' after 2 seconds
| Status | Closed |
| Id | 69561 |
Optional feature for phones, which are not able to send a '#', e.g. the iPhone.
They dial a feature like a cfu with a destination number and after two seconds, the feature code is automatically completed with a '#'.
PBX: HTTP request to initiate call for mobile phone
| Status | Closed |
| Id | 69570 |
To improve GSM client functionality
PBX: WSDL for SOAP API readable from box
| Status | Closed |
| Id | 69688 |
Simplifies Application development in some environments
SIP: Get display information from Call-Info header in register refresh response
| Status | Closed |
| Id | 69758 |
Get display information from Call-Info header in 200/OK
IP-DECT: Display update
| Status | Closed |
| Id | 69770 |
Now, the endpoint's display is updated if the registration forwards a display update.
Voicemail <pbx-query-obj> queries new properties
| Status | Closed |
| Id | 69781 |
"user_type": returns "1"(endpoint) or "2"(other object like gateway)
"pseudo_type": returns types alike "vm", "bc_conf", "trunk"
"pseudo_text": returns friendly texts alike "Voicemail", "Bc Conference", "Trunk Line"
Bug Fixes
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.
RTP-DTMF: Start handling of RTP-DTMF on reception of END event
| Status | Closed |
| Id | 68938 |
Workaround for Bug in MS Lync / Mediation server.
Mediation server changes destination port while sending RTP-DTMF redundancy retransmissions.
SIP: Presence interoperability with ESTOS UC server
| Status | Closed |
| Id | 69050 |
'Do Not Disturb' signaling without presence/tuple/contact element.
SIP: Support for Mediation Server Cluster
| Status | Closed |
| Id | 69051 |
Load balancing and fail-over acc. to spec.
Memory leak with each new telnet session
| Status | Closed |
| Id | 69157 |
Memory leak with each new telnet session (last packet upon exit not cleaned up)
SIP: Media negotiation problem
| Status | Closed |
| Id | 69159 |
On media-relay with exclusive codec we can answer an incoming SDP offer right away without passing to app.
PPP connection fails after LCP renegotiation with different authentication methods (PAP -> CHAP)
| Status | Closed |
| Id | 69167 |
the active authentication entity was not stopped when another authentication was started and and signalled layer down after some timeout
PBX Twinning: Blind transfer to other phone in a twin phone configuration was not possible
| Status | Closed |
| Id | 69170 |
Only normal call to other phone or consulation call was supported
Web-UI: Fixed layout of DynPBX configuration
| Status | Closed |
| Id | 69197 |
Use min-width for fieldset.left
IPVA didn't run on vSphere5
| Status | Closed |
| Id | 69232 |
IPVA didn't run on vSphere5
SIP: Interoperability of INFO(application/dtmf-relay) with Polycom
| Status | Closed |
| Id | 69247 |
Fail to decode DTMF signal, since "application/dtmf-relay" body does not contain any CRLF.
While CRLF is required according to "SIP INFO Package for DTMF".
IP222/IP232: residual echo in handset mode
| Status | Closed |
| Id | 69249 |
Change Codec config Status: ac_codec3.cpp
ip22/24/28/302/305: DSP affected by certain fax tones causing a trap
| Status | Closed |
| Id | 69250 |
Assert removed for testing
Status:
ac_dsp3.cpp
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
CX0-Wave-Encoding sometimes produced white noise
| Status | Closed |
| Id | 69286 |
CX0-Wave-Encoding sometimes produced white noise.
Also a memory leak was eliminated.
IP-DECT: Hanging call after call transfer
| Status | Closed |
| Id | 69309 |
The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer to an unassigned number and the transferred call is not disconnected.
SRTP: Bad key exchange on H.323/SIP interworking
| Status | Closed |
| Id | 69353 |
MKI was added in SDP due to uninitialized data object.
Pass DNS-TTL to SIP
| Status | Closed |
| Id | 69374 |
In order to do load-balancing SIP needs to ge hold on the TTL of DNS resource records.
SIP: Support of maddr parameter in redirect response for REGISTER
| Status | Closed |
| Id | 69386 |
Set when redirecting REGISTER.
Read when processing redirect response.
G726 codec obsolete
| Status | Closed |
| Id | 69388 |
The G726 codec was rarely used (if ever) in real life. In addition there are signaling problems specially with DECT peers when G726 is selected. Thus G726 is removed from the list of supported coders in all products.
IP222: Input ip addresses does not work
| Status | Closed |
| Id | 69393 |
Entering an ip address on the phone's menu does not work.
All digits are doubled.
IP222: Call reroute did not work
| Status | Closed |
| Id | 69455 |
Using redial key on in incoming not-connected call did not work.
PBX: Call Diversion/Forward to '-' did not always show expected result
| Status | Closed |
| Id | 69457 |
A call diversion to a destination'-' can be used to explicitly no execute a diversion of this type. So if a user has an CFU to '-' and this diversion is valid for a given call (Filter, Boolean), the phone should ring.
In fact the call was rejected.
There was also a problem with CFB in case of "busy on ... calls"
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: No CLIR on internal calls did not work for SOAP
| Status | Closed |
| Id | 69474 |
If the features "No CLIR on internal Calls" is activated on a PBX a CLI is sent to the called phone even if the call was sent with "CLI presentation restricted". The same should be case on SOAP/TAPI when monitoring this user.
Now when "No CLIR on internal Calls" is enabled all number information available is provided on SOAP.
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 Waiting: Not possible to send DTMF to Waiting Queue from myPBX Numeric Keyboard
| Status | Closed |
| Id | 69482 |
The Waiting Queue prohibited the facilities used for this to be sent to calling phone
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: In case of "Announcemen w/o Connect" together with Alert-Timeout 0, DTMF dialing was not possible
| Status | Closed |
| Id | 69496 |
In this case as only response to the incoming SETUP a PROGRESS was sent. This meant, that the caller was still in overlap dialing state, so a phone does not send DTMF, but translates input keys to INFO dialing messages.
A CALL-PROC is now sent before PROGRESS, which terminates the dialing.
SIP: Send BYE with Reason header with "Q.850 Recovery on timer expiry"
| Status | Closed |
| Id | 69500 |
If session refresh is outstanding the call is released with BYE with "Reason: Q.850;cause=16".
Better send BYE with "Reason: Q.850;cause=102"
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
SIP: Bug when decoding SRTP keys from SDP
| Status | Closed |
| Id | 69545 |
Bug in decode base64.
Web-UI: Input field to small for SIP-URI
| Status | Closed |
| Id | 69578 |
Input control on "UC" tab of External UC properties dialog was too small.
Only part of SIP-URI was visible.
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.
IP-DECT: Reset link
| Status | Closed |
| Id | 69584 |
The reset link of DECT System Config GUI page is fixed.
SIP: Do not check remote certificate name when calling PBX client with TLS
| Status | Closed |
| Id | 69598 |
Check is reqired only when calling an unregistered client or when giving INVITE to registrar.
| 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.
IP222: Alpha input using the num block
| Status | Closed |
| Id | 69646 |
Input mode changes back to numeric in screen "Wahlvorbereitung".
Should stay in alpha mode.
IP200: Manufacturing test fails is hwbuild is reprogrammed
| Status | Closed |
| Id | 69662 |
Old hwbuild was not cleared, because on INCA devices the hwbuild is located on a higher address was not cleared. Status: flash_firmware.cpp
H.323: Media Negotiation problem with Lync interop and SRTP
| Status | Closed |
| Id | 69687 |
With a retrieve from the lync after hold (which is signaled as a reinvite with sendrevc) new media parameters were sent, containing new SRTP keys. These new media parameters cannot be used, on the PBX which is initiating new end to end media negotiation at the same time. These media parameter were not ignored properly.
phone_orchid: wrong volume setting when monitor mode is entered
| Status | Closed |
| Id | 69734 |
when monitor mode ise entered by pressing the speaker key in a handset conversation the handsfree speaker is enabled in addition to the handset speaker.
the volume was reconfigured with the wrong value.
phone_orchid: Calls received with CLIR appear in call list with an empty entry
| Status | Closed |
| Id | 69737 |
Calls received with CLIR or without a number/name appeared in call list with an empty entry; now either "anonymous" (CLIR) or "unknown" is displayed instead of a name
SIP: Take SDP id and version as 64bit integer
| Status | Closed |
| Id | 69738 |
o-line of SDP offer/answer is defined as 64bit integer
Interoperability with MX-ONE
Relay: Forward facilities to local destinations
| Status | Closed |
| Id | 69764 |
This fix is related to the previous fix #66629 for V9 hotfix2. Now, facilities are only forwarded, if the destination is a physical interface, not e.g. a SIP provider.
SIP: Trap when handling NOTIFY(application/qsig)
| Status | Closed |
| Id | 69771 |
Traps if no progress indicator present in tunneled DISCONNECT message.
WEB-UI Config Upload screen blank after upload of a file with a lot of failing lines
| Status | Closed |
| Id | 69812 |
When uploading a file with about 50 failing lines the screen was left blank without any info about the failing operation.
phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences did not work with very long numbers/names
| Status | Closed |
| Id | 69816 |
speciall when using this option in conjunction with the "Display Name on Pickup/Partner Key" option the calling party info was not correctly displayed
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.
IP-DECT: Location update (OEM)
| Status | Closed |
| Id | 69863 |
Location update with message waiting information is fixed if the endpoint roams. This is only used by OEM devices.
Logging: Threshold for error event "SRTP authentication failed"
| Status | Closed |
| Id | 69920 |
Error event was triggered at the very first decrypt failure.
Some decrypt failure are expected during media re-negotiation.
Trigger this error event after 10 decrypt failures in line.
PBX Calls Page/SOAP wrong number
| Status | Closed |
| Id | 69921 |
In a configuration with escapes for calls from a slave and a node not the root node and the call forwarded to the master, because the number could not be resolved locally, wrong escapes were added to the called number
phone ip222: MWI LED not working
| Status | Closed |
| Id | 69924 |
The MWI LED was not switched on for pending voice mails and/or pickups
Trap during fax transfer
| Status | Closed |
| Id | 69950 |
Seen on Ip28, can happen on AC_DSP3 ( IP22/24/28/302/305/1060/301/6010 ) Status: ac_dsp3.h
phone_orchid: checkmark "Phone/User-x/General/Options/No DTMF Detection" has no effect
| Status | Closed |
| Id | 70114 |
If this checkmark is set DTMF digits entered via keyboard in a connected call shall be sent in-band as voice data, not encoded in RTP-DTMF packets as usual.
V9 Hotfix 5 (90600.05)
Changes included in Version 9 hotfix5 Definition
New Features
IP-DECT: Default frame size to 30ms
| Status | Closed |
| Id | 70140 |
Now the default frame size is 30ms.
SIP: Support for transparent message headers on transfer
| Status | Closed |
| Id | 70163 |
Interoperability with Telepo.
added support for USB tracing with Wireshark
| Status | Closed |
| Id | 70373 |
USB traces are now supported by the debug class and can be opened with Wireshark and a new innovaphone.dll.
SIP: Prefer P-Asserted-Identity with tel-URI
| Status | Closed |
| Id | 70417 |
Prefer P-Asserted-Identity with tel-URI over that with sip-URI.
In case there are two P-Asserted-Identity headers in INVITE.
Gateway: Forward Display Info received from ISDN Setup to H.323
| Status | Closed |
| Id | 70562 |
needed for compatibility with SecuGATE LI30
<pbx-getcallinfo> returning diversion reason
| Status | Closed |
| Id | 70697 |
as
<pbx-getcallinfo out-leg2-reason="..."/>
myPBX: Interface for IM provider
| Status | Closed |
| Id | 70867 |
Interface between the JavaScript client and the myPBX launcher.
Status:
IM to JS:
prepare_dial_name(value)
prepare_im(value)
ep_request(name,number)
JS to IM:
innovaphone_updateXml(xml)
innovaphone_sessionInfo(domain,name,number,cn,dn)
SIP: Diagnostics of transport error "SIP Overload"
| Status | Closed |
| Id | 70883 |
Limitation of buffer allocation
Status:
siptrans.cpp/h
sipmsg.h
IP-DECT: License restriction for OEM device
| Status | Closed |
| Id | 70887 |
A license restriction for an OEM device is added. It is not used in the IP1200.
SIP: Diagnostics of error "Registration expired"
| Status | Closed |
| Id | 70938 |
Missing registration refresh
Status:
sip.cpp/h
Possibility to clear the call list in mypbx
| Status | Closed |
| Id | 71202 |
A new button is now available in mypbx to clear the call list.
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
myPBX: Show implicit visibility in groups
| Status | Closed |
| Id | 71250 |
Active members of groups can see the presence and the calls of other group members. In order to make that clear to the user, now the visibility settings of myPBX show in what groups the user is visible.
IP-DECT: CSS changes for new OEM device
| Status | Closed |
| Id | 71282 |
For a new OEM device with a changed style some CSS classes are added.
PBX: New User property 'Do not Disturb'
| Status | Closed |
| Id | 71439 |
No calls are sent to the user if set.
Can be set by 'External UC' applications (e.g. Estos Procall)
PBX: Support up to six devices for a user
| Status | Closed |
| Id | 71506 |
Sometimes 4 devices are to little.
A little bit of cleanup on the user interface was done as well with this change
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
SoftwarePhone: New Tray Icon
| Status | Closed |
| Id | 71969 |
The SoftwarePhone has got a new tray icon differs from the myPbx one, now.
Bug Fixes
SIP: Message decoding error
| Status | Closed |
| Id | 70116 |
Only concerns Message headers whose value starts and with quotes, but are not quoted.
E.g.
Referred-By: "Huvudnummer"<sip:400@abcdef.ghi;fnrid=1759>;from-tag=5decdf1a;to-tag=2515833546;org-cid="6afa95ede909d311906f00013e11cdb3@192.168.2.115"
Linux: VLAN GUI removed
| Status | Closed |
| Id | 70131 |
It is not necessary to configure a VLAN to Linux. Now the GUI page Linux VLAN is removed.
ip24/ ip6010/phone_orchid: in-band ring back tone generation into voice channel did not work
| Status | Closed |
| Id | 70146 |
sometimes a ring back tone must be sent in-band to an ISDN channel. the tone was generated but passed to the wrong destination.
SIP: Problems with CLEARMODE
| Status | Closed |
| Id | 70149 |
CLEARMODE was not offered as prefered codec outgoing INVITE.
Destination accepted G711a instead of CLEARMODE:
Offer:
v=0
o=- 18 1 IN IP4 172.16.66.77
s=-
c=IN IP4 172.16.66.77
t=0 0
m=audio 16422 RTP/AVP 4 18 8 0 96 97 101 13
a=rtpmap:96 G726-32/8000
a=rtpmap:97 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexa=yes
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
Answer:
v=0
o=cp10 131833822144 131833822144 IN IP4 172.16.66.175
s=SIP Call
c=IN IP4 172.16.67.132
t=0 0
m=audio 33526 RTP/AVP 8 0 97
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:97 CLEARMODE/8000/1
a=ptime:30
IP-DECT: Master radio list sync highlighting
| Status | Closed |
| Id | 70150 |
In the IP-DECT Master radio list the active sync highlighting is fixed, now.
SIP: No update of name presentation after transfer on QSIG interworking
| Status | Closed |
| Id | 70155 |
Fix for QSIG/SIP interworking.
Display name of callTransferComplete was not passed to SIP.
IP222/IP232: Handset volume with 90772 andHF4 too low
| Status | Closed |
| Id | 70226 |
This is a side effect of the echo canceller setting, volume in IP222/232 increased. IP241 uses old settings.
SIP: Cannot use SIP phone with myPBX
| Status | Closed |
| Id | 70252 |
Blind transfer (sending REFER) did not work since v9hotfix3.
| Status | Closed |
| Id | 70263 |
tftp was activated with alt-key, on on IP240 menu-key is used
Status:
platform_orchid.c
boot241.y
IP222/IP232: Sporadic Problem with the LCD Display after soft-reset
| Status | Closed |
| Id | 70269 |
It seem to be problematic to reset all orchid modules, e.g. the DMA module during software reset.
Now only USB and ENET modules are reset, the display gets also reset.
The display reset is released in the firmware.
Status:
start_orchid.S
platform_orchid.c
phone_orchid.cpp
boot222.y
boot232.y
boot241.y
IP241: Missing fkey icons
| Status | Closed |
| Id | 70353 |
Icons types list, face, mask-white and mask-black are available now.
PBX: Conference trap
| Status | Closed |
| Id | 70362 |
Some reworks of the PBX conference object. Fixes traps with call transfers of conference calls and conference calls to other PBX objects or mobility. Object update is also possible without call and chat clearing, now. Set maximum call number takes effect for maximum incoming calls, now.
IP241: Status icon for conditional call diversion
| Status | Closed |
| Id | 70363 |
Now having 2 different status icons for diversion.
One for unconditional and one for conditional forwarding.
IP241: Moving focus on screen "Call Diversion" activates diversion
| Status | Closed |
| Id | 70377 |
Using UP/DOWN/LEFT/RIGHT keys on the "Call Diversion" screen makes ON/OFF changing it state.
Webmedia: Recording of G.722 did not work
| Status | Closed |
| Id | 70382 |
Empty file was created.
myPBX: Closing a chat window sometimes causes a Java Script error
| Status | Closed |
| Id | 70439 |
client.js, line 2033
The Problem is a collision of closing the window and incoming messages for that window.
SIP: DNS resolving _stun._udp.xten.com did not work
| Status | Closed |
| Id | 70449 |
Because of unusual DNS response.
Target attribute of SRV answer records did not contain host domain name, but ip address.
<result val="0" title="SUCCESS">
<answer_rrs title="Answer Records">
<rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.16"/>
<rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.14"/>
</answer_rrs>
<ns_rrs title="Authoritative Nameserver Records">
</ns_rrs>
<ar_rrs title="Additional Records">
</ar_rrs>
</result>
H.323: Renegotiation to Fax did not work under some conditions
| Status | Closed |
| Id | 70462 |
Problem happened
- If switch to fax was done right after connect. This is typically done by IP Fax Servers
- If multiple signaling hops (e.g. multiple PBXs) were used
- If connect to a tone interface happened during dialing
myPBX: Norwegian translation updated
| Status | Closed |
| Id | 70477 |
Sent => Send
Logginn => Pålogging
Permanent logginn => Forbli pålogget
ethernet broadcasts bearing unicast IP packets with an arbitrary destination address were passed to the local IP-stack
| Status | Closed |
| Id | 70503 |
some load-balancing implementations send unicast IP packets (specially TCP-SYN) as ethernet broadcast packets.
Such packets must be silently discarded if the IP destination address is not the address of one of the local interfaces.
| Status | Closed |
| Id | 70569 |
ESC was taken as BS (backspace).
ESC must be handled as ESC when BS is no longer possible (empty input).
PBX: Presence update for PBX objects
| Status | Closed |
| Id | 70577 |
Now the presence is updated in myPBX for other PBX objects if the access is changed to allowed.
PBX: Conference presence info
| Status | Closed |
| Id | 70581 |
Now the broadcast conference object updates the presence info and is shown as callable in myPBX.
myPBX launcher: Hotkey does not work with some applications
| Status | Closed |
| Id | 70623 |
Try to get the selected phone number using WM_COPY before sending CTRL-C. This should help in some of the cases where the hotkey didn't work before.
SIP: Do not try to map local listen port with STUN when SIP/TCP is used
| Status | Closed |
| Id | 70654 |
Mapping of local listen port is only required fur SIP/UDP
IP222: Use R-key as BACKSPACE on edit fields
| Status | Closed |
| Id | 70664 |
Not ESC-key.
ESC-key is for leaving screen only.
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.
IP-DECT: Hanging call after failed call transfer
| Status | Closed |
| Id | 70756 |
The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer and the call transfer fails.
SIP: Allow STUN to be used to map local media ports on SIP/TCP interfaces
| Status | Closed |
| Id | 70809 |
STUN cannot map signaling TCP port, but UDP media ports.
Status:
medialib.h
media.cpp
h323ch.h/cpp
sip.h/cpp
siptrans.h/cpp
IPVA, Keyboard Console, Credentials With Special Chars Couldn't Be Entered
| Status | Closed |
| Id | 70873 |
The Scancode table wasn't finished
SIP: Reduce memory footprint of SIP stack
| Status | Closed |
| Id | 70886 |
Free INVITE request buffer when receiving ACK. No need to keep any longer.
Delete INVITE client transaction when cancelling.
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
IP6010: Wrong timer under high load
| Status | Closed |
| Id | 71001 |
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQ´s a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs.
-The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled.
Status:
ip6010.cpp
ip6010.h
SIP: Media negotiation problem
| Status | Closed |
| Id | 71009 |
Internal re-negotiation during early media on incoming SIP call.
Provide received offer to app again.
SIP: Offer CLEARMODE only if bearer capabilities are "Unrestricted Digital Information"
| Status | Closed |
| Id | 71162 |
On "Unrestricted Digital Information" only CLEARMODE is offered (no audio codecs).
On other bearer capabilities no CLEARMOE is offered (only audio codecs).
phone: dialog and presence subscriptions sometimes got lost after PBX restart when phone config was stored on PBX
| Status | Closed |
| Id | 71198 |
This happened specially when both "Store Phone Config" and "Discard Config on Phone" was checked in the user object because of a unsubscribe/subscribe race condition.\t
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: Endpoints behind NAT could not register at public PBX
| Status | Closed |
| Id | 71266 |
PBX must send response to the (public) IP address where request was received from.
