asterisk users - May 2005 - Problem with calls on hold

Hello -

I recently offloaded some of the SIP traffic on to a separate 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. There is usually at least a delay and distortion before voice
works in both directions again. Also, when this happens, sometimes I can hear
calls that seem to bleed over from other active IAX - SIP sessions. Music on
hold is enabled and the other end can hear the music; however, when the call is
pulled off, the other end either still hears music or does not hear anything for
a short while.

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 have tried both enabling and disabling IAX trunking and jitterbuffers.
I've also added a zap card (fxo) and enabled it to allow for a timing
source.

The new system is running the latest CVS of Asterisk and libraries as of a
couple days ago, 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