hello everybody, i want to configure a sip trunk between my system which has asterisk 11.5.1 and a cisco router. this is my scenario:
Freepbx-----my system-----cisco-router----Freepbx my system acts like a router. if i set just one codec in dial-peers on cisco router, every thing is ok and i can make a call. but if i set different codecs in a voice class codec and assign it to dial-peers in cisco router, i can not make calls. if i change my scenario like: Freepbx------cisco-router------Freepbx calls are succeed without any problem. Freepbx are asterisk-base too, so i think something is wrong in my system (my asterisk configuration is not correct or something is missing). any body knows how should i fix this problem? any comments or hints are really appreciated. P.S: my sip.conf: [peer-1] host=X.X.X.X type=peer context=from-trunk allow=all qualify=yes insecure=port,invite [peer-2] host=Y.Y.Y.Y type=peer context=from-trunk allow=all qualify=yes insecure=port,invite
-- _____________________________________________________________________ -- 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