Not to the (private) IP address in Via header or Contact-URI.
ipva, update to _platform_tracing.xsl
| Status | Closed |
| Id | 71274 |
ip6 missing
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.
CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header
| Status | Closed |
| Id | 71290 |
CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header
DHCP Server Identifier could not be cleared via WEB interface
| Status | Closed |
| Id | 71305 |
When the field 'Server Identifier' was cleared and OK was pressed the just cleared value reappeared but after a reset 'Server Identifier' was clear.
IP241: Show ISDN display information in full length
| Status | Closed |
| Id | 71330 |
Use scrolling if required
NAT: Mapping to different internal UDP port did not work
| Status | Closed |
| Id | 71339 |
Configured destination port got lost after configuration
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: Send first NOTIFY(dialog) after sending 200/OK for SUBSCRIBE(dialog)
| Status | Closed |
| Id | 71413 |
NOTIFY(dialog) was sent before 200/OK for SUBSCRIBE(dialog)
\tRequest: SUBSCRIBE sip:150@192.168.200.14
\tStatus: 401 Unauthorized
\tRequest: SUBSCRIBE sip:150@192.168.200.14
\tStatus: 100 Trying
\tRequest: NOTIFY sip:158@192.168.200.214:5060
\tStatus: 200 OK
\tStatus: 481 Subscription Does Not Exist
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.
H.323 Signaling Timeout event was generated for 'normal' RAS re-transmissions
| Status | Closed |
| Id | 71434 |
A single retransmission is normal under heavy load, so this is no reason for an event. Signaling Timeout events are now generated only if they cause a state change.
H.323 re-negotiation: Don't reuse media proposals if a select was already sent
| Status | Closed |
| Id | 71435 |
This fixes compatibility issues with SIP, especially when SRTP is used.
IP241: Show both parties of each call on screen "Pickup List"
| Status | Closed |
| Id | 71455 |
Currently only calling party infomation is displayed.
| Status | Closed |
| Id | 71469 |
The function assigned to the menu item (for example pickup) was executed first and then the key-function (for example dial).
PBX: Wrong web page when submitting an object and an error happens
| Status | Closed |
| Id | 71470 |
If for example a dupicate number is detected, the same web page should be displayed including the error message for the duplicate number. But not the same page was displayed but a page which could contain information not related to the object.
v8 to v9 upgrade problem with gateway registration names containing non-ASCII characters
| Status | Closed |
| Id | 71474 |
In general this was a problem with config line arguments seperated by ':'. This happened with the <number>:<name> argument within gateway definitions. The ':' was url-encoded and <name> interpreted as <number>
PBX: CFU was executed on PRESENCE_PUBLISH/SUBSCRIBE calls
| Status | Closed |
| Id | 71479 |
This was unexpected behaviour. You want to see the presence status of the configured user and not the presence status of the destination to which this user has configured a call forwarding
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.
Phone: Presence-Fkey did not always show presence set by myPBX
| Status | Closed |
| Id | 71487 |
Problem:
1) Set presence A with IP phone (fkey shows A)
2) Set presence B with myPBX (fkey shows B)
3) Delete presence with IP phone (fkey shows no presence)
Now Fkey shows presence B.
PBX: Tooltip on "PBX/Config/Log Calls" checkmark wrong
| Status | Closed |
| Id | 71507 |
It read "If not checked PBX calls are logged", should be "If checked PBX calls are logged"
| Status | Closed |
| Id | 71513 |
See http://wiki.innovaphone.com/index.php?title=Reference9:Concept_Fine_grained_function_hiding#More_Information
for more information
Media: Discard RTP packets from wrong source
| Status | Closed |
| Id | 71515 |
Packets arriving at RTP port must be discarded if the source if not the expected one.
To be save against DOS attack and for interop with Lync.
In some scenarios Lync starts sending RTP packets while having the call set to 'inactive'.
PBX: Blind transfer with consultation to BC-Conference failed
| Status | Closed |
| Id | 71540 |
The call was disconnected
H.323: No Alarm/Event should be generated by shutting down registration due to reset
| Status | Closed |
| Id | 71545 |
no event is sent to the application about this kind of unregistration
phone-orchid: micro-speaker loop of embedded test did not work
| Status | Closed |
| Id | 71566 |
Der Test war im Treiber nicht freigeschaltet
myPBX: Send existing invitation links to users that join the chat session later
| Status | Closed |
| Id | 71608 |
Users A and B have a chat session.
1. A starts application sharing or audio conference
-> A and B receive an invitation link
2. A adds a third user C to the chat and C accepts
-> C should also receive the invitation link
This worked only for audio conference but not for application sharing.
IP-DECT: Ring back tone after transfer
| Status | Closed |
| Id | 71610 |
If in-band ring back tone is sent and the call is transferred to a new destination with no in-band ring back tone, a local ring back tone must be played to the DECT handset. This is fixed now.
Ring Back tone missing after transfer when in-band tone was provided before but not after transfer
| Status | Closed |
| Id | 71612 |
The initial local ring back tone must be restarted when no data is received after transfer.
PBX: Send Name Identification with CLIR calls if "No CLIR on internal Calls"
| Status | Closed |
| Id | 71643 |
The feature "No CLIR on internal Calls" did not work completetly. The number was sent, but the Name Id was still suppressed
phone_orchid: Ring Back tone missing when Silence Compression is enabled
| Status | Closed |
| Id | 71670 |
When Silence Compression was enabled the tone generation was not triggered
IP232: Fix for touch handling
| Status | Closed |
| Id | 71674 |
Do not open touch keyboard on controls with CTRL_READONLY.
Do not open touch keyboard on controls without CTRL_ACTIVATE.
Using KEY_SHIFT has modified key to uppercase permanently.
Cursor positioning on text controls did not work.
Multi-line editor control was not displayed after hiding touch keyboard.
Hide overlay keyboard after next touched key.
Move and resize editor control when activating touch keyboard.
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
phone_orchid: continuous dialing tone stops after 10 seconds and is not restarted anymore
| Status | Closed |
| Id | 71753 |
the continuous dialing tone as used in most countries must be played as long as no digit is entered.
IP230/IP240: Sporadic traps during manufacturing programming
| Status | Closed |
| Id | 71778 |
Inefficient loop during mac address programming
Status:
flash_firmware.cpp
IPVA, Reset didn't work on VMware Player 4.0
| Status | Closed |
| Id | 71818 |
IPVA, Reset didn't work on VMware Player 4.0
IP241: Slow screen update when changing fkey type on fkey configuration screen
| Status | Closed |
| Id | 71831 |
Removing and adding config controls is very slow.
ip6010 - calling & dtmf tone timing did not work
| Status | Closed |
| Id | 71966 |
the tone time was calculated much too short
IP232: Redesign of touch keyboard
| Status | Closed |
| Id | 71968 |
Redesign of touch keyboard
IPVA, Trap After Failed FW Upload
| Status | Closed |
| Id | 71978 |
If a firmware upload failed with "wrong checksum" the box trapped occasionally, leaving the boot disk in an inconsistent state.
V9 Hotfix 6 (90600.06)
Changes included in Version 9 hotfix6 Definition
New Features
IP232: Presence fkey shows current activity, but not current note
| Status | Closed |
| Id | 71611 |
Show current note in second line of fkey label.
myPBX: Cause codes for instant messaging
| Status | Closed |
| Id | 71878 |
Display causes why a chat session has been terminated.
For example: Unassigned number, rejected, no response...
myPBX: Make call state icons clickable
| Status | Closed |
| Id | 71886 |
Clicking a call state icon should have the same effect as clicking the text.
myPBX: Notification on disconnected chat sessions
| Status | Closed |
| Id | 71918 |
Display a text that the chat has been terminated.
Debug information on assertion
| Status | Closed |
| Id | 71961 |
More debug information on default event handler.
myPBX: Display reason why a browser is not supported
| Status | Closed |
| Id | 71972 |
Currently it only says "Browser version not supported".
It should also tell what feature isn't supported:
- XmlHttpRequest
- WebStorage
- PostMessage
myPBX: Forward invitation links to incoming chat calls that are added to a chat conference
| Status | Closed |
| Id | 72280 |
The same way like the invitation links (for application sharing and audio conferences) are forwarded to additional outgoing chat calls.
myPBX: Display invitation links in the chat window of the organizer
| Status | Closed |
| Id | 72282 |
The organizer of an application sharing session or a audio conference wants to see all corresponding links in the chat window.
SoftwarePhone: Password encryption tool
| Status | Closed |
| Id | 72431 |
A password encryption tool was added.
SoftwarePhone: Crash dump
| Status | Closed |
| Id | 72433 |
Now, the SoftwarePhone writes a crash dump file.
myPBX: Display shortened links in chat messages
| Status | Closed |
| Id | 72445 |
Shorten the displayed part of the URL in order to avoid text overflow.
Cut overflowing texts at the right boundary of the chat box.
PBX: New presence access flag 'online'
| Status | Closed |
| Id | 72538 |
With this flag a user allows access to his online status
myPBX: Configure visibility of online state and presence separately
| Status | Closed |
| Id | 72632 |
Phone status and IM status are now called online status.
Activity and note are now called presence.
The visibility of both is now configured separately.
PBX Mobility: Better support for Opticaller
| Status | Closed |
| Id | 72701 |
HTTP request to initiate call for mobile phone
myPBX: New translations
| Status | Closed |
| Id | 72768 |
For the other v9hotfix6 items regarding myPBX.
Support for more automated Fax Tests
| Status | Closed |
| Id | 72826 |
Switching from Audio to Fax to Audio to Fax. Special case which could happen if switch to Fax happened before connect.
myPBX: Highlight incoming alerting calls
| Status | Closed |
| Id | 73093 |
Highlight the alerting icon using an orange border.
SIP: Interworking with KIRK Wireless Server 300
| Status | Closed |
| Id | 73110 |
If "KIRK Wireless Server 300 PCS10__ r3327"
calls into PBX and is connected with Voicemail,
Voicemail may send re-INVITE with SRTP key.
Instead of accepting or ignoring the SRTP option,
KIRK Wireless Server 300 rejects the whole SDP offer.
Now we retry the re-INVITE w/o offering SRTP key.
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.
SoftwarePhone: Support for Jabra SPEAK 410 USB with product id 0x0410
| Status | Closed |
| Id | 73424 |
Now, the SoftwarePhone supports Jabra SPEAK 410 USB with product id 0x0410.
SIP: Overriding cause code mappings did not work
| Status | Closed |
| Id | 73477 |
Mapping of SIP response codes into Q.931 cause codes could not be overwritten.
Linux: Empty file check
| Status | Closed |
| Id | 73554 |
Now, an empty file is not accepted as a successful file upload.
Bug Fixes
myPBX: Make more obvious that closing the chat window terminates the chat
| Status | Closed |
| Id | 71893 |
Display warning message when the user tries to close the chat window.
Show disconnect icon instead of close icon in the window.
IPVA, Optimize Flash-Related I/O Operations
| Status | Closed |
| Id | 71954 |
Specifically to reduce boot-time. Utilizing 64KB I/O accesses where possible.
H.323 RAS: Registration with authentication to account without authentication failed in a strange way
| Status | Closed |
| Id | 71958 |
Information was missing that no authentication was supported, so it was continued to retry instead of giving up right away
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: Workaround for delays in media negotiation
| Status | Closed |
| Id | 72189 |
Workaround for delays in media negotiation caused by delayed Admission on H.323 calls.
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
IP232: Input of capital Latin Extended-A
| Status | Closed |
| Id | 72226 |
Input of capital Latin Extended-A
myPBX: Queue chat messages for im calls that are not yet alerting
| Status | Closed |
| Id | 72237 |
The following fix did not work across PBXes:
#71608: myPBX: Send existing invitation links to users that join the chat session later
The problem was that messages were only queued in alerting state. When the call is made across PBXes this state isn't reached, yet.
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.
IP232: Touch keyboard key highlighting
| Status | Closed |
| Id | 72268 |
Added highlighting for key like SHIFT, BACKSPACE, etc.
PBX: H.323 Names in some places (e.g. Waiting Queue Maps) containing non-ASCII charcters could break the user interface
| Status | Closed |
| Id | 72277 |
This happened esspecially for Names as destinations for Waiting Queue maps, but it could happen at other places as well.
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.
IP241: Disable Link LED not working
| Status | Closed |
| Id | 72338 |
Timing problem with access to the paged MDIO registers Status: orchid_drv.cpp
Trap: When accessing web interface
| Status | Closed |
| Id | 72362 |
Seen once.
Not reproducable yet.
Ip6010 DSP Disconnect timeout after fax session
| Status | Closed |
| Id | 72403 |
Debugs added
* enable with http://addr/debug.xml DSP trace and DSP control message trace to printout all packets to the DSP with a descriptive string. That allows to analyse the message flow to the DSP after a trap.
* for further testing old fax disconnect procedure can be enabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl with "t38 skip fax close".
Status:
ac_dsp3.cpp
ac_dsp3.h
ac_491.h
dsp.xsl
IP2x2: Fixed fkey handling of builtin test
| Status | Closed |
| Id | 72442 |
Fkey test did not work for touchkeys, key symbols were misplaced on screen
myPBX: Improve presentation of chat conferences
| Status | Closed |
| Id | 72444 |
Chat conferences should not be displayed as multiple calls but as a single item in the main window.
IP232: Parameters for touch sensor fixed for latest sensor
| Status | Closed |
| Id | 72455 |
latest touch was treated like the very old version that needed other parameters.
Old touch is not supported any more ( only 3 were build)
Status:
edt_touch.cpp
IP241,IP222,IP232: Make password configuration more convenient
| Status | Closed |
| Id | 72456 |
When entering a passwort on the phone's ui content is displayed as "****".
Show last enered character in plain text.
Ip6010 DSP Allow coder change from T38 to voice and back to T38
| Status | Closed |
| Id | 72457 |
..
Status:
ac_dsp3.cpp
| Status | Closed |
| Id | 72477 |
Center key should enter selected menu item.
Not leaving the current menu screen.
ESC key can be used to leave current menu screen.
Phone app will ask whether to save changes.
myPBX: Name and Number Display not correct on IM sessions across PBXs
| Status | Closed |
| Id | 72491 |
for local sessions (same PBX) the number and the Long Name was displayed for the remote party and this should be the same for remote session.
Sometimes the number was missing, sometimes the Name was displayed instead of Long Name.
The number was not adjusted correctly if different Nodes were involved
SoftwarePhone: Trap and media channel after conference
| Status | Closed |
| Id | 72511 |
If a call is disconnected during conference mode, the SoftwarePhone traps or there is no voice channel to the remaining party. This is fixed now.
HTTP: Accept authentication with both Latin1 and UTF-8 coding
| Status | Closed |
| Id | 72512 |
The server does not know what encoding is used in Basic- or Digest-Authentication. Therefore both encodings shall be tried.
ip28 codec hang when TEL port configured to '600Ohms'
| Status | Closed |
| Id | 72521 |
this option experienced problems in the past and should therefore not be selected. If selected the firmware will switch to default CTR21.
PBX: Cause for release/reject of a IM session not signaled
| Status | Closed |
| Id | 72533 |
Better feedback to the User
myPBX: New icons for unknown phone and im status
| Status | Closed |
| Id | 72536 |
The difference between open, closed and unknown was not understood well.
SoftwarePhone: Dialing outbound calls temporarily failed
| Status | Closed |
| Id | 72628 |
When there was a held call and a consultation call and the consultation call was released by the remote peer the SoftwarePhone did not accept further outbound calls until the held call had been released. This is fixed now.
IP232: Fix for touch keyboard handling
| Status | Closed |
| Id | 72651 |
When editing call diversions, one of the on/off controls may render across touch keyboard.
Content of multi-line-edit-control was mis-placed when touch-keyboard was activated.
Display of first matching directory entriy on indirect dialing screen.
ip241 - monitor mode (handset + speaker) did not work in V9hotfix5
| Status | Closed |
| Id | 72702 |
monitor mode (handset + speaker) was displayed but the speaker was mute
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
Web-UI: Wider input fields for domain names
| Status | Closed |
| Id | 72723 |
When configuring
- Remote Domain
- Local Domain
- Proxy
- STUN Server
on a Gateway interface there's need for more space.
Trap: When Dectmaster registers user at PBX using SIP protocol
| Status | Closed |
| Id | 72729 |
When Dectmaster registers user at PBX using SIP protocol
SIP: Add payload type for RTP-DTMF in case of media-relay
| Status | Closed |
| Id | 72732 |
Add payload type for RTP-DTMF (telephone-event) to SDP offer in case of media-relay.
Should help on Mobility scenarios.
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
IP4001/IP6000 -UART driver did not work
| Status | Closed |
| Id | 72743 |
the hardware requires the registers to be written 32-bit wise which was not respected by a fix one year before.
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
IP241,IP222,IP232: Wrong call state displayed
| Status | Closed |
| Id | 72748 |
"Destination reached" displayed instead of "subscriber busy" if consultation call was rejected.
PBX Waiting: Ringback missing when using DTMF to dial from one WQ to another which is alerting
| Status | Closed |
| Id | 72766 |
This was a collateral damage for a fix for Waiting Queue announcements from a Boolean Object
IP222,IP232: Long key press on numeric block while being offhook does not open directory search
| Status | Closed |
| Id | 72831 |
Long key press during offhook is expected to open directory search.
| Status | Closed |
| Id | 72833 |
Re-design
IP241,IP222,IP232: Support for unicode 0x308 (diaeresis/umlaut)
| Status | Closed |
| Id | 72859 |
When ä,ö,ü are encoded as a,o,u followed by diaeresis from unicode block 'Combining Diacritical Marks' only a,o,u where displayed.
Now ä,ö,ü are displayed.
IP241,IP222,IP232: Support for hebrew and arabic presence notes
| Status | Closed |
| Id | 72905 |
Hebrew and arabic text passages are rendered frmo right to left.
PBX: Conference no media
| Status | Closed |
| Id | 72915 |
There is sometimes no media for a conference member. This is fixed, now.
SIP: Memory leak during transfer
| Status | Closed |
| Id | 73003 |
Occured on internal testing only (002-conf-with-bcast.xml)
IP241,IP222,IP232: Update for CFU indication on idle screen
| Status | Closed |
| Id | 73013 |
Not displaying "cfu:241" on header of idle screen
but "Diverted to 241"
or "Umgeleitet zu 241"
or "Renvoyé sur 241"
etc.
myPBX: Simplify adding multiple users to a chat session
| Status | Closed |
| Id | 73016 |
Do not unselect the chat session, after a person has been added. Replace "start chat" buttons by "add to chat" buttons when a chat session is selected.
RTP-DTMF: Digit may get lost during media re-negotiation
| Status | Closed |
| Id | 73037 |
Receiver starts handling on END event.
Sender may stop sending before END event was sent.
myPBX: Do not show bubble for outgoing chat calls and messages
| Status | Closed |
| Id | 73039 |
Notifications are only needed for incoming calls and messages.
myPBX: Skript errors in chat window
| Status | Closed |
| Id | 73040 |
Closing the chat window caused different script errors in Internet Explorers and Firefox.
LDAP replication from v7 could stop
| Status | Closed |
| Id | 73052 |
LDAP replication from v7 could stop
SIP: Don't write SRTP key into T.38 part of media description
| Status | Closed |
| Id | 73112 |
v=0
o=- 1295 1 IN IP4 10.120.55.3
s=-
c=IN IP4 10.120.55.6
t=0 0
m=audio 20026 RTP/SAVP 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=inactive
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb
m=image 0 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb
PBX External UC: Support of multisite setup
| Status | Closed |
| Id | 73115 |
This requires all calls to the External UC to be routed to one interface. Before this fix the calls were routed within the local PBX only.
Now a Name (not Long Name as before) has to be configured for the External UC interface.
IP241,IP222,IP232: Show number of missed calls in status bar
| Status | Closed |
| Id | 73140 |
New extended symbol is more eye-catching.
PBX: Called party number wrong at original called phone with forking
| Status | Closed |
| Id | 73145 |
If the forking destination and the original called phone are assigned to different nodes
Media Negotiation: v9 XPARENT not compatible to v8 or earlier XPARENT
| Status | Closed |
| Id | 73153 |
In v9 a dynamic payload type is used for XPARENT to be compatible to SIP, whereas in v8 an earlier payload type 0 was used. Within the media negotiation this should be detected and switched back to payload type 0.
Phones: Update of phone text data
| Status | Closed |
| Id | 73211 |
Some translations missing (Polski,Eesti).
Support for language typical letters (Polski).
Some abbreviation removed (for color phones).
PBX: Conference GUI update with IE
| Status | Closed |
| Id | 73256 |
The PBX broadcast conference object window is not closed if IE is used. This fixed now.
SIP: support for "Content-Type: multipart/alternative"
| Status | Closed |
| Id | 73260 |
Lync interoperability
IP2x2: Keyboard handling on call reroute
| Status | Closed |
| Id | 73268 |
Redial key launches "Transfer" screen.
After entering the destination number, Redial key should cause execution of call reroute.
PBX Waiting: When using a waiting queue for outgoing dialing, the announcement was cut off at the beginning
| Status | Closed |
| Id | 73274 |
This was because the announcement was started already when the call was sent out and not when the call was connected
myPBX: Allow spaces in last LDAP attribute
| Status | Closed |
| Id | 73276 |
Currently "j doe innovaphone" finds all internal users starting with j. It should only find internal users that start with "j doe innovaphone".
