On Mon, 13 Dec 2004 11:40:46 +0000, Andy Burns <[EMAIL PROTECTED]> wrote: > If I have inbound SIP calls arriving from a provider's gateway to an > asterisk server on my LAN, which then routes the call back out via the > provider's gateway to a PSTN number, once the call is answered do all > the voice packets pass through my asterisk PBX, or is SIP intelligent > enough to patch the two PSTN ends of the call direct to each other going > only via two ports on the provider's gateway?
The data-heavy portion of the traffic is RTP, and that should be a direct connection using your providers gateway. Make sure you have 'canreinvite=yes' set in the appropriate section of your sip.conf. -- Sam Bashton _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
