Hello -
I recently offloaded some of the SIP traffic on to a seperate Asterisk box and interconnected our main Asterisk system with the new system via IAX. The SIP clients are running 7960's. When a call is put on hold, often times when the call is pulled off hold, there seems to be no RTP in at least one direction. There seems to only be voice in one direction.
Basically the call comes in via a ZAP channel over a PRI into our main system, is fed over IAX to our second system and then is connected to the SIP channel (client).
I've tried both enabling and disabling IAX trunking and jitterbuffers. I've also added a zap card and enabled it to allow for a timing source.
The new system is running the latest CVS of Asterisk and libraries as of yesterday, while the other one is running a CVS version as of Jun of last year. I'm using RSA for auth between the servers (IAX).
Any help would be appreciated. Thanks.
Jeroen _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
