Hi!

> In the [general] section of sip.conf I have :
> 
> disallow=all
> allow=g726
> allow=alaw
> allow=g729
> allow=gsm

So change the order there and see what happens.

> > * look at the variable SIP_CODEC for the inbound (first) call leg, and
> > in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
> 
> When I read the value of this variable just before the Dial()-statement,
> it is empty.

You need to set it, not read it.

Philipp


-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to