12 apr 2006 kl. 14.58 skrev Ronald Wiplinger:
Tiago Stein D`Agostini wrote:
Hi,
Ie been looking for some time how to use asterisk to initiate
SIP connections between 2 IP phones, but afetr initiated the
communication making the RTP go directly from one telephone to the
other, without passing by asterisk. Unfortunately I found no
explanations of how to do it.
Does anyone care to give a pointer to any explanation about how to
do it?
canreinvite=yes
and look at the options for dial()
Thanks in advance
Actually, it's the default mode. Just connect your phones to Asterisk
on the same LAN, and Asterisk will
get out of the media path.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
_______________________________________________
--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
