[App_rpt-users] SIP Help Needed

[Robert A. Poff WB3AWJ]

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.html 
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. 

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