IP232: Cursor positioning when touching ip address control
| Status | Closed |
| Id | 73407 |
Make sure input cursor is at right side of touched octet.
SIP: Configuration of SIP response code mappings did not work on IPxx10 and IPVA
| Status | Closed |
| Id | 73453 |
PBX: Reporting licenses counting could be wrong if config templates were used
| Status | Closed |
| Id | 73466 |
It could happen that a reporting license was used up by a config template object, if reporting was checked and other config templates where referenced
myPBX: Allow overriding implicit allows
| Status | Closed |
| Id | 73471 |
It was not possible to define an allow with the same name. This is needed to override allows that are inherited from templates or group memberships.
IP241,IP222,IP232: Trap when scrolling down long call list
| Status | Closed |
| Id | 73494 |
Out of memory trap when scrolling down long call list.
Trap: When using Diagnostics/Ping
| Status | Closed |
| Id | 73536 |
When leaving Diagnostics/Ping or Diagnostics/Traceroute web screen a trap may occur.
| Status | Closed |
| Id | 73567 |
Internet explorer buggy concerning url encoding of href attributes
IPVA, crash-dump without backtrace, if tracing was switched off internally
| Status | Closed |
| Id | 73574 |
IPVA, crash-dump without backtrace, if tracing was switched off internally
ISDN: Do not provide tones and no signaling of inband info for unrestricted digital information calls
| Status | Closed |
| Id | 73603 |
There is equipment, which is doing unrestricted digital information ISDN calls, which gets confused if there is an ALERT message indicating inband tones (ringback).
IP241,IP222,IP232: Symbol "new messages" and symbol "headset" do overlap in status bar
| Status | Closed |
| Id | 73608 |
Moved "headset" symbol to right side of status bar.
IP-DECT: DECT endpoint update for multicast identities
| Status | Closed |
| Id | 73613 |
The update information to DECT is changed if the user logs out a handset. This is used to update the multicast identities in some OEM handsets.
Ip6010 DSP Allow calls to start with T.38
| Status | Closed |
| Id | 73626 |
Needed for some T38 fax applications Status: ac_dsp3.cpp
IP241,IP222,IP232: Missing "Recall possible" display text on call completion
| Status | Closed |
| Id | 73628 |
Missing display text on 'recall possible' notification.
SIP: Media negotiation problem on outbound call from BC Conference object
| Status | Closed |
| Id | 73652 |
INVITE was sent without SDP offer.
test/9.00/pbx/conf/SIP/002-conf-with-bcast.xml
IP241,IP222,IP232: Extra digits dialed when using RIGHT key on screnn 'indrect dialling'
| Status | Closed |
| Id | 73694 |
'123123' instead of '123' was dialed.
ISDN Trunk: Transfer to ISDN Trunk with TONE interface failed
| Status | Closed |
| Id | 73695 |
There was not media after the transfer
H.323: Renegotiation to SRTP after dialtone failed
| Status | Closed |
| Id | 73698 |
Happens with Escape Dialtones configured in PBX and calls with SRTP enabled
Option added to wait for the DSP to disconnect fax, needed only for some fax devices.
| Status | Closed |
| Id | 73748 |
Use the prodedure of
Mantis 67821: ip22/24/28/302/305: DSP affected by certain fax tones
only if the option /t38-wait-fax-close set.
This avoids traps caused by disconnect timeouts if the DSP response is not received in time.
Status:
ac_dsp3.cpp
ac_dsp3.h
dsp.xsl
SIP: Using wrong remote port when registering
| Status | Closed |
| Id | 73784 |
Only affects IP-DECT when handset is switched OFF and ON and if the SIP runs on non-standard port.
IP232: Memory leak in display rendering
| Status | Closed |
| Id | 73864 |
Memory leak in display rendering
V9 Hotfix 7 (90600.08)
Changes included in Version 9 hotfix7 Definition
New Features
IP241,IP222,IP232: Some diagnostics to check phone's ui memory consumption
| Status | Closed |
| Id | 74049 |
Where have all the memory gone.
SIP: Support for "Content-ID" in "multipart/alternative" bodies
| Status | Closed |
| Id | 74161 |
Support for "Content-ID" in "multipart/alternative" bodies
| Status | Closed |
| Id | 74278 |
To make end of list of menu entries more discernible.
myPBX: Support static URLs for application sharing
| Status | Closed |
| Id | 74389 |
In some application sharing solutions a fixed link can be used to create and join meetings (GoMeetNow, BeamYourScreen).
There are two URLs configured, one for the presenter and one for attendees. When the user clicks the aplication sharing button the links are sent using chat messages.
PBX: Tracing flag turns on tracing in all dyn PBX's as well
| Status | Closed |
| Id | 74390 |
Helpfull to debug dyn PBX setups
debug method raw_ethernet
| Status | Closed |
| Id | 74897 |
A new debug method to trace raw ethernet data.
Support for new hardware
| Status | Closed |
| Id | 74990 |
.
possible use of push-pull drivers for new hardware
| Status | Closed |
| Id | 75051 |
not visible to customers
PBX/IP-DECT: User password length
| Status | Closed |
| Id | 75453 |
The maximum user password length in the PBX and IP-DECT is increased to 23.
H.323: Support for switch back to local media on endpoint
| Status | Closed |
| Id | 75522 |
needed to resume normal call after a conference using external conference unit
IP232: Touch gesture to open directory-search/indirect-dialing screen
| Status | Closed |
| Id | 75549 |
Vertical upwards on idle screen opens directory-search/indirect-dialing screen.
Vertical downwards on idle screen opens main-menu screen.
PBX-SOAP: Wsdl Versioning mechanism fixed
| Status | Closed |
| Id | 75552 |
Better mechanism, to avoid constantly adding arguments to the Version function with new Versions.
different port naming convention for new hardware
| Status | Closed |
| Id | 75575 |
BRI1..5 instead of TEL1..4+PPP
PBX-SOAP: UserHold without MOH to local User
| Status | Closed |
| Id | 75577 |
UserHold was sending MOH to the local and the remote User. With the argument remote=true, the MOH is sent to the remote user only
myPBX launcher: Use default configuration from local machine registry hive
| Status | Closed |
| Id | 75634 |
If no user configuration is given, the following values are copied from the local machine hive.
"Software\\innovaphone\\myPBX\\URL"
"Software\\innovaphone\\myPBX\\secondaryURL"
Also the autostart checkmark is disabled, if autostart is activated, globally. Autostart is controlled with the following registry key.
"Software\\Microsoft\\Windows\\CurrentVersion\\Run\\innovaphone myPBX"
SIP: support for proprietary message header
| Status | Closed |
| Id | 75864 |
Support for proprietary message header (CAL)
H.323: Call to SIP Trunk with MediaRelay and exclusive coder renegotiation from TONE to early media did not work
| Status | Closed |
| Id | 75959 |
No ringback and no audio after connect
SIP: Added Call-Info header to re-INVITE for hold/retrieve
| Status | Closed |
| Id | 76011 |
Call-Info: <urn:X-cisco-remotecc:hold>
Call-Info: <urn:X-cisco-remotecc:resume>
SIP: Support for active call pickup acc. to RFC-3891
| Status | Closed |
| Id | 76035 |
Sending INVITE with Replaces header to do pickup at SIP PBX's
phone ip222, ip232 : USB headset support (beta)
| Status | Closed |
| Id | 76463 |
general support for USB headsets added, some headsets are tested and working, others will follow soon
phone - Headset Function Key can be configured now as enable/disable or as call control key
| Status | Closed |
| Id | 76468 |
a headset key mode can be configured to use the key either to enable/disable the headset (Mode: Enable) or to start/accept/clear calls via headset (Mode: Control)
Bug Fixes
Ip6010 DSP Allow coder change from T38 to voice and back to T38 with local DSP
| Status | Closed |
| Id | 73193 |
..
Status:
ac_dsp3.cpp
ISDN: Enable fax detection only after connect
| Status | Closed |
| Id | 73213 |
On ISDN networks it can happen that the Connect message is delayed. This way fax tones are forwarded to the caller before the caller has received this Connect.
This way a renegotiation on voip to fax could happen before the connect, which is not supported by sip.
Phones: Update of phone text data
| Status | Closed |
| Id | 73775 |
External input
SIP: Handling of collision of transfer and release
| Status | Closed |
| Id | 73936 |
If one end releases a call while the other initiates an attended transfer, a "ghost call" may remain.
Resource leak.
IP241,IP222,IP232: Limitation of background image size
| Status | Closed |
| Id | 73957 |
Limit is 512 kBytes now.
To keep phone from out of memory trap.
H.323: Renegotiation after PBX Node dialtone failed sometimes
| Status | Closed |
| Id | 74020 |
This depended on timing. This happened if the call was sent after the dialtone to a master over a slow link.
| Status | Closed |
| Id | 74046 |
Touching header bar and moving vertically down opens the phone menu.
No need to use center key of 4-way control.
Dect User Gui: AD-replicated objects weren't listed under certain conditions
| Status | Closed |
| Id | 74051 |
Deleted object thate were re-replicated werent't visible in the Dect user GUI. A mandatory Ldap attribute was missing under such circumstances.
IP222: Handset Microphone level too low
| Status | Closed |
| Id | 74073 |
Handset micro increased by 6db
Handsfree micro unchanged
Status:
ac_codec3.cpp
IP222,IP232: Language configuration not saved
| Status | Closed |
| Id | 74087 |
Language can be selected on phone menu, but change is neither saved nor applied.
SIP: Don't take NOTIFY(application/simple-message-summary) with "Do-Not-Disturb: no" as FTY_MWI_DEACTIVATE
| Status | Closed |
| Id | 74162 |
Interop with Nortel CS1k PBX
Phones: Fine grained function hiding did not disable Fkey configuration
| Status | Closed |
| Id | 74174 |
Value 0x00000800 (PHONE_HIDE_US_FUNCTION_KEYS) hides function keys entry from user-list,
but fkeys can still be (re)configured by long fkey press.
For more information see:
http://wiki.innovaphone.com/index.php?title=Reference9:Phone/Protect
myPBX: Allow asterisk as a wildcard in LDAP search strings
| Status | Closed |
| Id | 74200 |
As it is done on our phones.
IP232: Pixel noise with some color values
| Status | Closed |
| Id | 74221 |
- changed to lower pixel clock Status: orchid_lcd.cpp
PBX: Conference to VM
| Status | Closed |
| Id | 74222 |
Now a call to the VM as a conference member works again.
RTP-DTMF: Digit may get lost during media re-negotiation
| Status | Closed |
| Id | 74232 |
Receiver starts handling on END event.
Sender may stop sending before END event was sent.
No media for calls from SIP provider forwarded back to SIP provider in case of NAT and MediaRelay
| Status | Closed |
| Id | 74290 |
NAT traversal depends on a packet being sent from inside the NAT to outside, to fix the RTP destination of the outside endpoint. This does not happen if both endpoints are outside.
Dummy packets are sent from the Media Relay function in this case to achieve this.
ISDN: Send HLC with mobility calls
| Status | Closed |
| Id | 74296 |
Some ISDN networks refuse the forwarding of a call to a mobile network if no HLC (High Layer Compatibility) Information Element indicating Telephony is included in the call.
IP241,IP222,IP232: Improved phone screen rendering
| Status | Closed |
| Id | 74301 |
Reduce map copy operations.
SIP: Accept INVITE's from alternative proxy
| Status | Closed |
| Id | 74310 |
If alternative proxy is configured, calls should accepted from alternative proxy,
even if primary proxy is alive and kicking.
Flash Directory: Substring match could miss results
| Status | Closed |
| Id | 74320 |
Searching by substring worked case sensitive in some cases, instead of functioning case insensitive.
PBX: CF at Gateway Type objects - additional dialed digits should be added to the destination
| Status | Closed |
| Id | 74348 |
This way a CFNR at a trunk object can be used to reroute the call to another trunk.
Gateway: Deleting of routes could result in duplication of routes
| Status | Closed |
| Id | 74356 |
This happened if an interface registration was disabled, for which automatic routes have been generated and then a route was deleted. The last route was duplicated.
IP241/222/232: Monitormode (Lauthören): Level too low
| Status | Closed |
| Id | 74362 |
Use independent analog codec channels for speaker and headset receiver.
Speaker volume in monitor mode is configured as in handsfree mode.
To change to speaker level in monitor mode the gain of the speaker can be configured with
config change AC-DSP0 RINGER /DualOutputModeGain level
config activate
level is from 0..63 0
0 -> -32dm
32 -> 0db
63 -> 31db
Status:
ac_codec3.cpp
ac_codec3.h
| Status | Closed |
| Id | 74367 |
Could not activate first builtin background image from phone menu if external background image is currently active.
IP241,IP222,IP232: Show multiple diversions on incoming calls
| Status | Closed |
| Id | 74371 |
If a call was diverted more than once, the phone shows first diverting party (original called number) and last diverting party.
(Not only the last diverting party)
myPBX launcher: Unhandled exception when accessing browser object
| Status | Closed |
| Id | 74380 |
The myPBX launcher could crash when accessing the browser document because of an unhandled exception.
IP232: Pickup fkey displays no information in active state
| Status | Closed |
| Id | 74392 |
Pickup fkey displays no information in active state
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)
NAT: Permanent UDP forwarding did not work
| Status | Closed |
| Id | 74549 |
Permanent UDP forwarding was unusable after first forwarded datagram.
PBX Waiting: If used as outgoing dialing device, coder list sent with the call was not correct
| Status | Closed |
| Id | 74593 |
Should be based on the coders paramter in the configured URL
Phone_orchid: Residual echo in handset ( sporadic)
| Status | Closed |
| Id | 74671 |
Echo canceller NLP sensitivity Mode changed to 2 Status: ac_dsp3.cpp
IP222,IP232: Function key "Hotdesk" did not work
| Status | Closed |
| Id | 74723 |
Function key "Hotdesk" did not work
IP232: Fix for long touch on fkey
| Status | Closed |
| Id | 74743 |
Opens fkey config, but mist not toggle control on touch-off.
Web-UI: Configuration of "Sync Server" does not return with HTML page
| Status | Closed |
| Id | 74778 |
Configuration of "Sync Server" does not return with HTML page
| Status | Closed |
| Id | 74792 |
Even after confirming to save
IPxx10: Flashdir Segments Default to 129(was 51)
| Status | Closed |
| Id | 74826 |
IPxx10: Flashdir Segments Default to 129(was 51)
Interop: Don't add an error log "SRTCP authentication failed" on non-RTCP packets
| Status | Closed |
| Id | 74903 |
Microsoft Lync send invalid RTP and RTCP packets at the beginning of each call.
In case of encrypted media, decryption of these packets fails.
phone_orchid: ethernet statics counters sometimes displayed as negative numbers
| Status | Closed |
| Id | 74913 |
the unsigned counters were printed as signed integers
IP241,IP222,IP232: Volume control not displayed when already at maximum
| Status | Closed |
| Id | 74927 |
Volume control is not displayed when trying to increase while being at maximum.
IP6000 crypto driver: Trap when buffers are depleted
| Status | Closed |
| Id | 74935 |
Avoid the trap and log an Event when the buffers are depleted.
IP241,IP222,IP232: Mute status indication does not work
| Status | Closed |
| Id | 74964 |
"MICROPHONE OFF" is displayed on status bar now.
SIP: Rare problem with blind transfer
| Status | Closed |
| Id | 74991 |
callIdentity must be zero when giving ctInitiate to PBX on blind transfer.
TLS: Flow control for incoming data
| Status | Closed |
| Id | 75004 |
The TLS socket has to wait for the application to process incoming data before sending the next RECV.
H.323: Renegotiating to Fax was rejected, if PCM switch and local networks were used
| Status | Closed |
| Id | 75006 |
If the side which initiated a switch to T.38 has configured PCM and the media address was classified as local due to local network configuration, the T.38 was rejected.
SIP: Interworking of divertingLegInformation1 improved
| Status | Closed |
| Id | 75025 |
Honour subscriptionOption
IP222,IP232: Changes not saved in some cases
| Status | Closed |
| Id | 75088 |
Some config screens did not write changed settings directly after "Save Changes" dialog.
If menu was left with DISC key (instead of ESC key) the changes have been discarded.
Also affects other phones: IP241,IP240,IP230,IP110
Changes are saved immediately when leaving the current screen.
Not when leaving "User Settings" or "Phone Setting" screen.
PBX Waiting: Diverting leg1 info not correct when diverting to a Waiting Queue
| Status | Closed |
| Id | 75091 |
If Waiting queue not defined in root node and escapes are used
PBX: Adjusting received leg1 info wrong
| Status | Closed |
| Id | 75097 |
In context of nodes with escapes
VM: <pbx-upd-obj type="cfu"..> without effect when invoked multiple times
| Status | Closed |
| Id | 75121 |
Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.
IP28 Watchdog leads to endless interrupt and trap
| Status | Closed |
| Id | 75186 |
.
IP800/6000: Problem mit FAX und CNG detect: Option to disable CNG detect added
| Status | Closed |
| Id | 75278 |
config change AC-DSP0 /t38-cng-detect-disable 1
config write
config activate
Status:
ac_dsp3.cpp/h
ac_fax3.cpp/h
SIP: Decrement Max-Forwards and interwork to H.323
| Status | Closed |
| Id | 75288 |
Instead of sending always "Max-Forwards: 70"
H.323: Renegotiation to T.38 did not work for slowstart call thru media-relay to efc endpoint
| Status | Closed |
| Id | 75305 |
A common scenario when this happend is an XCAPI (slowstart) calling thru a PBX with media-relay configured to an ISDN interface
Fix for MIPS counter
| Status | Closed |
| Id | 75310 |
MIPS counter was incorrect
PBX: Allow 'Max Calls' configuration of 0
| Status | Closed |
| Id | 75362 |
So that no call at all is sent to the object but a CFB (if configured) is executed
SIP: Send "305 Use Proxy" if INVITE is received from unexpected source
| Status | Closed |
| Id | 75380 |
Applies to registered interfaces only (e.g. phones).
TLS: Possible trap when restoring sessions
| Status | Closed |
| Id | 75394 |
Only in v9, depending on the certificate.
PBX Exec Object: Leg2 information send to secretary not adjusted correctly for nodes
| Status | Closed |
| Id | 75431 |
The leg2 information is used to display at the secretary the number of the exec which was called. This number was not correct if nodes with escapes were used
Gateway Interface Maps: Should be applied to leg1 info also
| Status | Closed |
| Id | 75437 |
The same rules used for a Calling Party Number are applied to leg1 info
SoftwarePhone: Timer tick resolution
| Status | Closed |
| Id | 75447 |
The timer tick resolution is increased. This fixes the call RTCP calculation transmitted to the PBX.
IP241,IP222,IP232: Indirect dialing of names not possible
| Status | Closed |
| Id | 75451 |
Could not dial a name from 'indirect dialing' screen
by moving the focus to input field and going offhook.
(Only fkey worked)
PBX: CFNR Loop check detected loops that weren't
| Status | Closed |
| Id | 75465 |
Esspecially a CFNR at a Waiting Queue was not executed if the destination of the CFNR has transfered the call to the Waiting Queue
Gateway: Event "No Media Data Received" was created for each SWITCH-PCM call
| Status | Closed |
| Id | 75483 |
There should be no event in this case, this is normal.
PBX: Conference id prefix/suffix configuration
| Status | Closed |
| Id | 75507 |
The broadcast conference PBX configuration is changed: now, the third party conference unit option is saved and must be enabled to use the configured id prefix and suffix. Otherwise they are ignored now and default values for the innovaphone conference interface of the current device firmware version are used. This fixes the configuration if the firmware is updated from V8 to V9 and the innovaphone conference interface is used.
Disabling the "Create Dynamic Conference Id" option in firmware V9 hotfix 5 and 6 is also fixed now.
IP241,IP222,IP232: Call duration display wraps after 100 minutes
| Status | Closed |
| Id | 75508 |
Now after 60 minutes the display changes from mode [mm:ss] to [hh:mm].
TLS: Duplicate alert message on malformed ClientHelloV2
| Status | Closed |
| Id | 75509 |
Only one alert should be sent per session.
TLS: Improved negotiation of protocol version
| Status | Closed |
| Id | 75510 |
TLS server unnecessarily rejected ClientHello messages with TLS 1.1 and higher. Instead of rejecting it should tell the client that it wants to use TLS 1.0.
TLS: Skip empty records
| Status | Closed |
| Id | 75511 |
TLS record layer should ignore records with zero length without doing anything.
myPBX: Script error after waking the PC up from hibernation
| Status | Closed |
| Id | 75532 |
Sometimes window.open throws an unhandled exception in this case.
SIP: Check proxy availability did not work in any scenario
| Status | Closed |
| Id | 75542 |
OPTIONS can be used to poll remote proxy's availablity
to avoid TCP timeout when INVITE is to be sent.
Signaling interface is marked as down and not used anymore.
IP232: Hiding touch keyboard by touching a control
| Status | Closed |
| Id | 75576 |
Touching a text control activates the touch keyboard.
Touching the text control again now deactivates the touch keyboard.
IP241,IP222,IP232: CFU information in header bar
| Status | Closed |
| Id | 75654 |
CFU information in header bar is now displayed even if there's not is enough space between name and number.
