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