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.<div>
<br></div><div>-Bruce<br><br><div class="gmail_quote">On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp <span dir="ltr"><<a href="mailto:jcolp@digium.com">jcolp@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class="im">----- "Tarek Sawah" <<a href="mailto:tareksawah@hotmail.com">tareksawah@hotmail.com</a>> wrote:<br>
<br>
> we started with them two days ago .. and we are facing plenty of False<br>
> Answer cases on several destinations although ppl said they have a<br>
> policy against FAS..<br>
> anyway i don't know i will be looking into another method to send the<br>
> RTP to another server,<br>
<br>
</div>The IP address (and port) of where to send audio is negotiated when<br>
the call is setup. You can't change it or specify an IP address to use.<br>
Even if you did change the IP address you would be sending it to the port<br>
associated with the session on the other media gateway. That would just<br>
not work.<br>
<br>
--<br>
Joshua Colp<br>
Digium, Inc. | Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
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