Either name or number is omitted is required.
IP241,IP222,IP232: Replace triangle by arrow to display diversion/transfer information
| Status | Closed |
| Id | 75689 |
Replace quite heavy 'BLACK RIGHT-POINTING POINTER'
by much lighter 'RIGHTWARDS ARROW'
to display diversion/transfer information
on call control, fkeys and call lists.
H.323: A forwarded HopCount>32 could result in a very small HopCount
| Status | Closed |
| Id | 75765 |
There are only 5 bits for transmitting a HopCount in H.323. A HopCount from SIP is typically 70 and this value was not reduced to 32 but only the 5 lower bits were transmitted, which resulted in a HopCount of 6
IP: Minor memory management change
| Status | Closed |
| Id | 75770 |
no use of malloc in ipproc
IP241,IP222,IP232: Two waiting calls are now displayed (instead of one)
| Status | Closed |
| Id | 75789 |
Waiting calls are displayed below the two main calls.
Only two lines of information and grey background.
SIP: Handling of some more proprietary Alert-Info and Call-Info in INVITE
| Status | Closed |
| Id | 75824 |
Alert-Info: <Bellcore-dr3>;info=alert-recall
and
Alert-Info: <http://not_used.com>;info=alert-autoanswer
and
Call-Info: <sip:127.0.0.1>;answer-after=0
PBX Waiting: Leg2 information sent with calls to operators not adjusted correctly for nodes
| Status | Closed |
| Id | 75830 |
If a Waiting Queue was configured in a Node not the root node, the leg2 info was not adjusted corrcectly. The leg2 information is used to signal to the operator which Waiting Queue is forwarding the call
SIP: Decoding of RFC-4412 definitions
| Status | Closed |
| Id | 75848 |
The 'Resource-Priority' Header Field
The 'Accept-Resource-Priority' Header Field
The 'resource-priority' Option Tag
417 Unknown Resource-Priority response
SIP: Handling of 422 Session Interval Too Small
| Status | Closed |
| Id | 75873 |
Get Min-SE and re-try INVITE
SIP: Send b=TIAS attribute in media description
| Status | Closed |
| Id | 75881 |
Send b=TIAS attribute in media description
remove indirect calls to os_mem_alloc() via malloc()
| Status | Closed |
| Id | 75886 |
malloc() shall not be used anymore in sources dedicated to run on innovaphone hardware
IPv6: Memory leak if packets received, which are not handled locally
| Status | Closed |
| Id | 75920 |
E.g. an IPv6 multicast UDP packet, for some other application was not deleted.
SIP: Fix for video negotiation
| Status | Closed |
| Id | 75954 |
Fix for video negotiation
phone_orchid: after leaving a conference the active call was mute sometimes
| Status | Closed |
| Id | 75955 |
The conference was not stopped internally in all cases.
Flashman: Show meaningful result after OEM certificate upload in production
| Status | Closed |
| Id | 75958 |
"Certificate" instead of "-unknown-"
SIP: Fix for Supported header
| Status | Closed |
| Id | 76016 |
Re-implemented to easily add more tags
IP222,IP232: Alpha mode did not work on multi-line edit controls
| Status | Closed |
| Id | 76165 |
Could no enter alpha chars using the num block
SIP: Trap when parsing very large History-Info header
| Status | Closed |
| Id | 76198 |
Trap when parsing very large History-Info header
SIP: SIP message may exceed available buffer
| Status | Closed |
| Id | 76203 |
Trace message like this can occur:
ERROR: SIP message buffer (1025) exceeded! (9,14,801d781c)
SIP: CSeq missing
SIP: Message encoding failed!
SIP: Contact not allowed in BYE/CANCEL/PRACK request
| Status | Closed |
| Id | 76210 |
Acc. to RFC-3261 there must not be Contact header line in BYE/CANCEL request
Acc. to RFC-3262 there must not be Contact header line in PRACK request
reset could fail, if issued during firmware update
| Status | Closed |
| Id | 76266 |
and even worse any following reset command were not accepted as well
IP241,IP222,IP232: Show "New message from ..." on active phone screen
| Status | Closed |
| Id | 76283 |
Show "New message from ..." on phone screen for 3 seconds when active with calls (non-idle).
Like on b/w phones.
IP241,IP222,IP232: Toggle of display-name/name-alias/number was buggy
| Status | Closed |
| Id | 76290 |
String termination was missing.
Garbage data was displayed.
IP232: Open call details when touching entry in call list
| Status | Closed |
| Id | 76292 |
Instead of initiating call to remote party in call list entry.
H.323: Merges from v10 for Media Negotiation fixes and special OEM protocol features
| Status | Closed |
| Id | 76301 |
- Problem with transparent forwarding of SDP
- special SRTP key exchange
- Video fixes
- Unit Testing support
IP-DECT: Memory leak with rejected radio registration
| Status | Closed |
| Id | 76308 |
Memory leak occurred with a rejected radio registration is fixed, now.
PBX-SOAP: When monitoring Waiting queue, the peer number was not indicated
| Status | Closed |
| Id | 76315 |
The number was considered presentation restricted, because of some bit manipulation error
ip22/24/28/302/305/6010/3010/1060: Fax failed if tones are sent during the fax call
| Status | Closed |
| Id | 76342 |
In some cases the Fax messages were missing due to tones send during the fxx call Status: ac_dsp3.cpp
Gatway: Transfer handling may cause re-routing after regular hang-up
| Status | Closed |
| Id | 76357 |
Re-routing is expected to take place on non-connected calls only.
PBX Number Map: Not possible to use overlap dialing to Number map with incomplete destination
| Status | Closed |
| Id | 76377 |
If a Number Object with incomplete destination was called and the number was to be completed with overlap dialing a wrong number was called.
This is a usefull feature to use Number Maps as quick dial to other nodes. In this case Number Maps are used with a destination of the remote node, so the number is incomplete, the number within this node has to be dialed in addition to the Number of the Number Map object.
SIP: Support for media recording
| Status | Closed |
| Id | 76396 |
Media recording to HTTP URL
Gateway: Conference interface, no voice
| Status | Closed |
| Id | 76419 |
The ADSP firmware is changed to version 122. This fixes a bug in the conference interface of IP6000/IP6010/... which results in conference calls without voice in one direction for a single member.
phone_orchid: spurious trap in long conference calls
| Status | Closed |
| Id | 76445 |
in long confernce calls the phone may trap because the DSP delivers an unexpected zero size packet.
IP241,IP222,IP232: Call held indication
| Status | Closed |
| Id | 76454 |
If remote side has put the call on hold
the user should get a visible indication
(in addition to hearing music on hold).
New call status "held" is displayed.
LDAP-Expert, Edit Object Dialog: Could show more than one object
| Status | Closed |
| Id | 76464 |
LDAP-Expert, Edit Object Dialog: Could show more than one object
Qsig: Connected Number could be encoded wrong
| Status | Closed |
| Id | 76472 |
Qsig: Connected Number could be encoded wrong
Gateway: Better handling of call-reroute requests
| Status | Closed |
| Id | 76515 |
Apply interface maps to numbers in reroute request
PBX-SOAP: Better error handling when using a Waiting Queue for outgoing calls
| Status | Closed |
| Id | 76544 |
- call was hanging if an invalid number or user without registration was called
- trap if outgoing call was done on operator connect
Spurious leaks in leak checking
| Status | Closed |
| Id | 76589 |
A problem which mostly shows up in automated release testing and then requires manual investigation.
SoftwarePhone: Auto start configuration
| Status | Closed |
| Id | 76595 |
The duplicate back slash occurred with the install directory is fixed now. Used by the run mode configuration 'At login'.
myPBX launcher: Window position was not always remembered
| Status | Closed |
| Id | 76626 |
When the user just moved the window without resizing it, the window position was not remembered for the next session.
V9 Hotfix 8 (90600.10)
Changes included in Version 9 hotfix8 Definition
New Features
myPBX: Log event if application sharing failed
| Status | Closed |
| Id | 76116 |
Log an event if the creation of a meeting failed.
myPBX: Localised date and time format in call list
| Status | Closed |
| Id | 76737 |
myPBX should show a localised date format instead of YYYY-MM-DD HH:MM:SS.
IP241,IP222,IP232: Support for placeholder in background image url
| Status | Closed |
| Id | 76756 |
Support for $serial and $product in background image url
URL
http://x.x.x.x/DRIVE/CF0/backgrounds_480x272/$serial.png
results in
GET /DRIVE/CF0/backgrounds_480x272/00-90-33-30-00-bc.png HTTP/1.1
URL
http://x.x.x.x/DRIVE/CF0/backgrounds_480x272/$product.png
results in
GET /DRIVE/CF0/backgrounds_480x272/IP232.png HTTP/1.1
SoftwarePhone: Support for Plantronics Savi 740
| Status | Closed |
| Id | 76975 |
HID Keypad Mapping support for the Plantronics Savi 740 with the product id 0xac01.
phone_orchid: headset "Plantronics - Voyager PRO UC v2" (USB/bluetooth) supported
| Status | Closed |
| Id | 77048 |
HID report definitions added
phone ip222, ip232 : Jabra Link 14201-30 DHSG/USB adapter cable supported,
| Status | Closed |
| Id | 77540 |
The Jabra Link 14201-30 connects the DHSG interface of Jabra Dual Mode Headsets (GN9350e, PRO 9450/9470 ..) to the USB port of an ip2x2. The USB port of the headset can be connected to a PC. Status: checked in to 10.00, 9.0, 90600
phone ip222, ip232 : Plantronics DA45 USB headset supported
| Status | Closed |
| Id | 77541 |
. Status: checked in to 10.0, 9.00, 90600
Trap during HTTP request with missing user/pwd in basic authentication header
| Status | Closed |
| Id | 77695 |
. Status: http.cpp
phone ip222, ip232 : USB headset support : Plantronics Blackwire C620, Jabra GN2000 USB Duo
| Status | Closed |
| Id | 77703 |
beim Kunden getestet Status: checked in to 10.00, 9.00, 90600
Bug Fixes
SIP: Fix for REFER handling
| Status | Closed |
| Id | 76309 |
Handling of blind transfer without userpart in Refer-To-URI.
SIP: Do not offer CLEARMODE on exclusive coder config
| Status | Closed |
| Id | 76654 |
Do not offer CLEARMODE if exclusive coder configured
myPBX launcher: Always save window position when closing the application
| Status | Closed |
| Id | 76763 |
The window position was only saved when the application was closed from the context menu of the tray icon. I.e. it was not saved when the user logged out.
SIP: Handling of 300 responses
| Status | Closed |
| Id | 76771 |
Select Contact-URI according to q-value.
Replace remote target URI and change remote transport address.
IP-DECT: Searching for users containing non-ascii characters failed
| Status | Closed |
| Id | 76778 |
A page indicating an xml error was displayed instead
ip22/24/28/302/305/6010/3010/1060/222/232/241: Silence compression with G711 coders not working
| Status | Closed |
| Id | 76826 |
Silence compression was used only for G729/G723. Now it is used for all codecs. Status: ac_dsp3.cpp
Random problems with T.38 (Fax) Media Relay
| Status | Closed |
| Id | 76870 |
Fax data could be corrupt
phone_inca: wrong display of a CFU to a name containing non ascii characters
| Status | Closed |
| Id | 76877 |
all names are stored UTF8-encoded internally. the UTF8 string was displayed instead of a decoded version.
Phones: Small change on screen "Directory input"
| Status | Closed |
| Id | 76883 |
Put a colonn at the end of label "Subscriber"
phone_orchid: no ring tone when ring volume was set to MAX via WEB config
| Status | Closed |
| Id | 76927 |
but ring tone was OK with any other volume
PBX: Update of remote number after call-transfer did not always work across PBXs
| Status | Closed |
| Id | 76942 |
If a call was done from a phone at a slave to a phone at the master and on the master phone a consultation call was the done to the slave again with transfer afterwards, one phone on the slave did still show as connected number the phone on the master.
The same wrong number was also indicated on SOAP/TAPI as well as on myPBX.
IP241,IP222,IP232: Config file option "/backlight-delay" to adjust display dimming
| Status | Closed |
| Id | 76948 |
Now works as it did on black 'n' white phones:
config change PHONE FORMS /backlight-delay 1800
IP222/IP232/IP241: residual echo in handset mode
| Status | Closed |
| Id | 76958 |
new dsp code Status: ac_494e.cpp
Memory Leaking In 802.1x Module
| Status | Closed |
| Id | 76960 |
Memory Leaking In 802.1x Module
PBX-SOAP: Call initiated by SOAP for softwarephone or IP-DECT was sent as transfered call
| Status | Closed |
| Id | 76962 |
The result was that call diversions or busy on ... calls settings were ignored
Edss1 Interworking: divertingLegInformation2 didn't contain redirectingNumber
| Status | Closed |
| Id | 77003 |
Edss1 Interworking: divertingLegInformation2 didn't contain redirectingNumber
IPVA, IP800, IP6000, IP6010: Typo RT30 instead of RT40
| Status | Closed |
| Id | 77027 |
The routing table entry RT30 was duplicate and RT40 was missing. This causes config changes to be lost if more then 29 routes are configured
Phones: Not able to disable option "Use Newline/OK Key as Headset Key"
| Status | Closed |
| Id | 77070 |
Not able to disable option "Use Newline/OK Key as Headset Key"
Gateway: Routes got lost when reading config file
| Status | Closed |
| Id | 77157 |
Gateway routes got lost when reading config file.
SIP: 0.0.0.0 in o line of SDP
| Status | Closed |
| Id | 77161 |
v=0
o=- 1 1 IN IP4 0.0.0.0
s=-
c=IN IP4 192.168.1.200
t=0 0
m=audio 16386 RTP/AVP 18 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
IP222,IP232: Fix for rendering problem of screen "Directory input"
| Status | Closed |
| Id | 77209 |
When leaving screen was refused with "Subscriber required" popup,
the screen "Directory input" was not rendered again.
SIP: Bug when interworking UUI of 1-byte length
| Status | Closed |
| Id | 77225 |
Results into bad formatted INVITE since User-to-User header contains uninitialized data.
SIP: SDP answer must have the same number of media descriptions as received offer
| Status | Closed |
| Id | 77277 |
SDP answer must have the same media descriptions as received offer
SoftwarePhone: Exception during start up
| Status | Closed |
| Id | 77288 |
Fix for fix #75886.
SIP: DNS refresh is not executed according to DNS-TTL
| Status | Closed |
| Id | 77309 |
Proxy IP address needs to be refreshed by DNS request according to DNS-TTL.
Phones: Show proxy IP address in web UI
| Status | Closed |
| Id | 77316 |
Useful if domain name has been configured and IP address has been resolved by DNS.
IP222/IP232/IP241: residual echo in handsfree mode
| Status | Closed |
| Id | 77358 |
changed ec settings Status: ac_codec3.cpp
SIP: Media negotiation during blind transfer caused ERROR log 0x70007
| Status | Closed |
| Id | 77385 |
Media negotiation during blind transfer caused ERROR log 0x70007 on the outbound call to transfer destination.
IP222/IP232/IP241/IP6010: trace feature tdm record: wrong endian
| Status | Closed |
| Id | 77508 |
..
Status:
ac_dsp3.cpp
ac_phone3.cpp
GUI: Adjust ETH DHCP link
| Status | Closed |
| Id | 77538 |
Adjust ETH DHCP link on page General Info and Reset is fixed now.
DECT: Reset link on page 'Features'
| Status | Closed |
| Id | 77539 |
The reset link on the page 'Features' is fixed now.
Media: Send CN frame with minimal noise level
| Status | Closed |
| Id | 77625 |
For NAT reasons innovaphone endpoints send up to 4 RTP packets to remote end
in order to open mapping on NAT router.
These CN packets carry one-byte payload with value of 0.
Value of 127 turned out to be more clever.
late RTP data sent from DSP driver to media endpoint may cause a trap
| Status | Closed |
| Id | 77642 |
this was observed in a phone_orchid test setup with media recording activated
IP222/IP232/IP241: instable echo canceller in handsfree mode
| Status | Closed |
| Id | 77661 |
changed ec settings, halfduplex on start only switch on again.
IP241 gains adjusted
Status:
ac_codec3.cpp
IP241,IP222,IP232: Illegal characters when editing on a text control
| Status | Closed |
| Id | 77735 |
E.g. when configuring call forwarding destination illegal character (0x8a) may appear in result string.
IP222,IP232: Redial key does not start calling when used in call list
| Status | Closed |
| Id | 77746 |
Redial key does not start calling when used in missed call list
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
IP222,IP232: "Lock phone" does not work
| Status | Closed |
| Id | 77778 |
Authentication dialog for "Lock phone" could not be acknowledged.
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
V9 Hotfix 9 (90600.11, withdrawn)
Changes included in Version 9 hotfix9 Definition
New Features
PBX-SOAP: UserCall new option diversion override
| Status | Closed |
| Id | 75773 |
To initiate calls, which are not diverted by a configured CFU.
Use Info[] element with type="fty-no-cf"
SIP: Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"
| Status | Closed |
| Id | 76658 |
For interoperability:
Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"
Phone->PBX LDAP Search returns normalised number to be dialled by phone
| Status | Closed |
| Id | 77261 |
1)Phone->PBX LDAP Search returns normalised number to be dialled by phone.
2)Phone receives info about escape digits when registering at its PBX.
With 1) the PBX includes an object's normalized number into the LDAP search result.
With 1) the phone is able to dial that normalized number.
With 2) the phone is able to prefix required escape digits to the received normalized number.
Status:
checked in to 10.00, 9.00, 90600
PBX soap method SetPresence
| Status | Closed |
| Id | 78092 |
PBX soap method SetPresence added to set the presence of a PBX user.
phone ip222, ip232 : USB headsets need not to be configured anymore
| Status | Closed |
| Id | 78114 |
If an USB headset with a known signature (vendor/product id) is plugged it is automatically enabled. This is indicated by the headset symbol in the status line.
"Phone/Preferences/Start Outbound Call on Electronic Hook Switch (EHS) Signal" is implied in this case because some headsets will loose state if a hoook signal is ignored.
Status:
checked in to 10.00, 9.00, 90600
phone: ip222, ip232: USB headset support - Plantronics C420, GN2000 USB - MS OC Version
| Status | Closed |
| Id | 78201 |
. Status: checked in to 9.00, 90600
PBX Mobility: No-Alert checkmark as workaround for provider which do not provide Alerting
| Status | Closed |
| Id | 78410 |
Some SIP provider do not provide an Alerting signal when a mobile phone is called. This could result in no ringback signal to the caller or the min/max-alert feature not working.
This new checkmark provides a fake Alerting in case Progress is received
show linux shutdown warning on firmware reset page
| Status | Closed |
| Id | 78665 |
If linux is running, a warning is now shown, that linux should be shutdowned before a firmware reset is performed.
Linux: Enable/Disable support link
| Status | Closed |
| Id | 78786 |
Now the Linux menu is always shown and a link is provided to enable or disable the Linux support (RAM reservation). The support state is also saved in the downloaded configuration file and restored with the upload.
Update: The support state is only saved in the downloaded configuration with password. The state is not saved in the configuration file with standard password or if downloaded by the update server. Please use the next or a later hotfix instead, see also fix #78836.
Bug Fixes
H.323: Media Negotiation problem with conferences on IP-DECT
| Status | Closed |
| Id | 76314 |
A channel was not switched to the conference after a hold/retrieve cycle
PBX Mobility: Dialed digits could get lost, when using Opticaller
| Status | Closed |
| Id | 76598 |
There could be a collision of a dialed digit with media renegotiation. For example if with the first digit a media was switched to inband information from a carrier.
phone: Ring Tone Titles containing apostrophes garble phone configuration
| Status | Closed |
| Id | 76745 |
When under "Phone/Ring Tones/Add Ring Tone" a title containing apostrophes is entered the page "Phone/User-x/Preferences" cannot be edited anymore because of a XML-Error. Status: checked in to 10.00, 9.00, 90600
SIP: Record-Route handling on outbound subscriptions
| Status | Closed |
| Id | 76831 |
Processing of Record-Route in SUBSCRIBE response.
Simply missing.
phone: Pickup fuction key is not displayed while all alerting calls are displayed on a Partner key
| Status | Closed |
| Id | 77393 |
alerting calls displayed on a Partner key are not displayed on the Pickup key.
if nothing has to be displayed on the pickup key the key should not disappear but display the 'idle' label
Status:
checked in to 10.00, 9.00, 90600
H.323: Media Negotiation problem with transfer in Gateway (not PBX)
| Status | Closed |
| Id | 77683 |
Under special conditions a blind transfer happend in the Gateway could result in a call without media. This only happened if the call was transfered twice and the destination of the first transfer was a physical interface.
PBX Twinning: When calling another (twin) phone, the call was sent to the original phone also
| Status | Closed |
| Id | 77829 |
It still works, but this waiting call could be confusing
PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX
| Status | Closed |
| Id | 77874 |
A call from a object within a node with escapes on a slave PBX was not routed to the master if the destination was within the same node and not known on this slave but was sent to the node-extern destination directly
PBX Waiting: Name Id missing in calls initiated with SOAP
| Status | Closed |
| Id | 77911 |
When a Waiting Queue is used by applications to initiate outgoing calls, the name of the waiting queue should be sent with these calls as calling name. This name id was missing
RTP: Potential random trap when closing channels
| Status | Closed |
| Id | 77918 |
Happens if there is a collision with a received packet and closing of the channel. Window for this is very small, so it should happen very rarely. Probability can increase with high load.
