[Asterisk-Users] IAX and calls on hold

[Sean Kennedy]

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