It is not clear how exactly g729 pass-through can be enabled. I have a SIP call off a gateway come into an Asterisk menu, and then I send the SIP call to another SIP gateway using Dial(). Even though codec preferences have g729 listed first, it never gets used.
Both gateways have separate peer entries in sip.conf, and both have canreinvite=yes set. Can Asterisk change the media type during a re-Invite? The call is answered as g711u initially, and then Asterisk plays a menu, and then does a Dial(). I can see Asterisk doing the reInvite, but the protocol stays at g711u. Tom _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
