out* of india. On Sun, Apr 11, 2010 at 2:26 AM, bruce bruce <bruceb...@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 <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 >> > >
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