asterisk users - May 2005 - IAX and calls on hold

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 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