[asterisk-users] Asterisk not transcoding between installed codecs

Jaswinder Singh vicky.r at gmail.com
Sun Mar 16 15:27:44 CDT 2008


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 <gservat at gmail.com> wrote:

> 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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