[asterisk-users] problem with asterisk - calls where both sidescannot hear each other

[Singer Wang]

I have bindaddr=0.0.0.0 in my sip.conf; what my major problem is that it
only happens 5-8% of the time.. 

On Wed, 2006-12-06 at 09:56 -0500, Ed Nuñez wrote:
> If you use both the public and private interfaces for VoIP in the Asterisk Server, make sure you don't specify one of them for the binding in sip.conf
> 
> Example
> 
> bindaddr=0.0.0.0
> 
> will allow SIP traffic on any of your interfaces.
> 
> 
> 
> Ed Nuñez
> IT/Telecom Engineer
>  
> 4037 Metric Drive
> Winter Park, FL
>  
> (o) 407-384-4200 x 1656
> (f) 407-384-4222
> (c) 732-925-0730
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Singer Wang
> Sent: Tuesday, December 05, 2006 4:23 PM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] problem with asterisk - calls where both sidescannot 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
> 
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