Hey Vincent -
1. What are the settings (in sip.conf?) to tell Asterisk to use specific ports for RTP?
I guess you didn't see my reply earlier today - that setting is in rtp.conf
2. With this kind of setup, does Asterisk stay in the loop to forward RTP packets, or do X-Lite at home and the VoIP provider send RTP to each other directly?
That all depends on your "canreinvite" setting (in sip.conf, per peer or user). If set to no, then asterisk will stay in the picture throughout your call. If set to yes, the call will be passed off so the rtp traffic goes directly from X-lite to the VoIP provider. Keep in mind that if you do have it set to yes, the VoIP provider may not necessarily use the same rtp ports that you want to use. - Noah _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
