[asterisk-users] Asterisk not transcoding between installed codecs

[Gonzalo Servat]

On Wed, Mar 12, 2008 at 12:58 PM, Brent Davidson <
brent at texascountrytitle.com> wrote:

>  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.
>

Brent,

Thank you veeeery much for replying. I thought the message went unseen but
found your reply when I went to look at the thread :)

You're absolutely right. Looks like the SIP client was messing up (or
something) when different codecs were used. I tried canreinvite=no and it
worked perfectly, but as you said, it's best to bypass Asterisk when talking
between clients on the same network. I tried a different IAX client and it
had no problems using different codecs (with canreinvite=yes) so all is
good.

Thanks again!
Gonzalo
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