David Ludlow
2005-Feb-20 22:26 UTC
[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 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050220/0dce1168/attachment.htm
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