Hi, I cannot get SIP channel working with folowing codec configuration:
[sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others
Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them.
When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec and is trying transalate g723.1 to alaw, instead of choose ilbc and translate to alaw.
Thanks in advance
Petr Michalek _______________________________________________ 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
