[asterisk-users] problem with asterisk - calls where both sides cannot hear each other

[Singer Wang]

Okay, a bit more information..

Some more information:

the non connected problem only happens to about 5-10% of the calls, the
others go through properly.. and yes, for the rest both parties can talk
and hear each other..


asterisk version:
Asterisk 1.2.13 built by root @ [hostname] on a i686 running Linux on
2006-11-14 16:53:46 UTC

We get about 50-60 calls a day with five agents.., on busy days maybe
80-100 calls

I'm setup as a call center, people call in and their calls are routed in
from analog lines through the TDM400Ps. they are connected to agents in
India via VoIP who use PolyCom IP Phones, we're using a RingAll
strategy. we have a secure IPSec tunnel between the Canada/India via
Cisco PIX501Es..


the Asterisk server has both an public (for web interface to the logs)
and private IP (10.x.x.x) and the phones have private IPs (10.x.x.x),
the traffic between them is tunneled via Cisco PIX501s.. there isn't any
NATing going on between the Asterisk and the phones..



On Tue, 2006-12-05 at 15:32 -0600, Kyle Sexton wrote:
> 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
> 
> 
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