[Asterisk-Users] SIP to SIP calls have no audio until put on hold and taken back off

A previous poster mentioned the same thing, with no response: http://lists.digium.com/pipermail/asterisk-users/2004- December/080161.html Fresh asterisk 1.0.5 install on FC3, started with make samples, nothing fancy. It's so bland, I'm surprised the list isn't full of people having the same

[Asterisk-Users] SIP to SIP calls have no audio until put on hold and taken back off - SOLVED

Thanks to Pau (the original person to pose the question on this list), it's fixed. The firewall was getting in the way. I needed to open up UDP ports 1 to 2 for RTP traffic. See the following for more info: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20rtp.conf