[asterisk-users] Asterisk not transcoding between installed codecs

[Brent Davidson]

Do you have canreinvite=no in the sip client configuration?  If not then 
the two sip phones are probably issuing a reinvite command and taking 
asterisk out of the call path.  If that happens and the phones can't 
reach consensus on a codec then you run into audio problems.  If you're 
not a provider and just using asterisk as a PBX then it's probably 
better to set the phones up with a matching codec set and allow them to 
establish a direct connection between each other to keep load off the 
Asterisk server.  Otherwise set canreinvite=no and Asterisk should 
transcode correctly.

Good luck,
-Brent

Gonzalo Servat wrote:
> Hi All,
>
> I have 2 SIP clients configured and connected to Asterisk. When I 
> place a call from SIP1 to SIP2, if both codecs are the same then 
> everything works as expected. I then allowed one of the clients to use 
> alaw instead of ulaw and there were audio problems (couldn't hear the 
> other end, etc). Same thing happened when I tried to use gsm<->alaw/ulaw.
>
> Any ideas? I'm using 1.6.0-beta4.
>
> Thanks!
> Gonzalo
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