[App_rpt-users] SIP Help Needed

[Bill Hurlock]

Right after I hit send I saw that in SRC there was a 2nd rtp.conf example that listed what you have sent. I have added this to my RTP.conf file and and added these ports to my router config. I’ll be out of the house and will check later today. Thanks for your help.

Bill Hurlock

From: Robert A. Poff WB3AWJ [mailto:wb3awj at comcast.net]
Sent: Wednesday, February 12, 2014 11:10 AM
To: Bill Hurlock
Cc: app rpt-users
Subject: Re: [App_rpt-users] SIP Help Needed

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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