Jeroen Moetwil wrote:
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
Jeroen,
I am by no means a guru, so take what I saw with a healthy sized grain of salt. You say you have two * boxes, connected with IAX. Are they both on the same subnet?
My thoughts are this: The first box is trying to directly establish a route to the sip device, bypassing the SIP concentrator ( the second * box ).
Again, I am probably wrong, but that's the only thing I can think of that would cause problems. The only times I've had problems with SIP and one way audio was across a vpn/nat system, so that might be something you have to take into account as well. In fact, now that I am thinking about it, if you haven't already I'd check the sip.conf file and make sure the bind address is correct.
Hope some of that helped a little bit.
Sean _______________________________________________ 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
