Here's what is in mine verbatim. Ah....... I just looked into an ACID install of mine. Didn't see rtp.conf there at all. You may need to create it. When I first starting to add extensions, I used the Asterisk book as a guide. Very helpful. Here's the online version : http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.ht... It covers Asterisk 1.8, but most of the config stuff still applies to 1.4 I find. Anyhow, here's what I have in rtp.conf on my node that has SIP phones attached: ; ; RTP Configuration ; [general] ; ; RTP start and RTP end configure start and end addresses ; These are the addresses where your system will RECEIVE audio and video stream$ ; If you have connections across a firewall, make sure that these are open. ; rtpstart=10000 rtpend=10999 And here's a link to a page concerning rtp.conf : http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf Hope this helps.