Web-UI: Visually separated commands
| Status | Closed |
| Id | 77954 |
Visually separated commands 'Clear' and 'Save' on Maintenance/diagnostics/Events.
Also 'download' and 'delete' on General/License.
IP241,IP222,IP232: Show special symbol for call completion entries in call list
| Status | Closed |
| Id | 77966 |
Show special symbol for call completion entries in call list to separate from usual missed calls.
IP241,IP222,IP232: Status message "Recording" not displayed
| Status | Closed |
| Id | 77992 |
Status message "Recording" needs to be displayed during recording.
Gateway: Wrong Media info sent for calls forwarded from a VOIP interface to another VOIP interface
| Status | Closed |
| Id | 78024 |
This could cause wrong information to be displayed on a PBX calls page. Esspecially SRTP was not indicated.
Better to not send any Media Info in this case.
IP241,IP222,IP232: Wrong display name in call list
| Status | Closed |
| Id | 78047 |
Name identification of remote party is to be displayed,
but name identification of diverting party was displayed instead
on incoming calls list.
phone: the call which was the active call at start of a conference was not automatically cleared on a release from remote
| Status | Closed |
| Id | 78086 |
The call which was the active call at start of a conference had to be cleared manually but the call which was on hold at start of conference was cleared automatically.
Now any call will be automatically cleared when relesed from remote.
Status:
checked in to 10.00, 9.00, 90600
phone_orchid: remaining call mute after remote relase for the call which was the active call at start of a conference
| Status | Closed |
| Id | 78091 |
did not happen after a remote release for the call which was on hold at start of the conference. Status: checked in to 10.00, 9.00
phone: ip222,ip232: Muting USB headset micro via microphone key did not work but MICROPHONE OFF was displayed
| Status | Closed |
| Id | 78159 |
. Status: checked in to 10.00, 9.00, 90600
IP241,IP222,IP232: Trap in font rendering
| Status | Closed |
| Id | 78210 |
Some characters at beginning of line may cause trap.
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
phone: ip241: memory leak when ip230x extension module is attached and used
| Status | Closed |
| Id | 78311 |
one packet lost per keystroke on extension module
PBX: Dynamic group function keys did not work with non-ascii characters in group name
| Status | Closed |
| Id | 78322 |
Name was not correctly converted to internal utf-8 representation
phone: ip222,ip232: trace USB media data only if explicitely requested
| Status | Closed |
| Id | 78326 |
use "config add USB-HOST AUDIO /trace" if data is needed
H.323: Potential Trap in case of special malformed RAS registration
| Status | Closed |
| Id | 78368 |
A NULL pointer access could happen
SIP: Removed b=TIAS attribute from media description
| Status | Closed |
| Id | 78379 |
Causes trouble on some SIP providers (neotel.at)
DHCP server leases with hostnames containing non-ascii latin1 characters (for example Umlauts) could not be displayed
| Status | Closed |
| Id | 78382 |
In pre-V9 firmware hostnames were stored latin1-encoded. Names contaning non-ascii latin1 charaters must be converted to UTF8 before display. Status: checked in to 10.00, 9.00, 90600
MyPBX: Changing password did not work any more
| Status | Closed |
| Id | 78391 |
Collateral damage of fix #75453: PBX/IP-DECT: User password length
phone: ip222, ip232: USB headset sometimes mute after disconnect
| Status | Closed |
| Id | 78414 |
depending on isochronous transfer state at disconnect time. seems to happen more often with relatively cheap headsets.
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.
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer
| Status | Closed |
| Id | 78487 |
channel was reconfigured on remote switch to T38, without a new activate fax relay command Status: ac_dsp3.cpp
PBX Boolean: Avoid unnecessary load by boolean monitoring
| Status | Closed |
| Id | 78504 |
A call was sent to the monitoring endpoint every 10s
IP241,IP222,IP232: No RTP send on outgoing call in some cases
| Status | Closed |
| Id | 78550 |
No RTP send on outgoing call in some cases
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.
PBX-SOAP: Limit on size for Admin call too small
| Status | Closed |
| Id | 78592 |
Only 4k was allowed
SoftwarePhone: Default ToS value for RTP
| Status | Closed |
| Id | 78593 |
Now the default ToS value for RTP packets is 0xB8.
PBX Filter: Maximum length increased from 13 to 17 digits
| Status | Closed |
| Id | 78594 |
13 digits could be too little if an international number including a prefix of an external line was to be checked
IP241,IP222,IP232: Wrong call direction indication on PARTNER fkey
| Status | Closed |
| Id | 78642 |
When partner calls someone else it my displayed as if partner is called.
(Only if option "Show connected Party in busy State" is enabled)
IP222 IP232 IP241: Sporadic noisy sidetone
| Status | Closed |
| Id | 78649 |
Analog and digital sidetonegain were enabled,now only the analog sidetonegain is used. Status: ac_codec3.cpp
Gateway: Handling of blind transfer gets stuck in routing
| Status | Closed |
| Id | 78709 |
Routing of transfer call stops with: reason='incomplete'
But during transfer no dialing digits will follow.
V9 Hotfix 10 (90600.12)
Changes included in Version 9 hotfix10 Definition
New Features
Linux: Enable/Disable configuration
| Status | Closed |
| Id | 78836 |
With the new feature #78786 the configuration is only saved in the downloaded configuration file with password. Now the information is also included in the configuration file with standard password and in the file downloaded by the update server.
Bug Fixes
PBX: Trap if enabling Unknown Registrations
| Status | Closed |
| Id | 78904 |
Collateral Damage from
fix: #77261: Phone->PBX LDAP Search returns normalised number to be dialled by phone
V9 Hotfix 11 (90600.14)
Changes included in Version 9 hotfix11 Definition
New Features
HTTP-Client: MD5-sess authentication
| Status | Closed |
| Id | 77773 |
HTTP Digest Authentication with alogrithm=MD5-sess.
Choose the first supported "WWW-Authenticate" line from 401 response headers.
Needed for new versions of IIS.
Status:
http://wiki.innovaphone.com/index.php?title=Support:DVL-Feature_Requests#HTTP_Client
Phone: Possibility to reject incoming SIP calls with customized reason phrase
| Status | Closed |
| Id | 77928 |
E.g. "SIP/2.0 480 Do not disturb" instead of "SIP/2.0 480 Temporarily unavailable"
Active Directory Replication: Editfield for Poll Timer added
| Status | Closed |
| Id | 78631 |
If change notifications cannot be received from an AD, a poll timer can be specified. A re-replication is going to take place after the poll timer expired.
X.509: Support for PKCS#12 files
| Status | Closed |
| Id | 78820 |
Support for certificate import using password encrypted PKCS#12 files.
Currently the following encryption types are supported:
pbeWithSHAAnd128BitRC4 (1.2.840.113549.1.12.1.1)
pbeWithSHAAnd40BitRC4 (1.2.840.113549.1.12.1.2)
pbeWithSHAAnd3-KeyTripleDES-CBC (1.2.840.113549.1.12.1.3)
pbeWithSHAAnd128BitRC2-CBC (1.2.840.113549.1.12.1.5)
pbewithSHAAnd40BitRC2-CBC (1.2.840.113549.1.12.1.6)
myPBX: Pass language code to reporting
| Status | Closed |
| Id | 78874 |
For localization of the call list.
Phone: Show number of voice messages on label of MWI fkey
| Status | Closed |
| Id | 78894 |
Use "%u Message(s)" as label text for active state of MWI fkey.
IP-DECT: Cisco SIP conferencing
| Status | Closed |
| Id | 78988 |
Cisco proprietary softkey events implemeneted to start a 3pty conference
IP-DECT: Feature Code for calling predefined service URIs
| Status | Closed |
| Id | 79028 |
User for Cisco compatibility
Cisco Softkey Features, Status updates and Park
| Status | Closed |
| Id | 79030 |
suported_mask needs more flags
Softkey Feature Park
IP-DECT: Cisco Park
| Status | Closed |
| Id | 79032 |
Send Park softkey event
| Status | Closed |
| Id | 79036 |
.
IP-DECT: SRTP for an OEM device
| Status | Closed |
| Id | 79061 |
Now SRTP is configurable for an OEM device.
| Status | Closed |
| Id | 79104 |
.
Merge of v10 changes: Dynamic Timer Management/oem serial no
| Status | Closed |
| Id | 79127 |
Better power management on some platforms
Display of oem serial no
stanard behaviour of delete on NULL
Merge of v10 changes: Memory Management
| Status | Closed |
| Id | 79129 |
for OEM needs
Merge of v10 changes: Firmware File check when updating improved
| Status | Closed |
| Id | 79130 |
for OEM needs
Merge of v10 changes: More ciphers
| Status | Closed |
| Id | 79134 |
for OEM SRTP and other Certificate formats
Merge of v10 changes: Internal interfaces enhanced
| Status | Closed |
| Id | 79135 |
for OEM support
Merge of v10 changes: ZipZip Tone definition updates
| Status | Closed |
| Id | 79136 |
should overlay normal voice
Merge of v10 changes: Library update
| Status | Closed |
| Id | 79140 |
.
Merge of v10 changes: permit DHCP mode change between client/disabled without reset
| Status | Closed |
| Id | 79152 |
for OEM support
ASN.1 BER: Support for indefinite-length encoding
| Status | Closed |
| Id | 79260 |
needed for PKCS#12 import
only decoding
ASN.1 BER: Support for constructed octet strings
| Status | Closed |
| Id | 79261 |
needed for PKCS#12
only decoding, only one level of nesting
IP810: Config Wizard
| Status | Closed |
| Id | 79284 |
Mechanism needed to add platform specific files (IP810 uses same firmware as IP6010, IP3010, ...)
Merge of v10 changes: Support for SIP specific phone features
| Status | Closed |
| Id | 79306 |
for OEM support Status: checked in to 9.00
phone: permit to select the directories to be searched on indirect dialing
| Status | Closed |
| Id | 79307 |
if there is for example a huge external directory used for inbound name resolution and dialing is restricted to internal partners it may be hard to find internal numbers via combined directory search.
config add PHONE DIR-UI /dial-mask <selection>
selection:
0x02 - local directory
0x04 - PBX directory
0x08 - external directory
0x06 = local + PBX
0x0E = local + PBX + external
IP-DECT: Signal waiting calls to handset
| Status | Closed |
| Id | 79349 |
Was only acoustic information.
IP-DECT: LDAP phonebook (IP1202)
| Status | Closed |
| Id | 79483 |
The new feature LDAP phone book for the IP1202 is added now.
IP-DECT: Three party conference with innovaphone PBX
| Status | Closed |
| Id | 79503 |
Now it is possible to make a three party conference with DECT handsets with an innovaphone PBX (an innovaphone device with the CONF interface). The conferencing unit must be configured in the DECT master. The conference call is established with the feature code 'R' + '3'.
This fix also includes a rework of the DECT radio module. It can handle more than one waiting or hold call now.
Gateway: 'cn' attribute for test interfaces
| Status | Closed |
| Id | 79506 |
The 'cn' attribute is included in the test interfaces SIG0/1 for internal tests now.
Bug Fixes
IP222/IP232/IP241: Ethernet link configuration
| Status | Closed |
| Id | 78067 |
In some case ethernet link configuration and display didnt work.
Now the autoneg status is shown if available, and the phy status if no autoneg didnt complete.
An option to use autoneg with fixed speeds and a link trace option (/ltrace) are added.
Status:
orchid_drv.cpp
orchid_drv.h
orchid_drv.xsl
Media Relay: Don't send dummy RTP data on incoming calls
| Status | Closed |
| Id | 78252 |
Dummy RTP data is sent just in case a NAT router is within the media path to set a UDP mapping in case both legs of the call contain a NAT router.
An example for such a situation is a call coming from a SIP provider thru a NAT router to the PBX, which forwards the call back out to the SIP provider. The NAT router won't get RTP data from inside to set the mappings.
The dummy RTP was sent to all legs of the call, but it is better to send it to outgoing call legs only, because endpoints calling in may turn off a local ringback tone when receiving dummy RTP
H.323/SIP: Avoid delayed SDP within outgoing calls as far as possible
| Status | Closed |
| Id | 78466 |
If media renegotiation is needed, to one side of the call an request for a media proposal (in SIP terms, this is an INVITE without SDP) is sent. The media proposal (in SIP terms SDP offer) is then forwarded to the other side.
The request for an offer should if possible not sent with the initial call, because there is equipment which does not handle 'delayed SDP'
Media Relay: Don't terminate T.38 protocol in media relay, forward transparently
| Status | Closed |
| Id | 78610 |
This should add robustness and reduce CPU load
Flash Directory: Display Error Messages
| Status | Closed |
| Id | 78656 |
Flash Directory: Display Error Messages
Shedding light on replication problems.
currently only in 10.00
RTP: No check for changed media destination because of received multicast
| Status | Closed |
| Id | 78698 |
When receiving RTP normally a check is done if the source of the RTP is the same as we are sending to. If this is not the case, we assume the destination of the RTP is behind a NAT router and we change the destination address to the source address of the received RTP.
This does not make sense for received multicast.
IP22 IP24 IP28 IP305: Faxempfang gelegentlich gestört
| Status | Closed |
| Id | 78803 |
DSP erhält zum falschen Zeitpunkt ein close-rtp. Status: ac_dsp3.cpp
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax receive max packet size increased to 1024
| Status | Closed |
| Id | 78811 |
For internet fax devices Status: ac_fax3.cpp
phone: ip222, ip232: USB controller sometimes hangs
| Status | Closed |
| Id | 78824 |
happened with certain headsets after fast connect/disconnect sequences
Status:
checked in to 10.00, 9.00
merged to 90600
SIP: SDP answer sometimes contains 2 media descriptions for audio
| Status | Closed |
| Id | 78833 |
... instead of one for audio and one for video.
Offer:
v=0
o=- 1 2 IN IP4 130.30.1.111
s=CounterPath Bria
c=IN IP4 130.30.1.111
t=0 0
m=audio 1902 RTP/AVP 0 8 18
a=alt:1 1 : 1A98X9/s 31UwK5IA 130.30.1. 111 1902
a=fmtp:18 annexb=yes
a=rtpmap:18 G729/8000
a=sendrecv
a=x-rtp-session-id:ACE43888C33A4AC8895C7F8F69380914
m=video 28212 RTP/AVP 115 34 123 124
a=alt:1 1 : yjz/mbyQ aWGuD8xJ 130.30.1. 111 28212
a=fmtp:115 QCIF=1;CIF=1;I=1;J=1;T=1
a=fmtp:34 QCIF=1;CIF=1
a=fmtp:123 profile-level-id=42801e; packetization-mode=0; max-mbps=48600
a=fmtp:124 profile-level-id=42801e; packetization-mode=1; max-mbps=48600
a=rtpmap:115 H263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:123 H264/90000
a=rtpmap:124 H264/90000
a=sendrecv
a=x-rtp-session-id:694677A9F65D419FB7B9E451FFCA02CC
Answer:
v=0
o=- 3141 1 IN IP4 10.230.5.1
s=-
c=IN IP4 10.230.5.1
t=0 0
m=audio 16410 RTP/AVP 0 13
b=TIAS:64000
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
m=audio 16410 RTP/AVP 0 13
b=TIAS:64000
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
Conference: Permanent noise in conference channel
| Status | Closed |
| Id | 78838 |
Permanent noise in conference channel.
Increasing with every new participant.
PBX: When doing a show users, unnecessary data was sent to the browser
| Status | Closed |
| Id | 78878 |
The data could be reduced and thus speeding up the display of big users lists.
PBX Admin UI: Editing profiles was not possible if the user had too many contacts
| Status | Closed |
| Id | 78883 |
The form is now submitted using POST instead of GET.
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.
SIP: Send even anonymous Diversion header
| Status | Closed |
| Id | 78954 |
If diverting party's identity is unknown/hidden.
Diversion: ;reason=user-busy
SIP: Failed to decode presence XML from CUCM
| Status | Closed |
| Id | 78967 |
Support for namespace "urn:ietf:params:xml:ns:pidf:status:rpid" added.
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer #2
| Status | Closed |
| Id | 78969 |
tone flag was not disabled Status: ac_dsp3.cpp
H.323: Media problem with SIP call-completion on IP-DECT
| Status | Closed |
| Id | 78970 |
There was no media on the successful completion call (SDP recived in Progress was not forwarded)
SIP: Send 'Connected Number' in P-Asserted-Identity header of 200/OK
| Status | Closed |
| Id | 78974 |
Send 'Connected Number' in P-Asserted-Identity header of 200/OK
if different from original called number.
SIP: Re-transmission of ACK with SDP answer
| Status | Closed |
| Id | 78975 |
Handling of re-transmissions of 200/OK with SDP offer.
ACK must contain SDP answer.
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: DSP packet debug didnt show some packets, version endian was wrong
| Status | Closed |
| Id | 78985 |
cleanup
Status:
ac_491.cpp
ac_494.cpp
IP-DECT: Radio/Master calls switched to slowstart
| Status | Closed |
| Id | 78987 |
If first media answer was received with a PROGRESS message, the call leg from the radio to the master was switched to progress (no EFC features are defined for the Progress message).
This could cause media problems later in the call with hold/retrieve/transfer.
SIP: REFER does not work as expected on Gateway interfaces without registration
| Status | Closed |
| Id | 79008 |
Bad side effect of a previous fix in HF8:
http://mantis.innovaphone.com/view.php?id=76309
Introducing new SIP interop tweak "Take Refer-To URI as Remote Target URI"
to be found on the Gateway interface config (GWx).
See http://mantis.innovaphone.com/file_download.php?file_id=66299&type=bug
IP-DECT: Better handling when receiving a call, while a previous call is being released
| Status | Closed |
| Id | 79022 |
Call to service number, which cause a callback can create problems.
Edss1 Interworking: Sending divertingLegInformation2 as PublicPartyNumber(was UnknownPartyNumber)
| Status | Closed |
| Id | 79025 |
Such that the type of number resulting from an interface cgpn-map can also be asn.1 encoded.
Gateway: Mapping of Message Center Number of MWI
| Status | Closed |
| Id | 79039 |
Mapping of Message Center Number of MWI according to CGPN mapping.
Qsig Interworking: Apply interface CGPN map to DGPN
| Status | Closed |
| Id | 79060 |
Qsig Interworking: Apply interface CGPN map to DGPN
PBX: A CFB at a PBX object was executed on call to busy user
| Status | Closed |
| Id | 79199 |
It should only be executed, if the calls exceeded the Busy On number.
PBX: Mapping of Message Center Number of MWI
| Status | Closed |
| Id | 79232 |
Mapping of Message Center Number of MWI.
PBX-SOAP: Conference Id was missing in call-info for parked calls
| Status | Closed |
| Id | 79243 |
This caused problems with the innovaphone operator
PBX-SOAP: UserClear cause should not be used for remote party
| Status | Closed |
| Id | 79249 |
This can result in strange call clearing without tones
PBX: Include error in resulting XML if submit-object with insufficient rights was attempted
| Status | Closed |
| Id | 79289 |
Problem with innovaphone operator.
phone: ip222,ip232,ip241: wrong AM/PM time display on status line
| Status | Closed |
| Id | 79302 |
15:19 PM was displayed instead of 03:19 PM Status: checked in to 10.00, 9.00, 90600
Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0))
| Status | Closed |
| Id | 79303 |
Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0))
May from now on be overriden by variable "$_pbxmwiservice".
PBX: Local objects could not be called from Nodes with escapes as expected
| Status | Closed |
| Id | 79317 |
Was not possible to call at all, or the number was wrong
PBX-Trunk: Handling for Incomplete, Invalid, Busy, No Anser destinations for Media calls only
| Status | Closed |
| Id | 79319 |
For example this should not be done for presence/dialog subscriptions
Refactored some ASN.1 BER Handling
| Status | Closed |
| Id | 79357 |
Became necessary in the run of: #79260: ASN.1 BER: Support for indefinite-length encoding
PBX: Conference & VM
| Status | Closed |
| Id | 79359 |
The last connected user is not disconnected although it is configured. It occurs if a VM PBX object forwards the call to the BC Conference object (like the innovaphone conferencing script). It is fixed now.
IP-DECT: Subscription could get lost randomly with logout/login cycle
| Status | Closed |
| Id | 79397 |
Problem with user database handling
H.323: Media problem if PBX rtp-proxy is activated for CFNR to external destination
| Status | Closed |
| Id | 79430 |
no voice
HTTP: Could not have sockets with same ports but different remote addresses
| Status | Closed |
| Id | 79458 |
This was a general problem with the new TCP stack, used together with IP6. Problem only happened for HTTP because only for HTTP this new stack is currently used.
ISDN: Fix for call completion interworking
| Status | Closed |
| Id | 79498 |
CCBS/CCNR does not work in some cases
SIP: From-URI may got lost after call transfer
| Status | Closed |
| Id | 79536 |
UPDATE may be send out missing From-URI.
AD Replication stuck after connection loss
| Status | Closed |
| Id | 79541 |
Didn't reconnect. Side effect of after-hf10 fix
PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object
| Status | Closed |
| Id | 79549 |
If the CFNR was executed, the CDR for the call to the Waiting Queue/Broadcast object ended without release.
