Ed Nuñez
2006-Dec-06 07:53 UTC
[asterisk-users] problem with asterisk - calls where both sidescannot hear each other
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 Nu?ez
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Singer Wang
Sent: Tuesday, December 05, 2006 4:23 PM
To: asterisk-users@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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Ed Nuñez
2006-Dec-06 07:56 UTC
[asterisk-users] problem with asterisk - calls where both sidescannot hear each other
Singer
I would be interested to see the rest of your configuration pertaining to how
you are recording the calls. I am having trouble with this part.
Are you using monitor or MixMonitor from extensions.conf of are you using the
queues.conf or agents.conf monitor ?
Ed Nu?ez
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Singer Wang
Sent: Tuesday, December 05, 2006 4:23 PM
To: asterisk-users@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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Singer Wang
2006-Dec-06 08:29 UTC
[asterisk-users] problem with asterisk - calls where both sidescannot hear each other
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 Nu?ez 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 Nu?ez > 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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Singer Wang > Sent: Tuesday, December 05, 2006 4:23 PM > To: asterisk-users@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 > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users