Hi, we are experiencing a problem with voicemail on TB2.6. The problem is only affecting Polycom phones as it works fine on others, Linksys, Snom, X-Lite.
When a client dials *97 on the Polycom IP650 or IP550 they get ' you have 1 new' then VM hangs up... As I said it is only with the Polycom Phones, any clues anybody ( Polycopms running FW v3.0.1.0032
Polycom 550 / 650 with TB Voicemail
In addition to the earlier post, I haver the same phones 'out of the box running the factory set Firfmware v2 and the VM feature is fine, is there a problem with Firmware v3 and TB Voicemail?????
Is it possible to downgrade the firmware to the previous versions using my TFTP server???
Jedski
Yes, you can use your TFTP server on TB to change firmware versions. You can just put the files in a tftpboot/ or something similar. Then change the boot server address in the phone to point to the new directory. You can use the same directory too but you risk overwriting changes made previously.
I recommend you set up a server on your laptop and point the phone to your laptop as a boot server, modify some default Polycom files so that you can test one phone without risking anything on the Trixbox.
Can you get a wireshark capture of this issue? That will show you if the Polycom or the TB is sending the SIP BYE message.
Did you try *98 ? What was the result
Hi, I tried *98 and the result is the same, howvever when using a different make of phone i.e. Linksys SPA942 the problem does not transpire and the VM voices are ok?, could it be the Polycom phones sip config files???
Hi this is the output when the VM does not work with the Polycom
-- Executing [*97@Reality:106] VoiceMailMain("SIP/204-b756c820", "204@default") in new stack
--
--
--
--
== Spawn extension (Reality, *97, 106) exited non-zero on 'SIP/204-b756c820'
-- Executing [h@Reality:1] Macro("SIP/204-b756c820", "hangupcall|") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/204-b756c820", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/204-b756c820", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/204-b756c820", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/204-b756c820", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/204-b756c820", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/204-b756c820", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/204-b756c820' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/204-b756c820'
Extension Changed 204[ext-local] new state Idle for Notify User 202
The first problem you have is telling a customer to dial the code. You should have the phones setup so the messages key works.
Can you get a Wireshark capture of this issue? This would show a lot. Could it be a codec issue? What is different about the other phone models?
I really suggest that you load default Polycom config files on to one phone and test. It very well could be a bug or simply a mis-configuration but some due diligence needs to be done to verify.
Files can be downloaded at www.polycom.com/support/voice
Hi thanks for the reply, this is realy weird as it is happenning on other models of phones too, Linksys SPA, Yealink, I have installed the new TB with AST 1.6.09 and it does on that too. Also all phones are now on original Firmware and it still happens.
Thought it might be a damaged audio VM file so swapped them out but still does it, whats surprising is it only does it if you have new message(s)
the VM attendant will say ' you have 2 new ' and then the line drops, so it is always at the same point for all phones and all TB systems. I have even tried different soundsets ( different sound file formats) and it still drops out.
I have arranged for some wireshark captures to be done and will post as soon as I can, thanks for the input so far guys..
Hi,
this is getting stranger and stranger, it seems the problem doesn't affect all VM boxes or all phones,
I have a wireshark capture of a dropped call but it is 66,900 lines long should i post or can i email it to somebody
tHANKS
jEDSKI

Member Since:
2007-04-08