I'm pretty sure the info has been posted to the mailing list several times and should be in the searchable archives.
On Wed, 2003-09-10 at 14:28, Peter Pauly wrote: > On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote: > > That would be reinvite= and canreinvite= in the user entry for each SIP > > endpoint. Asterisk will allow the endpoints to talk directly to each > > other if both those settings are = yes (the default, I think) AND both > > endpoints use the same protocol (SIP) AND the same codec. > > > > This is the single most useful bit of info I have seen > on the mailing list since I have joined. Thanks Mr. Wieling. > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
