On Wed, Mar 12, 2008 at 12:58 PM, Brent Davidson <<a href="mailto:brent@texascountrytitle.com">brent@texascountrytitle.com</a>> wrote:<br><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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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.<br>
</div></blockquote><div><br>Brent,<br><br>Thank you veeeery much for replying. I thought the message went unseen but found your reply when I went to look at the thread :)<br><br>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.<br>
<br>Thanks again!<br>Gonzalo</div></div>