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

[David Ludlow]

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
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