[Asterisk-Users] IAX and calls on hold
Jeroen Moetwil
darkstar at linuxforge.net
Wed May 11 11:19:54 MST 2005
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
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