PBX: Web User interface problem with IE and non-ascii PBX names
| Status | Closed |
| Id | 79568 |
For example groups at a user assigned to a PBX with non-ascii characters could not be edited.
The problem is a bug in IE XSL translation which does special handling of href attributes. Same thing with onclick attribute works.
ip3010 TEL port not working in NT mode
| Status | Closed |
| Id | 79570 |
TEL port of ip3010 gateways configured in NT mode do not get Physical Link up indication. This problem applies to V9hotfix7 up to V9hotfix10. Status: ip6010.cpp
SIP: Max forward value of 32 could be too small for some provider
| Status | Closed |
| Id | 79578 |
For a starting value of max-forwards a value of 32 was used, because this is the maximum value in H.323. This was too small for some sip providers. Starting value now increased to 64 and on H.323 the half value is transmitted.
SIP: Re-negotiation for T38 did not work in media-relay scenarios
| Status | Closed |
| Id | 79583 |
Bad SDP answer was generated.
Status:
Introduced by
#77277: SIP: SDP answer must have the same number of media descriptions as received offer
V9 Hotfix 12 (9061009)
Changes included in Version 9 hotfix12 Definition
New Features
myPBX: Syslog for sessions and application sharing
| Status | Closed |
| Id | 77289 |
Syslog entries should be useful for debugging session timeouts and problems with the external application sharing solution.
SIP: Pass display names of <dialog-info> to phoneapp
| Status | Closed |
| Id | 79543 |
Pass display names of <dialog-info> to phoneapp.
Gateway: Overlap dialing timeout configurable
| Status | Closed |
| Id | 79639 |
The default of 4s is not good for all applications. Now configurable between 500ms and 6000ms.
IP-DECT: OEM Configuration of Cisco Features changed
| Status | Closed |
| Id | 79684 |
should depend on installed license
myPBX launcher: Automatically move main window into visible area of screen
| Status | Closed |
| Id | 79697 |
The window might be outside the visible area if the user changed the screen resolution or disconnectes one of the screens.
If the main window is not visible on any screen, it is now moved to the center of the primary screen, when the user clicks the tray icon.
PBX-SOAP: FindUser optional argument 'nohide'
| Status | Closed |
| Id | 79734 |
Needed if the FindUser is used to look for users which have the 'Hide from LDAP property' set. Status:
PBX Mobility: Pick mobile call on fixed phone
| Status | Closed |
| Id | 79794 |
by dialing number of Mobility object on fixed phone
Gateway: Allow setting of system time from ISDN time
| Status | Closed |
| Id | 79889 |
In ISDN connect messages a public network is sending a local date and time. By a configuration option this can now be used to update the system time
IP-DECT: New Master/Radio behavior for license incompatibility
| Status | Closed |
| Id | 79914 |
The behavior of the DECT Master is changed, if an unlicensed Radio tries to register in. This is only used for an OEM license model.
PBX: Better handling of presence information without activity
| Status | Closed |
| Id | 80006 |
This type of presence information could contain a note which refers to the future
RPCAP trace: Indicate transmit/received packets by setting the remote mac address to 00-90-33-00-00-00
| Status | Closed |
| Id | 80152 |
When reading a trace it is currently not obvious if a packet is sent or received, we need to find out the devices IP address, e.g. by reading the config. If the devices on MAC adress is used a source only if a packet is sent and as destination only if a packet is received this process is simplified.
This is now done for UDP/TCP traces!
SIP: Support for Alert-info:<Bellcore-dr1>
| Status | Closed |
| Id | 80174 |
Proprietary tagging of internal calls
phone: ip222,ip232: more USB headsets supported
| Status | Closed |
| Id | 80224 |
for a complete list see http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset
Status:
checked in to 10.00, 9.00
Phones: Switch for phoneapp to disable auto-answer
| Status | Closed |
| Id | 80233 |
Disable/enable auto-answer support on phoneapp level.
PBX: DECT attributes for DECT security
| Status | Closed |
| Id | 80300 |
In preparation for the new DECT feature DECT security there will be new attributes for the endpoint data which must be taken over. With this fix the innovaphone PBX supports the new attributes if the user is edited.
phone: ip222, ip232: Jabra USB Headset feature "Reject incoming call" supported now
| Status | Closed |
| Id | 80401 |
A double tap on talk button rejects a ringing call with 'busy'
IP-DECT: Signal waiting calls to handset
| Status | Closed |
| Id | 80408 |
New event to the DECT system to add a waiting call to the call list.
This is step two of the implementation and related to the case #79349.
IP-DECT: Anonymous endpoint information
| Status | Closed |
| Id | 80421 |
Information about anonymous endpoint is added to the event to the DECT system. In preparation for the new feature "easy subscription".
IP-DECT: DECT security
| Status | Closed |
| Id | 80424 |
Some changes in preparation for the new feature "DECT security". This feature will be available later.
Phone: Accept MWI from Exchange Server
| Status | Closed |
| Id | 80446 |
MS Exchange Server sends unsolicited NOTIFY(message-summary) to served user
with served user's number as destination and origin.
But phones expect to receive MWI message center number as origin.
MWI fkey would not light up.
SIP: New config option /take-zero-addr-for-hold
| Status | Closed |
| Id | 80516 |
From now on "c=IN IP4 0.0.0.0" is no longer accepted as hold signaling.
Config option /take-zero-addr-for-hold is introduced to get back old handling.
IP-DECT: User log in, endpoint data added
| Status | Closed |
| Id | 80620 |
During the user log in with a change of the IP-DECT Master the endpoint data (product id/software version) is sent to this new one now.
SIP: Pass display information to application
| Status | Closed |
| Id | 80632 |
Pass display information received in Call-Info header in 200/OK for BYE to app.
IP-DECT: Idle display update
| Status | Closed |
| Id | 80654 |
Handling of idle display update message in the call release message is added. Used with an OEM PBX.
myPBX launcher: Configurable hotkey action
| Status | Closed |
| Id | 80684 |
Options are:
- Copy selected phone number to myPBX
- Show myPBX
myPBX launcher: Autocomplete configured URLs
| Status | Closed |
| Id | 80689 |
If the user configures just an IP address, it shall be replaced by the full myPBX URL.
For example 192.168.0.10 will be replaced by http://192.168.0.10/PBX0/MY/client.htm.
Bug Fixes
PBX: CFNR at Gateway object with incomplete destination
| Status | Closed |
| Id | 79605 |
A CFNR at a gateway object is executed if there is no registration. Any additional digits dialed should be added to the CFNR destination. This did not work if the original CFNR destination was incomplete and only completed with additional digits dialed.
PBX Waiting: Filter did not work anymore for CFNR
| Status | Closed |
| Id | 79610 |
Collateral damage from
fix: #75465: PBX: CFNR Loop check detected loops that weren't
LDAP Server caused crash on port-scanner attack
| Status | Closed |
| Id | 79634 |
LDAP Server caused crash on port-scanner attack
myPBX: Quotes in connected names caused java script errors
| Status | Closed |
| Id | 79636 |
When the connected name of a call contained the ' character there was a script error.
PBX: Status displayed on boolean function key could be wrong
| Status | Closed |
| Id | 79643 |
In case of severe network problems, it could happen that the status displayed on a Boolean function key was wrong and was only corrected when the boolean status changed.
DHCP Server Identifier was cleared after editing the DHCP-Server page
| Status | Closed |
| Id | 79651 |
The value of "IP4/ETHx/DHCP/Server Identifier" was cleared when the OK or Renew button was pressed on the "IP4/ETHx/DHCP-Server" page. This bug was introduced with V9hotfix5.
SIP: Wrong Contact-URI in outgoing NOTIFY
| Status | Closed |
| Id | 79700 |
Contact-URI should match the Request-URI of the SUBSCRIBE.
Also the Message-Account URI in "simple-message-summary" was wrong as result of the wrong Contact-URI.
Config: Could not dynamically set or reset /trace on the LICENSE module
| Status | Closed |
| Id | 79709 |
Missing handling of MODULE_UPDATE.
PBX: Allow call completion only if recall is not prohibited by CFU
| Status | Closed |
| Id | 79739 |
A CFU loop results in a rejection with busy. A subsequent call completion attempt was allowed and a recall possible was signaled right away. This was very confusing.
SIP: Handling of Call Pickup
| Status | Closed |
| Id | 79741 |
Failed to handle INVITE with Replaces as Call Pickup.
Must redirect Pickup call to alerting party.
myPBX: Problem with checking browser capabilities in Firefox
| Status | Closed |
| Id | 79757 |
When checking the supported browser features, Firefox thows an uncaught exception if cookies are deactivatd by the user. Therefore myPBX is stuck in the "Loading" screen instead of displaying a configuration hint.
IP241,IP222,IP232: Show lengthy number information on Partner fkey
| Status | Closed |
| Id | 79779 |
Toggle between 'show head' and 'show tail' presentation.
SIP: DNS resolving of STUN server failed
| Status | Closed |
| Id | 79788 |
Wrong STUN server port used (5060 instead of real port).
Fax: Channel memory leak
| Status | Closed |
| Id | 79815 |
Memory leak is fixed in fax channel now.
PBX: Export to CSV, utf-8 byte order mark was missing
| Status | Closed |
| Id | 79844 |
Without the utf-8 BOM the file was not correctly interpreted as utf-8 by Microsoft Excel and other applications.
PBX: For registrations containing name and number, use name to identify the device
| Status | Closed |
| Id | 79846 |
It did depend on the sequence of the name and the number. If the number was first, the first device was selected and the name was ignored. Now the name is used to select the device regardless of sequence.
This is a problem with endpoints which always send name and number for registration.
SIP: CLEARMODE does not work since Hotfix10
| Status | Closed |
| Id | 79858 |
Invalid SDP answer is send:
v=0
o=- 4 3 IN IP4 10.28.108.8
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 125 101
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
memory violation in http client when processing an URI > 8000 byte
| Status | Closed |
| Id | 79860 |
happens when the digest authentication header is constructed
Status:
checked in to 10.00, 9.00
PBX Broadcast: Forking/Mobility configured at Broadcast object did not work
| Status | Closed |
| Id | 79863 |
Mobility can be used now for forking with delay
Potential trap in packet libraray
| Status | Closed |
| Id | 79866 |
Ther was a not obvious size limit for packet::put_head and packet::put_tail, which caused a trap if the size was exceeded. This could happen if a CDR exceeded a certain size.
SIP: Domain Name System (DNS) names compared case sensitive
| Status | Closed |
| Id | 79887 |
Domain Name System (DNS) names are "case insensitive"
SIP: Interworking with "AUDC-IPPhone"
| Status | Closed |
| Id | 79892 |
RFC-3551 4.5.2
Even though the actual sampling rate for G.722 audio is 16,000 Hz,
the RTP clock rate for the G722 payload format is 8,000 Hz because
that value was erroneously assigned in RFC 1890 and must remain
unchanged for backward compatibility. The octet rate or sample-pair
rate is 8,000 Hz.
Codec description must be: a=rtpmap:9 g722/8000
but "AUDC-IPPhone" sends: a=rtpmap:9 G722/16000
in SDP offer.
SIP: Problems with DNS resolving of proxy adresses
| Status | Closed |
| Id | 79907 |
If resolving of the primary proxy failes, the backup proxy is never resolved.
IP-DECT: Reset required if Radio password changed
| Status | Closed |
| Id | 79929 |
Now reset required is shown if the Radio password for the Master registration is changed.
Phone: PBX (operator) initiated outbound call was connected but mute when another call was ringing on phone already
| Status | Closed |
| Id | 79943 |
the phone was not switched from ring mode to handsfree/headset mode
SIP: Interworking with HD audio client
| Status | Closed |
| Id | 79953 |
Support for unknown codecs.
Passing transparentley.
Phone: Out-Of-Memory-Trap
| Status | Closed |
| Id | 79980 |
If LOG server is configured but not reachable
the device will buffer arising LOG entries until a limit of 300kB.
This limit was to high for old black/white telephones.
Now not more then 1% of DRAM size is used for LOG buffer.
PBX: When editing a Node object it was changed to a PBX object
| Status | Closed |
| Id | 79982 |
This was a collateral damage of fix: #78878: PBX: When doing a show users, unnecessary data was sent to the browser
ISDN: Problem configuring negative volume levels
| Status | Closed |
| Id | 79992 |
Collateral Damage from
79028: IP-DECT: Feature Code for calling predefined service URIs
Dect: AD replication makes IPEI disappearing on all users
| Status | Closed |
| Id | 80015 |
Side effect of a previous fix #72672 aiming at ad-replicated and deleted objects coming back to life.
myPBX: Setting CFx with boolean did not work in IE8
| Status | Closed |
| Id | 80018 |
Using Internet Explorer 8 the selected Boolean object was not saved.
Internet Explorer 8 requires option tags to have a value attribute.
IP-DECT: Handover with IP1202
| Status | Closed |
| Id | 80022 |
Only one handover is possible since the last fix with the IP1202. It is fixed now.
Name and Number configured for a registration must be completely replaced by registration result
| Status | Closed |
| Id | 80039 |
If for example Number and Name is configured but the registration result provides the Number only, the configured Name must not be used in further signaling operations (diversion queries ...)
Status:
checked in to 10.00, 9.00
PBX: Possible trap on calls from misconfigured nodes/PBXs (node parent loop)
| Status | Closed |
| Id | 80093 |
If a node or PBX is configured with a parent node configured to itself in the most simple case, a call from an endpoint configured for this node to a destination which cannot be found in this node, will cause a trap.
This is a collateral damage of fix: #79317: PBX: Local objects could not be called from Nodes with escapes as expected
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305 IP222 IP232 IP241: receveid RTP packets limited to 480bytes
| Status | Closed |
| Id | 80111 |
increased to 640 Bytes to allow G711 80ms.
DHCP Survivability mode doesn't work after a DHCP restart
| Status | Closed |
| Id | 80112 |
``Survivability´´ mode is used by WLAN phones. In this mode the settings of the 'saved lease' (IP addr, mask, ...) are used until a fresh lease is received. The 'saved lease' is the last lease received from a server, it is kept over a reboot.
When WLAN coverage is lost for a while and then regained a DHCP restart is requested to get a fresh lease from a server in a possibly different network. If this happened while using the 'saved lease' the phone lost it's (saved) IP address.
Status:
checked in to 10.00, 9.00
AC-DSP3: Switch trace off if the DSP Host interface shows an error
| Status | Closed |
| Id | 80130 |
SIP: Keep backup registration while calls are active
| Status | Closed |
| Id | 80137 |
Terminating backup registration will active calls (through backup system) get disconnected.
SIP: Content of Allow-Events header must be treated case-insensitive
| Status | Closed |
| Id | 80173 |
Event names are case insensitive.
IP-DECT/Analog Features: Call Park should be done on the last active call
| Status | Closed |
| Id | 80209 |
Szenario is an active call, then a waiting call comes in, which is accepted, then call park is executed. This call park should be done on the accepted waiting call and not the original.
IP-DECT/Analog Features: Call completion
| Status | Closed |
| Id | 80212 |
The type of the call is changed back to normal state if the call completion is executed, and facility conversion is added for the call completion state.
This fixes the reusing of features for a call completion callback call, used if IP-DECT/analog features are enabled. This also fixes missed remote hold and retrieve events to the gatekeeper.
SIP: Handling of 488 for encrypted media
| Status | Closed |
| Id | 80246 |
If SRTP offer is rejected by remote endpoint with "SIP/2.0 488 Not Acceptable Here"
we should re-try offer without encryption keys.
Phones: Show 'tel' presence on configuration screen
| Status | Closed |
| Id | 80274 |
Show latest presence on 'Presence' fkey,
but show 'tel' presence on phone config menu.
PBX-Waiting: Calls to a Waiting queue object in altering state could not be cleared with SOAP
| Status | Closed |
| Id | 80291 |
A SOAP UserClear issued for the call monitored at the Waiting Queue object didn't do anything
Phones: Sometime fkey could not be deleted
| Status | Closed |
| Id | 80389 |
Sometime fkey could not be deleted when using the fast edit mode (long fkey press).
IP-DECT: Handset display
| Status | Closed |
| Id | 80402 |
Names with special characters are not correctly shown in the handsets. This is fixed in the IP1200 now.
IP210 IP230 IP240: Handset receiver volume increased, especially at lower frequencies
| Status | Closed |
| Id | 80403 |
For better performance with jinlida receiver.
Gateway: Pass through ctSetup facility
| Status | Closed |
| Id | 80437 |
Lync sends diverting party information inside Referred-By header.
Referred-By is interworked to ctSetup facility.
ctSetup facility needs to be passed through by Gateway application.
Linux: IP address with external DHCP server
| Status | Closed |
| Id | 80471 |
Assigning a IP address to Linux by a external DHCP server is not working if the network interface which is used is configured with a fix IP address (DHCP disabled). This is fixed now.
IP232: Invalid text on second line of fkey label
| Status | Closed |
| Id | 80484 |
Invalid text on second line of fkey label when changing registrations.
PBX: Routing problem from sub-slave to master to object in sub-slave node, but registered to master
| Status | Closed |
| Id | 80591 |
Collateral damage from #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX
SIP: Calls may remain in clearing state
| Status | Closed |
| Id | 80623 |
SIP calls may remains undeleted.
SIP: Code optimization
| Status | Closed |
| Id | 80635 |
Reduce number object constructions/destructions during message encoding.
PBX: Update of conference id did not always work after multiple transfer accross PBXs
| Status | Closed |
| Id | 80656 |
A scenario which did not work was A calls B, B does consultation to C, B Transfers, C does consultation to D, C transfers with B on different PBX then A.
After this the conference id on the call on A should be identical to the conference id on D. This was not the case.
SOAP/TAPI applications which are keeping track of transfered calls could have a problem with this.
AD Replication: LDAP filter encoding failed, when Poll Timer was configured
| Status | Closed |
| Id | 80658 |
AD Replication: LDAP filter encoding failed, when Poll Timer was configured
Gateway: Routing of incoming SIP calls may not work
| Status | Closed |
| Id | 80709 |
... because the To-URI is not reduced to a called number (CDPN).
Gateway routing is based on CDPN and does not work for URIs.
Before this fix:
.LOG CALL 2 Alloc
.LOG CALL 2 A:Call -> / GW8::->*::
.LOG CALL 2 B:Call :081604998@212.13.249.90->:018108680@192.168.5.230 / GW8::->GW1::
After this fix:
.LOG CALL 1 Alloc
.LOG CALL 1 A:Call -> / GW8::->*::
.LOG CALL 1 B:Call 081604998->018108680 / GW8:081604998:->GW1:018108680:
SIP: Memory leak when closing signaling interface
| Status | Closed |
| Id | 80752 |
Memory leak when closing signaling interface while DNS request is pending on a call.
Phone: Message function key cannot be configured
| Status | Closed |
| Id | 80766 |
Configuration is not saved
SIP: Diversion information was wrong on incoming SIP calls
| Status | Closed |
| Id | 80812 |
Parsing of History-Info header was wrong, but only if multiple History-Info headers were present.
PBX: Routing problem with nodes/escapes/slaves with calls to extern numbers in other PBX defined nodes
| Status | Closed |
| Id | 80853 |
Could happen that the number dialed when sending the call to the extern interface was not adjusted correctly
Build Number format changed
| Status | Closed |
| Id | 80925 |
The build number of the hotfixes changes from the 90600.xx format to the 9.061xxx format. This is due to organizational changes without any other significance.
IP-DECT: Memory leak for SIP calls
| Status | Closed |
| Id | 80937 |
buffer for received name-id was not freed. Could happen with other facilities as well.
SIP: Transcation handling was wrong
| Status | Closed |
| Id | 80989 |
Double delete on a INVITE client transaction object.
0:0027:988:5 - SIP_TAC_INVITE:serial::delete(805b0c80) caller=800440ec
PBX: Adjusting called number, when sending call to extern
| Status | Closed |
| Id | 81005 |
depending on the node of the extern object and the called node, the called party number has to be adjusted (escapes added, prefixes added/removed). This did not work unders some conditions.
V9 Hotfix 13 (9061024)
Changes included in Version 9 hotfix13 Definition
New Features
myPBX: Interface for tracing
| Status | Closed |
| Id | 80973 |
The web application can now write messages into the trace file of the myPBX launcher.
Oem Code: Accelerating Boot Snmp Traps
| Status | Closed |
| Id | 81132 |
Oem Code: Accelerating Boot Snmp Traps
PBX: Handling of call limits at PBX objects improved
| Status | Closed |
| Id | 81232 |
A call from a PBX, which is sent back to the same PBX is not counted anymore. This can happen because of node-extern.
Incoming calls at master, which are above the limit are rejected now. They can be rerouted on the slave with "Route Master calls if no Master to"
SIP: New config option for endpoints not refreshing their registration during call
| Status | Closed |
| Id | 81243 |
Interoperability:
New config option /keep-active-endpoints for endpoints not refreshing their registration during call.
AVM FRITZ!Box Fon WLAN 7270 v2 (UI) 54.05.21 (Apr 2 2012)
UI hint where to update Linux AP
| Status | Closed |
| Id | 81295 |
Changed a hint on the upload/update tab and added a link to the update/upload page on the firmware side.
SIP: New config option /no-certificate-check
| Status | Closed |
| Id | 81601 |
New config option to disable validation of remote certificate name
when opening outbound TLS connection.
IP-DECT: Own priority for idle display set by messages
| Status | Closed |
| Id | 81616 |
The idle display set by messages has got an own priority now.
