Hi Alex,
I'm new to this list, but I had this problem too, and I solved it looking at the codecs the sip handsets use, and then I converted the voice prompts to that codec just like Philipp said.. Ondrej On Tue, Aug 31, 2010 at 10:04 AM, Alex Ferrara <a...@receptiveit.com.au>wrote: > 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 > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- -- Ondrej Škopek
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users