[asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Ondrej Škopek]

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 <alex at 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 [849 at smallanimals:1] Playback("SIP/812-00000074",
> "custom/ceh-meetingmsg") in new stack
>    -- <SIP/812-00000074> Playing 'custom/ceh-meetingmsg.gsm' (language
> 'en')
>    -- Executing [849 at 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 [812 at 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/849 at smallanimals' (thanks
> to SIP/812-00000047)
>    -- Executing [849 at smallanimals:1] Playback("Local/849 at smallanimals-b5dd;2",
> "custom/ceh-meetingmsg") in new stack
>    -- <Local/849 at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100831/1568158f/attachment.htm