[asterisk-users] Sending RTP media to a different server than SIP Signaling

[bruce bruce]

out* of india.

On Sun, Apr 11, 2010 at 2:26 AM, bruce bruce <bruceb444 at gmail.com> wrote:

> There you go. This confirms that SIP signaling determines where the calls
> should go. I would take their word with a grain of salt specially with their
> whole support center our of India. No disrespect, but it is bad service
> overall.
>
> -Bruce
>
>
> On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp <jcolp at digium.com> wrote:
>
>> ----- "Tarek Sawah" <tareksawah at hotmail.com> wrote:
>>
>> > we started with them two days ago .. and we are facing plenty of False
>> > Answer cases on several destinations although ppl said they have a
>> > policy against FAS..
>> > anyway i don't know i will be looking into another method to send the
>> > RTP to another server,
>>
>> The IP address (and port) of where to send audio is negotiated when
>> the call is setup. You can't change it or specify an IP address to use.
>> Even if you did change the IP address you would be sending it to the port
>> associated with the session on the other media gateway. That would just
>> not work.
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> Check us out at:  www.digium.com  & www.asterisk.org
>>
>> --
>> _____________________________________________________________________
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>
>
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