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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Singer Wang > Sent: Tuesday, December 05, 2006 4:23 PM > To: [email protected] > 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 _______________________________________________ --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
