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

[Singer Wang]

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

Sorry, I forgot to add one detail. Its only happening to about 5-10% of
the calls on an average day. Most of the calls goes through properly and
both sides talk and can hear each other.




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