On 03:10, Sat 04 Mar 06, Tzafrir Cohen wrote: > On Tue, Feb 28, 2006 at 05:25:40PM -0700, Damon Estep wrote: > > Try nat=yes and qualify=yes in sip.conf. > > So a call between two SIP phones will have to go through the remote > server? Or can those two phones be aware of each other?
Yes. But without this things will not work. What you can do is put a local asterisk in there to do the routing of internal numbers so reinvites work again. Or you can use something like SER to do the outbound routing -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?" _______________________________________________ --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
