Hi everyone,
This is my first post to the list, although I am a long term user of Asterisk.
I have recently found a problem that I just can't seem to solve.
I have a client that has an Ubuntu x64 based Asterisk server with and ISDN
Dahdi interface and about 25 SIP handsets. Everything was working fine in
Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one
single issue that I can't explain.
I have an extension that if you call it, it will play a sound file and hangup.
Pretty simple stuff. Below is the extensions.conf entry for this extension.
exten => 849,1,Playback(custom/ceh-meetingmsg)
exten => 849,n,Hangup
The following happens if I dial it from a SIP handset
== Using SIP RTP CoS mark 5
-- Executing [...@smallanimals:1] Playback("SIP/812-00000074",
"custom/ceh-meetingmsg") in new stack
-- <SIP/812-00000074> Playing 'custom/ceh-meetingmsg.gsm' (language 'en')
-- Executing [...@smallanimals:2] Hangup("SIP/812-00000074", "") in new
stack
== Spawn extension (smallanimals, 849, 2) exited non-zero on
'SIP/812-00000074'
The scenario is during the day, if my client has a staff meeting, they simply
turn on call forwarding on the reception phone to this extension. In the past,
the audio would start as soon as the caller dials in.
After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
finishes. On the Asterisk console, I can see that the sound file is indeed
playing, but we can't hear it. This happens if I am dialing the from a SIP
extension on the phone system, or if I dial in from the public phone system.
== Using SIP RTP CoS mark 5
-- Executing [...@smallanimals:1] Dial("SIP/811-00000046", "SIP/812,60") in
new stack
== Using SIP RTP CoS mark 5
-- Called 812
-- Got SIP response 302 "Moved Temporarily" back from 192.168.1.148
-- Now forwarding SIP/811-00000046 to 'Local/8...@smallanimals' (thanks to
SIP/812-00000047)
-- Executing [...@smallanimals:1]
Playback("Local/8...@smallanimals-b5dd;2", "custom/ceh-meetingmsg") in new stack
-- <Local/8...@smallanimals-b5dd;2> Playing 'custom/ceh-meetingmsg.gsm'
(language 'en')
I have tried so many things that I have lost count, and I humbly ask the
collective intelligence of the Asterisk community for assistance.
Many thanks
aF
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