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 <jc...@digium.com> wrote: > ----- "Tarek Sawah" <tareksa...@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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