SoftwarePhone: Support for Jabra PRO 930
| Status | Closed |
| Id | 81618 |
Support for Jabra PRO 930 with product id 0x1016 added.
SIP: New config option "No Inband Disconnect" on GW interfaces
| Status | Closed |
| Id | 81803 |
Introduced VOIP_OPTION_NO_INBAND_DISC.
Option was missing on Gateway interfaces.
SIP: New config option /product-id-format
| Status | Closed |
| Id | 81880 |
Change the User-Agent string from
User-Agent: (innovaphone IP232/10.00 dvl [90910/90879/501])
into
User-Agent: innovaphoneIP232x90910x501
with /product-id-format 1
phone: ip222,ip232: more USB headsets supported
| Status | Closed |
| Id | 81972 |
Sennheiser and some more Jabra Headsets added. Look for "(since V9hotfix13)" in
http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset
Debug information on assertion
| Status | Closed |
| Id | 81973 |
More debug information on default event handler.
phone: ip222,ip232: support for advanced USB headset functions (redial, reject call, accept waiting call)
| Status | Closed |
| Id | 81975 |
A lot of USB headsets generate special indications to request redialing of last number dialled, to reject a ringing call, to accept a waiting call and to put the active call on hold or to switch between an active and an held call.
Bug Fixes
IP222 equalizer update
| Status | Closed |
| Id | 80004 |
- handsfree speaker equalizer enabled
- handset mic and receiver equalizer smoothed
- ADC gain reduced, input gain increased ( after ec ) to avoid clipping
- halfduplex mode disabled
IP222 IP232 IP241: repeated ethernet link status 1000M wrong
| Status | Closed |
| Id | 80029 |
.
IP222 IP232 : Default LCD backlight standby brightness reduced
| Status | Closed |
| Id | 80031 |
IP241 handsfree equalizer update, IP222 handsfree mic equalizer update
| Status | Closed |
| Id | 80157 |
- IP241 handsfree speaker equalizer enabled
- IP241 handsfree micro equalizer enabled
- IP222 handsfree micro equalizer enabled
- All three filter calculated with measured frequency response
- IP241 ADC gain reduced, input gain increased ( after ec ) to avoid clipping
phone: ip222, ip232: Pressing Talk button on USB Headset when there is an active and a held call does not transfer
| Status | Closed |
| Id | 80413 |
Instead of transfer the active call is disconnected and the held call is retrieved Status: checked in to 10.00
IP241,IP222,IP232: Show name of diverting party in incoming calls
| Status | Closed |
| Id | 80839 |
Show name of diverting party in incoming calls (not only number).
PBX-SOAP: UserRedirect - original called number got lost
| Status | Closed |
| Id | 80854 |
For example if an application used an Waiting Queue object to monitor for incoming calls and redirected these calls to agents. The agent receiving the call could not see if the call was diverted to the waiting queue already.
SIP: Send 200/OK for MESSAGE(text/plain) when accepted by application
| Status | Closed |
| Id | 81017 |
Application gives "NormalCallClearing" to SIP stack.
SIP stack better sends "200 OK" instead of "603 Decline".
SIP: Un-escape content of XML elements
| Status | Closed |
| Id | 81019 |
XML element content requires some resevered characters to be escaped (<>).
These escape sequences (> or <) must be un-escaped onthe receiving end.
IP241,IP222,IP232: Two status symbols may overlay each other
| Status | Closed |
| Id | 81027 |
Symbol for "Call diversion" and symbol for "PIN-locked" overlay each other.
Hide "Call diversion" as long as phone is PIN-locked.
SIP: Handling of multiple 401/407 responses
| Status | Closed |
| Id | 81045 |
Implemented handling of up to 3 401/407 responses per transaction.
SIP: Trap while releasing call
| Status | Closed |
| Id | 81089 |
Timer fires during call release and causes trap.
H.323: Memory leak when sending special OEM H.235 key elements
| Status | Closed |
| Id | 81160 |
Only happens if Avaya SRTP is enabled in OEM products
H.323: Potential Trap in case of high load and media renegotiation/call clearing collision
| Status | Closed |
| Id | 81173 |
There is a very unlikely situation when media-renegotiation is started and then the call is cleared, which could cause a message related to the media-renegotiation to be sent to a already deleted call object. High load could make this situation more likely.
IP241 handset equalizer
| Status | Closed |
| Id | 81174 |
Enable IP241 handset micro and speaker equalizer.
Same values as on IP222.
Reduces noise in G711 mode.
IP1060/3010/6010/22/24/28/302/305: sequence number to DSP changed from byte to word, caused jiter buffer warnings
| Status | Closed |
| Id | 81189 |
IP222/232/241/1060/3010/6010/22/24/28/302/305: min jitter buffer changed to 10ms
| Status | Closed |
| Id | 81197 |
was 35ms, now 10ms as on ip6000/800
IP-DECT: Duplicate call setup
| Status | Closed |
| Id | 81213 |
The IP-DECT Master sends in some circumstances a call twice to the same radio in the same time. This affects only the IP1202 and OEM devices, not the IP1200, and is fixed now.
IP-DECT: OEM configuration option
| Status | Closed |
| Id | 81214 |
The configuration format of an OEM configuration option was changed for the backward compatibility.
phone_orchid: micro mute when a waiting call was connected after an outbound call setup had been cancelled
| Status | Closed |
| Id | 81229 |
- A goes offhook
- B calls A, a waiting call from B is indicated on A
- A goes onhook, phone rings
- A goes offhook again and is connected to B
- A hears B, B doesn't hear A
SIP: Pass display information to application
| Status | Closed |
| Id | 81230 |
Pass display information to application when registration comes up.
IP0010 IP1060 IP3010 IP6010: During long reset the ready LED was blinking orange and the Ethernet LEDs were swaped
| Status | Closed |
| Id | 81236 |
Now the ready LED shows the green blinking during long reset.
The Ethernet LEDs are initialized directly after reset to overwrite the default setting that swaps link and speed.
X.509: Creating certificate containing IPv4 address did not work
| Status | Closed |
| Id | 81251 |
When creating a certificate or a request with an IPv4 address, the IPv4 address was mapped to an IPv6 address. This was not the expected behaviour.
172.16.10.32 -> ::ffff:172.16.10.32
Now both IPv4 and IPv6 addresses can be used.
ip222,ip232: waiting calls mute after having been accepted/connected on USB headsets, mainly seen with Jabra LINK 14201-30
| Status | Closed |
| Id | 81312 |
In some cases only the accepted call was mute and the next call was OK again but the Jabra LINK 14201-30 lost the USB connection in most cases.
Delaying the HID-commands sent to the headset solves this problem.
PBX: Objects list filter for numbers did not work correctly anymore
| Status | Closed |
| Id | 81376 |
only top level nodes were displayed
Collateral damage of fix
79982: PBX: When editing a Node object it was changed to a PBX object
SIP: Pass display information to application
| Status | Closed |
| Id | 81379 |
Pass display information received in REGISTER response to app.
IP241,IP222,IP232: Show H323-ID instead of Display Name if Display Name is not available
| Status | Closed |
| Id | 81419 |
Like on old b/w phones.
IP-DECT: Memory leak with special Cisco features
| Status | Closed |
| Id | 81463 |
Only with special OEM features.
SIP: Problems with DNS resolving of proxy adresses
| Status | Closed |
| Id | 81522 |
Not always re-tried when failed at startup.
IP241,IP222,IP232: Rendering errors on Fkey configuration screen
| Status | Closed |
| Id | 81556 |
When scrolling or leaving Fkey configuration screen.
List (Toggle) controls were not rendered correctly.
IP-DECT: Mobility Master registration
| Status | Closed |
| Id | 81571 |
The Mobility Master does not accept clients with the OEM name IP1202. This is fixed now.
HTTP: Possible trap on many simultaneous sessions
| Status | Closed |
| Id | 81597 |
Sorting the TCP sessions did not work correctly, which caused an assertion because it could happen that it was not possible to remove a session
IP-DECT: System GUI, disabled local coder options
| Status | Closed |
| Id | 81603 |
If the PARI function (only IP1202) of the IP-DECT Master is disable, configuration changes on the System GUI do not effect anything. The settings for the local coder are disabled on this GUI page now.
IP-DECT: System settings not to dynamically connected radios
| Status | Closed |
| Id | 81612 |
System settings should not be sent if the radio-master connection is a dynamic one (IP1202 only). This is fixed now.
Support for old versions of Jabra BIZ 2400 headset with 48 kHz output sampling frequency
| Status | Closed |
| Id | 81613 |
the newer versions come with the usual 16 kHz sampling frequency but have the same signature (vendor/product) as the older ones.
SoftwarePhone: Trap with trace and mute ringer event
| Status | Closed |
| Id | 81619 |
A trap occurs if the trace is enabled and the ringer mute option is toggled.
IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios
| Status | Closed |
| Id | 81693 |
In scenarios that operate one Ethernet port with 1000M and the other with 100M the switch througput was low.
Now the 1000M port is reconfigured to 100M, and the throughput is high.
SIP: Support for multiple audio media descriptions
| Status | Closed |
| Id | 81712 |
One without encryption and one with encryption:
v=0
o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144
s=SIP Call
c=IN IP4 10.30.1.144
t=0 0
m=audio 5012 RTP/AVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
m=audio 5010 RTP/SAVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
IP232: Show more of collapsed fkeys
| Status | Closed |
| Id | 81807 |
140px when expanded.
100px when collapsed (80px before).
SIP: Double Replaces header in INVITE after REFER
| Status | Closed |
| Id | 81847 |
Reject by "Swyx IpPbxSrv/8.1.0.246" with "400 Bad Request(Multiple values in single-value header Replaces)"
PBX Mobility: No response for call to busy mobile phone, if no fixed phone
| Status | Closed |
| Id | 81850 |
If a user is called with mobility configured and no fixed phone and the mobile phone was busy, then the call did not complete and was hanging as if the number was not complete. The call should be answer with busy instead.
Chrome compatibility issue on PRI statistics page
| Status | Closed |
| Id | 81859 |
If clear was clicked an error message was displayed
PBX Admin UI: Not possible to change DECT parameters for users with cf/grp admin rights
| Status | Closed |
| Id | 81871 |
The user interface prohibited the editing of DECT parameters based on the rights of the object to be edited not based on the login
phone: ip222,ip232: when a call is released from remote while another call is waiting the waiting call cannot be accepted.
| Status | Closed |
| Id | 81907 |
When the call is released from remote a buys tone is generated for two seconds. Therafter phone rings to indicate that the waiting call can be accepted now.
When trying to accept this call by pressing the headset talk button the call was disconnected instead.
SIP: Memory leak in SIP stack
| Status | Closed |
| Id | 82058 |
Memory leak in SIP stack.
V9 Hotfix 14 (9061046)
Changes included in Version 9 hotfix14 Definition
New Features
phone: support PBX-directory access via TLS
| Status | Closed |
| Id | 78275 |
A "Use TLS" checkmark has been added to the PBX directory config of the phone. If set, port 636 is used instead of port 389.
PBX-Mobility: Data Call Thru
| Status | Closed |
| Id | 81513 |
To speed up dialing with the mobility client, the called number can be posted to the PBX via HTTP before the call
IP-DECT: DECT security (2)
| Status | Closed |
| Id | 82191 |
Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).
IP-DECT: No fall-back after unattended call transfer
| Status | Closed |
| Id | 82198 |
Now there is no fall-back after an unattended call transfer and the behavior is consistent with the other call transfer types (attended, semi-attended). To switch back to the hold call the R-key must be pressed.
IP-DECT: Idle display update
| Status | Closed |
| Id | 82199 |
Handling of idle display update message in the call release message is added. Used with an OEM PBX.
phone: ip222,ip232: the variable KEYS0/HID-MAP permits to map new USB headsets to builtin descriptors
| Status | Closed |
| Id | 82635 |
Sometimes USB headsets come with a signature different from the signature of similar headsets which are already supported.
A "vars create KEYS0/HID-MAP p <map>" maps the new signature to an existing one.
<map> format is
manufacturer:product=manufacturer:product
the second manufacturer:product tuple is the signature of an already supported headset,
'manufacturer' and 'product' are plain 4 digit hex numbers without a "0x" prefix.
IP2x2, IP241: Coder Preferences for prefered coder G.722 suboptimal
| Status | Closed |
| Id | 82815 |
In case G.7222 was selected as prefered coder and the called endpoint did not support G.722, as next best coder G.729 was selected. This is typically not what is desired in such a case, G.711 is the better alternative in this case
myPBX: Show version of launcher in the list of sessions
| Status | Closed |
| Id | 82821 |
The version of the used launcher is now shown on the page PBX/myPBX.
Status:
checked-in: 10.00
checked-in: 9.00
H.323: Support for Avaya SRTP with AES128/80
| Status | Closed |
| Id | 82829 |
Needed for OEM
SoftwarePhone: HID Support for new headsets
| Status | Closed |
| Id | 83184 |
With this new version the following headsets are supported for call control:
- Jabra GO 6430 (Jabra LINK 350 USB with firmware 5.4.17 or later) with product id 0xa342. Please select the first device.
- Jabra SUPREME UC (Jabra LINK 360 USB) with product id 0xa346. Please select the first device.
- Jabra PRO 9470 with product id 0x1042.
- Sennheiser VoIP USB headset (SH 350 IP) with product id 0x0008.
- Sennheiser DW Office with product id 0x740a. Please select the first device.
- Sennheiser CEHS-CI 02 (USB adapter cable) with product id 0x0030. Please select the second device.
PBX: Description was missing for DECT System object
| Status | Closed |
| Id | 83198 |
A configurable description is useful for the DECT System object as for all other objects
Permit logging to a second (shadow) log server
| Status | Closed |
| Id | 83206 |
"Services/Logging/Log Server/Log Server Shadow/Address" defines the adress of a second server.
"Services/Logging/Log Server/Log Server Shadow/Enable" starts/stops logging to the second server.
Except the address the configuration for the second server is copied from the first server.
Bug Fixes
Incorrect disk usage calculation for more than 4GB
| Status | Closed |
| Id | 81209 |
The disk usage calculation was wrong, if more than ~4 GB of the card were used.
PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2
| Status | Closed |
| Id | 81370 |
Announcement disconnect changed state
PBX: Switch from Music on Hold to inband ringback on Alert after unpark oder transfer
| Status | Closed |
| Id | 81407 |
To give the caller feedback that soon somebody may answer the call
PBX: Don't forward in-band info indicator if no media channel
| Status | Closed |
| Id | 81879 |
Not good to indicate in-band info available if no media channel can be negotiated
SIP: Trap when configuring user presence
| Status | Closed |
| Id | 81996 |
Trap may occur when configuring user presence.
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.
phone: ip222,232.241: no notification tone on a successfull redial attempt
| Status | Closed |
| Id | 82166 |
On a failing or unanswered call the menu key opens the "Recall" menu. If "Redial" is selected the call is automatically redialed for 20 minutes in intervals depending on the result of the previous attempt. On success the user should be notified about the connection.
IP-DECT: Trap during subscribing handsets
| Status | Closed |
| Id | 82190 |
A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.
IP-DECT: Wrong release code
| Status | Closed |
| Id | 82195 |
The IP-DECT sends the wrong release code "User not reachable" if the handset was not connected and the remote parts disconnects. This is fixed now.
IP-DECT: No delay with semi-attended call transfer and SIP
| Status | Closed |
| Id | 82197 |
If the SIP protocol is used and the user do a semi-attended call transfer, the call transfer is directly confirmed again. The semi-attended call transfer is stored in the base station and executed as an attended call transfer if the target party connects.
IP-DECT: Display info with failed user log-in
| Status | Closed |
| Id | 82200 |
If the user log-in fails, the cause is shown in the display.
IP-DECT: Short tone info by PBX only one time
| Status | Closed |
| Id | 82204 |
Tone information with defined length and requested by the gatekeeper is played only one time to the handset. This is used by an OEM PBX and fixed now.
IP-DECT: Hanging aborted semi-attended call transfers with SIP
| Status | Closed |
| Id | 82205 |
If the SIP protocol is used, a semi-attended call transfer is done by the user and the call transfer can not be executed, the remaining call party is not disconnected. This is added now.
IP-DECT: LDAP server GUI description
| Status | Closed |
| Id | 82258 |
The LDAP server GUI description is wrong and corrected now. (The LDAP server can not be a HTTP server.)
DHCP: Name registration at WINS was not refreshed as requested by TTL in registration response
| Status | Closed |
| Id | 82289 |
If the DHCP-client gets a lease containing a WINS-server address and a NETBIOS node type P or M (1 or 2) the client tries to register it's NETBIOS-name (ipxxx-xx-xx-xx) with the WINS-server. The TTL returned by the server in the registration response determines when a name refresh has to be sent.
Phones: Presence info during ringing state may show garbage data
| Status | Closed |
| Id | 82306 |
In case a presence update arrives at the phone while phone is in ringback state.
Have been observed in conjunction with call forking with mobility only.
myPBX: Support contact names containing a single quote
| Status | Closed |
| Id | 82323 |
Configuring contacts with a H.323 id containing a single quote (') caused script errors in the web application.
PBX: RTP-DTMF was disabled by Voice Mail object during re-negotiation
| Status | Closed |
| Id | 82332 |
RTP-DTMF acc. to RFC-2833 was disabled by Voice Mail object during re-negotiation.
Gateway CDR with '0. 0' charge amount
| Status | Closed |
| Id | 82359 |
Should be '0.00' instead
phone: call completion did not recover when DND(busy) was set at the phone requesting the call completion
| Status | Closed |
| Id | 82390 |
when DND(busy) was set on the phone reqesting the call completion and was cleared some time later a "Recall possible" was not indicated anymore although a pending call completion was indicated on the called phone.
ipv6: memory leak when sending fragmented packets
| Status | Closed |
| Id | 82394 |
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.
Kerberos: Allow editing multiple fields in admin UI
| Status | Closed |
| Id | 82425 |
This is a fix for the page General/Kerberos in the admin UI.
Editing more that one item at a time could result in strange effects. Now it is possible to edit multiple fields.
phone_inca: when a call completion was set up with CLIR active the called party was not displayed on a possible recall
| Status | Closed |
| Id | 82435 |
A sets CLIR, A calls B, B is busy
A sets a CCBS request via Menu/Recall
B goes on Hook
A rings and sees 'anonymous' instad of the number of 'B', status line is empty (should show "Recall possible")
IP-DECT: Display info with remote control call
| Status | Closed |
| Id | 82444 |
The display info shown after accepting a remote control call is fixed now.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: enable modembypass
| Status | Closed |
| Id | 82458 |
Modembypass is enabled on all calls with disabled T.38 and coders G711A or G711U.
Switch to modem bypass is indicated in the trace by "switch to modembypass".
The feature can be disabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl
Modembypass works best if T38 is disabled on both sides.
If T38 is enabled on the called side the CED may trigger a T38 session, this changed back to voice and modem bypass is enabled (if G711 is active). The first modem tone is interupted, but we still have modembypass on both sides.
If T38 is enabled on the calling side the calling side stays on regular G711.
IP800 IP6000: reduce probability of false DTMF detection
| Status | Closed |
| Id | 82471 |
Change DTMF detector signal to noise ratio from 12db to 18db.
Gateway: 'Enable PCM' option added at the CONF interface of the IP800
| Status | Closed |
| Id | 82481 |
The option 'Enable PCM' is available at the CONF interface of the IP800 now.
Gateway: PCM mode of the CONF interface fixed
| Status | Closed |
| Id | 82482 |
The PCM mode of the CONF interface is not activated for a call even though it is configured and possible. This is fixed now.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: T38 trace flag added
| Status | Closed |
| Id | 82484 |
Allows to trace T38 connection on the PCM port and on the DSP host interface.
Use this if fax modem problems are suspected.
Enable at http://addr/debug.xml at trace->T38 trace.
Gateway: Routing problem with blockdial route and following matching non-blockdial incomplete routes
| Status | Closed |
| Id | 82486 |
Example:
Blockdial Route 00->
After this a non-blockdial route with 0...
If now a number of 001 was dialed, the first route should match and after the enbock dialout the call should be sent to the destination of the route. Instead the call was rejected with "no destination found"
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.
Debug "HTTP_GET LOG_HTTP.1: retry, authentication failed" removed
| Status | Closed |
| Id | 82499 |
Phone: Display text received with BYE
| Status | Closed |
| Id | 82525 |
Pass display text to phoneapp.
SIP: Trap during call handling
| Status | Closed |
| Id | 82544 |
Trap during call handling
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: update to DSP code 660.27
| Status | Closed |
| Id | 82551 |
660.27 is the released version on the audiocodes web.
Fax relay and modem bypass improved.
IP222 IP232 IP241: update to DSP code 660.27
| Status | Closed |
| Id | 82554 |
660.27 is the released version on the audiocodes web.
Improved acoustic echocanceller.
phone: call forwarding options offered even when call forwarding was not possible
| Status | Closed |
| Id | 82567 |
Call forwarding is not supported when running SIP. But when the menu key was pressed after entering a number call forwarding options were offered (happened with the primary registration only).
