asterisk users - Dec 2006 - problem with asterisk - calls where both sides cannot hear each other

Hi,

I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:

I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.

We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.

I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked. 


My extensions configuration is roughly the following:

[opened]
exten => s,1,SetVar(LOOP=1)
exten => s,2,Answer
exten => s,3,Wait(1)
exten => s,4,Background(open-hiq)
exten =>
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten => s,6,Queue(support||||3600)
exten => s,7,Voicemail(100|us)

exten => 1,1,Goto(opened,s,6)

exten => 500,1,Voicemail(500)


thanks,
Singer Wang

On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang
wrote:
> Hi,
> 
> I'm looking for some help with a problem in Asterisk (possibly), and
I'm
> confused as heck what is going on. I've updated to the latest Asterisk
> version and the problem is still occur. My setup is as follows:
> 
> I've got Asterisk running on a high end Pentium-IV box running Linux
> serving 5 calls, it is located in Canada. The calls come in via analog
> lines through TDM400P cards to Asterisk box, which then converts it to
> G729 channels to a call center in India over the Internet. Connection
> between the Asterisk Server and the India call center is done via two
> Cisco PIX501 devices, The call center in India is running 5 agents using
> PolyCom phones, and we're using G729 to save bandwith. And yes, we
> purchused 5 licenses of G729 codec.
> 
> We're using SIP and a ring all strategy, with the first agent that
picks
> up getting the call. The problem we're having is that about 5-10% calls
> are not connecting properly. In that both sides can talk but do not hear
> each other. Since we have recording in step s,5 (in the configuration
> below), I can verify that it is happening. In these problematic calls,
> both sides of the call talk but they cannot hear the other side at all.
> 
> I've gone through most of the documentation and spend hours on Google
> search, does anyone have any idea what could be the problem? I'm
willing
> to provide more information if asked. 
> 
> 
> My extensions configuration is roughly the following:
> 
> [opened]
> exten => s,1,SetVar(LOOP=1)
> exten => s,2,Answer
> exten => s,3,Wait(1)
> exten => s,4,Background(open-hiq)
> exten =>
>
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
> exten => s,6,Queue(support||||3600)
> exten => s,7,Voicemail(100|us)
> 
> exten => 1,1,Goto(opened,s,6)
> 
> exten => 500,1,Voicemail(500)
> 
> 
> thanks,
> Singer Wang
> 
Have you made sure there isn't a firewall in the way that could be blocking
your audio?  You might need to punch some holes through to allow your RTP
stream.

--
Kyle Sexton