iax.conf doesn't take canreinvite=no . It's only for sip . For iax2 its transfer=yes/no (1.4 ) or notransfer=yes (1.2) , there's one more parameter in 1.4 with which u can transfer only audio stream . Check voip-info wiki for all options .
On Thu, Mar 13, 2008 at 7:48 AM, Gonzalo Servat <[EMAIL PROTECTED]> wrote: > 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 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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