On Wed, Mar 12, 2008 at 12:58 PM, Brent Davidson < [EMAIL PROTECTED]> 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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