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: 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