In sip.conf:
canreinvite=no
And u're done.
J
On Tue, 19 Aug 2003 18:02:18 -0500 (CDT) Brian West <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of CommuniGate(tm) Pro*
Let me try this once again. :P The reason I wanted everything to go thru
the * server is so you can monitor calls with res_monitor.
bkw
On Wed, 20 Aug 2003, Jamie Carl wrote:
Seeing as no one else has replied, I figured I may give it
a shot. At least it'll start something.
Now, correct me if I'm wrong someone, but as far as I understand in this situation you can do both. Normally the RTP packets would be swtiched through *, but you can set in you sip.conf file the 'canreinvite=yes' option which will allow the RTP stream to be direct if a compatible codec is negotiated.
I'll double check if I ever get my server up and running again.
J
On Tue, 19 Aug 2003 11:17:20 -0500
"Jorge Cisneros Flores" <[EMAIL PROTECTED]> wrote:
>Hi
>
>
> Is posible to make a call from site A to Site C, and
>my question is, the rtp data is from A to C or is from A
>to B to C
>
>
>
>
> Site A Site B
> Site C
> ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186
>
>Thanks
Regards,
Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web: www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Regards,
Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web: www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
