[OpenSIPS-Users] opensips rtpproxy

[Brad Smith]

I will give it a try. Thanks.


--- original message ---
From: "Bogdan-Andrei Iancu" <bogdan at voice-system.ro>
Subject: Re: [OpenSIPS-Users] opensips rtpproxy
Date: February 24, 2010
Time: 7:0:38  AM

Hi,

First of all , be sure that rtpproxy is called two time for each call - 
once a INVITE time, second at 200 OK time.

Put xlogs in the script check this first.

Regards,
Bogdan

bradleyd wrote:
> Hello,
>
> opensips ver: 1.6. (not svn)
> rtpproxy ver: 1.2.1 (not svn)
> Debian lenny
>
> I am trying to setup opensips with rtpproxy. I cant seem to get it to 
> work. I either have no audio or one way--depending on the changes.  We 
> are not registering any users and do not require authorization for any 
> incoming calls.  The incoming calls will go either to a set of ivr's or 
> to a set of gateway's via dialplan and dynamic routing. I can get the 
> audio/dtmf  to work if I set a src_ip conditional in on_reply route; but 
> this seems like a awful idea. I am unsure where to go from here, any 
> help would be appreciated.
> Here is a link to my config http://pastie.org/837255
>
>
> Thanks
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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