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

Sun, 20 Feb 2005 21:27:44 -0800

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 10000 to 20000 for RTP traffic.

See the following for more info:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20rtp.conf
http://www.voip-info.org/wiki-Asterisk+firewall+rules 
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to