canreinvite=yes
Aterisk stays in the signaling path so unless you're running tcpdump or the
like you'll never notice this.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of bijan
Sent: Thursday, December 23, 2004 4:46 PM
To:
In wiki pages it is stated that The audio channels (RTP) may go directly
from phone to phone or may go through Asterisk's media bridge.
Currently with my settings, I notice that all rtps are passing through
my asterisk. How could I achieve that they go directly from phone to
phone? I
Look at canreinvite= in the sip.conf.
If you remove Asterisk from
the stream them you are using Asterisk more like a Proxy and less like a PBX.
If this is the case and you want to support tons of users look at
something like SER. Asterisk is not a Sip proxy but rather a PBX and Media