PBX Mobility: Trap in case of Transfer of a call from a mobile endpoint to another mobile endpoint
| Status | Closed |
| Id | 82584 |
The Trap happens in the following call scenarion
- Mobile endpoint calls in, using mobility two-stage dialing
- call is accepted at local phone
- on local phone a consultation call is initiated to another user with mobility
- when mobile phone rings, a transfer is initiated on local phone
- the called mobile phone accepts the call
- the trap happens when the called mobile phone hangs up
There could be other call scenarions where the trap happens as well
SIP: SRTP key exchange failed
| Status | Closed |
| Id | 82616 |
Bug in base64 decoding of SRTP key.
PBX Trunk: Number to Name Feature did not work for calls to busy endpoints
| Status | Closed |
| Id | 82619 |
Such a call was not rejected with cause 'User Busy', but was just hanging.
PBX Mobility: Call to mobile phone was sent with invalid diverting information
| Status | Closed |
| Id | 82622 |
A call to a mobile phone is sent with a diverting leg2 info, which means, the call contains the information, that it was diverted by the called user to the mobile phone. So in theory this could be displayed on a mobile phone.
The coding of this information was wrong and created interop problems with some networks.
SIP: Trap on subscription handling
| Status | Closed |
| Id | 82623 |
Trap on RAS_REGISTRATION_VERIFY between sending SUBSCRIBE and receiving 200/OK.
SIP: Removed cisco-special retrieve signaling
| Status | Closed |
| Id | 82637 |
Removed cisco-special retrieve signaling.
phone: ip222,ip232: Plantronics APU70 - Savi 7xx - Radio Link not cleared on release from remote
| Status | Closed |
| Id | 82654 |
On a release from remote for a call set up by pressing the Talk button (headset or base) the Radio Link between base and headset was not cleared until the Talk button was pressed again.
SIP: Mobility did not work due to RTP-DTMF
| Status | Closed |
| Id | 82674 |
DTMF must be passed through signaling channel to get mobility working.
Suppress RTP-DTMF capability in SDP answer also.
IP3010 IP810: number of DSP channels and number of conference channels was wrong.
| Status | Closed |
| Id | 82675 |
Now the IP810 shows 20 DSP channels and 30 conference channels.
Now the IP3010 shows 42 DSP channels and 60 conference channels.
SIP: Restart NAT discovery if failed
| Status | Closed |
| Id | 82676 |
Restart NAT discovery if failed
IP222 IP232 IP241: Reduce DTMF level
| Status | Closed |
| Id | 82685 |
During DTMF receive and transmit levels similar as on IP240 are used
-10db level
0xc0=208--> 22db attenuation
also insgesamt ein level von -32db ( bei Vollauschlag ) oder -29dbm.
Der alte Wert beim ac_phone3.cpp war -9db
Weitere Diskussion:
Es gibt den Fall das inband DTMF zum IP Netz geschickt wird, da gabs in Fall 59846 die Änderung zum IP mit LEV=0x28 -->-10db und Attenuation 0xff-->18db, also mit -28db zu senden.
Da das gut funktioniert und die beiden Pegel nicht so unterschiedlich sind unde der ac_dsp3 nicht unterschiedliche Pegel zum IP und zum Codec kann nehmen wir die -28db=-25dbm.
Laut www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-Q.24-198811-I!!PDF-E&type=item Tabelle A-1 sind die -25dbm am unteren Ende, da ist vom Wert A
-22..-30 je nach Administration die Rede. Wahrscheinlich ist das kein Problem,
bei Audiocodes (ac_dsp2) kann man die sensitivity von -28 bis -38db einstellen (DTMF_DETECTION_ENERGY_THRESHOLD__28dBm)
-->
IP222,IP232: Going offhook in call list always dials first list entry (not touched list entry)
| Status | Closed |
| Id | 82732 |
Going offhook in call list always dials first list entry (not touched list entry)
Memory leak after Firmware or Bootcode download
| Status | Closed |
| Id | 82740 |
After a Firmware or bootcode download via Maintenance/Download the associated command processor instance was not released.
IP-DECT: Call transfer compatibility with SIP
| Status | Closed |
| Id | 82742 |
For compatibility reasons with the SIP protocol the call transfer initiate result message should not be sent until the connect message is received. This is changed now again.
SIP: Trap if multiple calls arrive at same time
| Status | Closed |
| Id | 82743 |
Trap if multiple calls arrive at same time.
SIP: Bug in digest authorization
| Status | Closed |
| Id | 82761 |
Sometimes wrong method is used in digest calculation.
phone: ip222,ip232: USB headset mute after activation of changes in codec parameters, headset icon cleared on status line
| Status | Closed |
| Id | 82823 |
After for example
config add AC-DSP0 HEADSET /InputGain 32
config activate
the headset icon was cleared on status line and the headset was mute although the headset Talk key was handled.
PBX Waiting: Evaluate Busy on ... Calls for calls to an operator
| Status | Closed |
| Id | 82858 |
If a operator has configured Busy on 1 call, there should be no call from the Waiting Queue if the operator is already busy.
PBX: Avoid signaling loop after call transfer
| Status | Closed |
| Id | 82865 |
A signaling loop could be created by calling from a phone registered at one PBX to a phone at another PBX, then putting the call on hold and do the same call again, accept on the other side the waiting call. If both parties do then a transfer there is the signaling loop.
Such loop ist now detected and the call is cleared.
PBX Broadcast: No diverting name sent with broadcasted call
| Status | Closed |
| Id | 82880 |
The diverting and original-called name info was missing from the diverting leg2 info generated by the broadcast object
IP0010 IP1060 IP3010 IP6010 IP810 IP302 IP305: ISDN: Enable fax detection only after connect - fixed
| Status | Closed |
| Id | 82900 |
Fax detection was not enabled in all cases.
Happens in test/10.00/box/dsp/ip6010 with
fix: #78316: SIP/H.323: Don't complete media negotiation if no media can be seen
IP1060: Memory size is not correctly shown
| Status | Closed |
| Id | 82906 |
The memory size is not correctly shown in the IP1060 in diagnostics counter page.
phone: a calling party name found by inverse directory lookup for an external call was sometimes not stored in call list
| Status | Closed |
| Id | 82919 |
When the external call setup came in with a name identification provided by the external source and there was another name found by inverse directory lookup the name from directory was displayed on the call screen but the name identificication was stored in the call list. Now the name found by inverse directory lookup will be stored.
PBX Routing: Node extern did not work for calls from a trunk marked as local object
| Status | Closed |
| Id | 82948 |
The call was not routed back to the originating slave
IP222 IP232 IP241: New equalizer and volume setting
| Status | Closed |
| Id | 82952 |
- use equalizer up to approx 6db
- use digital volume to adjust volume
SIP: Trap on out of memory
| Status | Closed |
| Id | 83004 |
SIP-Client allocations not deleted.
Outbound control calls without facility interworking.
PBX Routing: A CFNR at a PBX object for WAN re-routing did not work if it contained escapes
| Status | Closed |
| Id | 83017 |
The number configured at the PBX object is interpreted in the context of the node of this PBX object. If escapes were needed to dial the WAN trunk, it did not work.
External-UC: Presence info assigned to wrong PBX object
| Status | Closed |
| Id | 83020 |
When presence eventlist is received from External-UC
all presence info was assigned to same PBX object.
PBX: Editing Config Templates impossible, if by some old firmware a strange config ended up in a User object
| Status | Closed |
| Id | 83033 |
The problem was a empty hardware id (hw=""). This is now ignored.
PBX mobility: The forking destination put in for mobility at a user should be dialed from the node of the user
| Status | Closed |
| Id | 83040 |
This number was dialed from the node of the Mobility object. This was confusing, because this number was configured at the user and it was also different behaviour as with forking without mobility
PBX: Standyby PBX generated alarms for missing slave registrations, even if active PBX up
| Status | Closed |
| Id | 83089 |
There should be only an alarm, if the standby PBX is active
SIP: One-way audio after mutual hold on dect systems
| Status | Closed |
| Id | 83126 |
Dect ep gets FTY_HOLD_NOTIFY but no FTY_RETRIEVE_NOTIFY.
X.509: Avoid alarms on missing system time after reboot
| Status | Closed |
| Id | 83137 |
Allow 60 seconds for setting the system time before an alarm is set. Also certificates are now rejected silently, in that time.
SIP: Problems with presence signaling on External-UC link
| Status | Closed |
| Id | 83177 |
Missing parameters on Contact-URI in 200/OK for SUBSCRIBE(presence):
maddr
transport
Wrong Contact-URI in presence XML in PUBLISH.
IP-DECT: Support for more than 32 LDAP attributes
| Status | Closed |
| Id | 83191 |
The IP-DECT devices support up to 256 LDAP attributes now.
PBX: Support for more than 64 LDAP attributes
| Status | Closed |
| Id | 83192 |
The PBX supports up to 256 LDAP attributes now.
IP-DECT: LDAP replication alarm loop
| Status | Closed |
| Id | 83249 |
If the LDAP replication is configured, but it can not be synchronized, there is an alarm loop. This is fixed now.
Gateway: On IP24, IP302, IP305 it could happen that multiple routes disappeared if a route was deleted
| Status | Closed |
| Id | 83278 |
Hard to predict when this happened.
Gateway: No Media received event was generated for T.38 calls, which started without actual RTP Traffic
| Status | Closed |
| Id | 83433 |
This happend for calls to/from Fax servers, which switch to T.38 without sending any RTP packets first.
SIP: Multiple subscriptions for 'message-summary'
| Status | Closed |
| Id | 83469 |
After every RAS_START another subscription for 'message-summary' is established.
V9 Hotfix 15
Changes included in Version 9 hotfix15 Definition
New Features
New remote control codes to be used for phone tests by soap applications
| Status | To-decide |
| Id | 83468 |
The new UserRc codes are executed only when the addressed phone is either in handset, headset or handsfree mode, i.e when calling, connected or disconnected but not when alerting:
16 - change to handset mode
17 - change to headset mode
18 - change to handsfree mode
19 - monitor mode on (add speaker to handset or headset mode)
20 - monitor mode off (back to plain handset or headset mode)
Alarm/Event handling: Authentication for received remote Alarms/Events
| Status | To-decide |
| Id | 83603 |
Allow by configuration to only accept authenticated alarms or events
PBX: Preparations for objects visible only if appropriate license installed
| Status | Closed |
| Id | 83615 |
for v10
PBX: New Feature to allow registration with password to devices regardless of address filter
| Status | Closed |
| Id | 83794 |
It is now possible to configure a flag at a device to allow a registration for this device even if there is an IP Filter which does not match. This is useful if registrations from the public internet to the PBX shall be possible. Without this feature this could be opened only for the complete PBX. Now it can be restricted to a few devices.
SoftwarePhone: Signature added
| Status | Closed |
| Id | 83915 |
The SoftwarePhone installer and the install package is signed now.
IP-DECT: Phone book 'Use TLS' option
| Status | Closed |
| Id | 84001 |
'Use TLS' option added for the central phone book search. This changes the standard port from 389 to 636 if no port is configured. The central phone book search is only available with the IP1202.
simple static logging interface
| Status | To-decide |
| Id | 84204 |
to simplify sending of log messages, alarms and errors a simple static interface to the logging module was added. log_if::log(class serial src, const class event & event) passes the given event to the primary logging module (aka LOG0). This works also with 'src' = 0.
myPBX: Pass selected device to launcher
| Status | Closed |
| Id | 84556 |
Needed for version 10 remote video
SIP: Announcing "a=T38MaxBitRate:14400" in T.38 offer
| Status | Closed |
| Id | 84770 |
Announcing "a=T38MaxBitRate:14400" in SDP offer for T.38
since all current devices support that modem speed.
SIP: Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE
| Status | To-decide |
| Id | 84939 |
Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE.
IP-DECT: Phone book error events
| Status | Closed |
| Id | 85001 |
The phone book module sends an error event now if the LDAP directory search fails.
Bug Fixes
IP-DECT: Trap during subscribing handsets
| Status | Closed |
| Id | 83690 |
A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.
Trap identification:
XCPT: no 2 (TLB load) pc 942e23d8 ra 942e23cc va 00000000
PBX Waiting: Trap on leak-check if dtmf maps are configured
| Status | Closed |
| Id | 83691 |
Only happened if debug.xml leak check was used
IP222 IP232 IP241: Adjust gains for better echo canceller performance
| Status | To-decide |
| Id | 83703 |
..
myPBX: Remove unimportant notifications
| Status | Closed |
| Id | 83707 |
The following events are not so important that the user has to be notified using a windows bubble notification.
- Visibility requests
- Missed calls
IP222 IP232 IP241: Enable noise reduction
| Status | To-decide |
| Id | 83715 |
Enable noise reduction on all handset/handsfree/DHSG headset micro.
Use DSP code 660.27.pa.03
Noise reduction parameter can be tuned at
http://addr/AC-DSP0/mod_cmd.xml?xsl=phone-dsp.xsl
Voicemail: Prevent <prompt>, <record> on incoming control calls
| Status | Closed |
| Id | 83826 |
Turned out to cause idle-reset requests never being processed.
An administrative solution also exists: Watch out for the URL variable "$_noctl" in this article:
http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables
HTTPCLIENT: Allow configuration of a http authenticated URL even if the server needs no authentication
| Status | To-decide |
| Id | 83900 |
If a password was configured and the server didnt need it, a HTTP put created a file of zero length.
PBX Waiting: Trap on collision of operator connect and two-stage dialing
| Status | Closed |
| Id | 83926 |
If two-stage dialing (Maps) is used to call a Trunk or Gateway object, the call is sent after a blockdial timeout. If an operator connected the call before this timeout happened, a trap occured.
For this to happen DTMF maps and operators have to be configured on the same Waiting Queue object, with is kind of unusual
PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received
| Status | To-decide |
| Id | 83940 |
Calls from a Trunk must be explicitly marked as internal with respective connected number, otherwise they should be treated as external
HTTP: Chunked transfer fails if the last 2 bytes of the chunk header are in the next tcp packet
| Status | To-decide |
| Id | 83986 |
Seen with the application platform as broken pipe
PBX: Registration with Name/Number did not work correctly if default device not first
| Status | To-decide |
| Id | 83989 |
For a registration with name or number, the information if the PBX password shall be used was always taken from the first device regardless if this was the default device (hw-id identical to name) or not
PBX: Partial Rerouting was prohibited in Alerting State (CFNR)
| Status | To-decide |
| Id | 83993 |
This was done under the assumption the partial rerouting is not supported as CFNR by public networks.
PBX Waiting: Call to operator with Twin Phone Checkmark did not work anymore
| Status | Closed |
| Id | 84084 |
Collateral damage of
fix: #82858: PBX Waiting: Evaluate Busy on ... Calls for calls to an operator
SoftwarePhone: Product string
| Status | Closed |
| Id | 84119 |
The product string is changed, used e.g with the PBX registration.
phone: provide complete dialog info to a phoneapp
| Status | To-decide |
| Id | 84192 |
Both group indications and dialog infos are signaled via a group indication facility. For dialog infos the parked_to_alerting member was overloaded to provide the info as expected by the existing phoneapp.
Now the parked_to_alerting member is passed to a phoneapp as received.
IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added)
| Status | To-decide |
| Id | 84200 |
In scenarios with frequent transistions of the attached PC to sleep renegotiating the link speed may be undesired. For this case the force same speed mechanism can be disabled.
Other changes:
1000M is only changed to 100M if the other port runs at 100M. The previous version changed from 1000M to 100M if the other port runs at 100M or 10M.
The statistics can be collected from the PC port or from the LAN prot or from both.
Packet forwarding on the PC port is disabled if the port is down to avoid misleading collision counter behaviour.
CDR fixes for external call detection
| Status | To-decide |
| Id | 84211 |
The type="ext" attribute was not set reliably. Additionaly an attribute pseudo was added to the <user/> tag to indicate the type of object the CDRis created for.
http client : authentication was not retried after a failure when the offending request was repeated in the same session
| Status | To-decide |
| Id | 84217 |
When a httpclient user repeated a failing request in the same session the authentication was not tried again. Thus a change of the client side URL password or a change of the server side password had no effect until a new session was started.
IP6000: Prevent blinking error LED on old IP6000 with HW-Build <110
| Status | To-decide |
| Id | 84227 |
Conference DSP driver was started on old hardware that doesnt support the conference DSP
IP222 IP232 IP241: pressing speaker key when phone is in handset mode switches to handsfree mode instead to toggle monitor mode
| Status | To-decide |
| Id | 84297 |
handset/headset plus speaker is not supported
myPBX: Inconsistent display of group monitoring rights
| Status | Closed |
| Id | 84350 |
In the visibility settings groups were not displayed if the membership is "dynamic out". But members of that group still have monitoring rights.
phone: when scrolling directory search results sometimes one of the numbers of a contact was not displayed
| Status | To-decide |
| Id | 84362 |
the tag characters assigned to the different numbers were not included in sort order.
phone: status messages for outbound external calls were sometimes garbled on display
| Status | To-decide |
| Id | 84365 |
happened when for a preceeding outbound internal call in alerting state presence info was displayed
IP0010: DSP didnt start with build 9061044
| Status | To-decide |
| Id | 84388 |
Page Ldap/Replicator/Status didn't display in WebKit
| Status | Closed |
| Id | 84400 |
Page Ldap/Replicator/Status didn't display in WebKit
phone_orchid: displaying both dialed and connected number may be misleading, it's better to omit dialed number
| Status | To-decide |
| Id | 84422 |
when for example 022222222 was dialed and the network reported a connected number 03022222222 the display info "022222222 -> 03022222222" looked like a transfer.
Hide LDAP Server Password For Viewer Accounts
| Status | Closed |
| Id | 84557 |
Was accessible for viewers
phone: enable directory search function key also in connected state
| Status | To-decide |
| Id | 84559 |
it's sometimes useful to browse the directories for a number while talking.
myPBX: Only send one command at a time
| Status | Closed |
| Id | 84568 |
Queue commands instead of sending overlapping commands. This limits the number of open AJAX connections to two.
H.323: Accidential fallback to slowstart if faststart response received in PROGRESS
| Status | To-decide |
| Id | 84601 |
Only happened in some H.323/SIP interop scenarios
IP222 IP232 IP241: Updated gain and equalizer setting
| Status | To-decide |
| Id | 84605 |
Tuned for high MOS values according to ETSI ES202737 ES202738 ES202739 ES202740
phone: ip222, ip232: status stage added to all USB control transfers
| Status | To-decide |
| Id | 84617 |
the status stage is mandatory as well for IN as for OUT control tranfers
H.323: More information on "Unexpected Message" event
| Status | To-decide |
| Id | 84699 |
The message type and the state for which this message was unexpected is needed to find out what the problem is.
SIP: Trap during channel handling
| Status | To-decide |
| Id | 84800 |
Rare trap when re-assigning channels.
IP232,IP222,IP241: Rendering errors when trying to use backround image with indexed colors
| Status | To-decide |
| Id | 84814 |
PNG mode "indexed colors" is not supported.
Trying to use an indexed color png leaves phone screen in bad shape.
NAT: Don't forward DNS requests from public network
| Status | To-decide |
| Id | 84842 |
As kind of denial of service attack, bursts of incoming DNS requests were seen. The nat process was forwarding these requests to the public DNS. This is a useful function for DNS requests from the private network, but not for requests from the public network.
These DNS requests are now discarded
H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC
| Status | To-decide |
| Id | 84848 |
Caused SIP interop problems with CUCM
Maximum LDAP PDU Size Too Small
| Status | Closed |
| Id | 84851 |
Maximum LDAP PDU Size Too Small. Now internal maximum allocation unit plus a bit for encoding overhead.
PBX Broadcast: Call to group members was not cleared when CFNR was executed
| Status | To-decide |
| Id | 84857 |
This was a collateral damage from
fix: #79549: PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object
IPVA: V10 Code Merge Aiming To Stabilize FW Upload To DRAM
| Status | Closed |
| Id | 84862 |
Addresses a problem in conjunction with innovaphone's automated test environment
PBX: Filters should only be applied to calls with media channels
| Status | To-decide |
| Id | 84892 |
A join group operation could be prohibited with filters. This was unexpected.
phone: ip222, ip232: raise alarm if an USB Headset does not respond on USB bus anymore
| Status | To-decide |
| Id | 84893 |
SIP: Fix for media negotiation on calls re-routed from TONE interface to outbound SIP
| Status | To-decide |
| Id | 84932 |
Helps on gateway interfaces configured for media-relay with exclusive codec.
IP-DECT: Call transfer timer
| Status | Closed |
| Id | 84951 |
The call transfer timer is stopped with the call proceeding event now. This fixes a call transfer to e.g. a mobile user with a delayed alert.
IP-DECT: Reverse phone book
| Status | Closed |
| Id | 84953 |
The reverse phone book does not work till hotfix 11. This is fixed again now.
| Status | To-decide |
| Id | 84991 |
Loosing remote IP addresses when DNS becomes temorarily unavailable.
Results in interfaces without remote ip addresses.
PBX Routing: A CFNR on a slave gateway object redirecting the call to the master failed
| Status | To-decide |
| Id | 85017 |
This was a collateral damage of
fix: #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX
SIP: Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI)
| Status | To-decide |
| Id | 85047 |
Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI).
IP222 IP232 IP241: Headet volume adjust added, re-enable noise reduction in headset mode
| Status | To-decide |
| Id | 85058 |
Headset volume adjust can be done with the webinterface at
http://addr/dsp.xml
This feature can be used to adapt to different DHSG headsets.
The noise reduction in headset mode is re-enabled to avoid sporadic noise with
SIP: Memory leak
| Status | Closed |
| Id | 85083 |
Memory leak.