On Wed, 21 Jul 2004, Michael Wang wrote: > How do I change configuration of Asterisk so that phone B can use > aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address?
sounds like * is using reinvite to get itself out of the loop and let the phones send RTP directly between themselves. Because of the NAT, this won't work. To prevent * from sending the reinvite, and to keep RTP traffic flowing through *, try using nat=yes and/or canreinvite=no in sip.conf (you choose which section, general or phone-specific) Greg _______________________________________________ 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
