[Asterisk-Users] Endpoint-to-Endpoint RTP Packets

Eric Wieling eric at fnords.org
Wed Sep 10 12:56:30 MST 2003


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
> Asterisk-Users at lists.digium.com
> 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)




More information about the asterisk-users mailing list