I am trying to setup asterisk as a registrar and sip server only.
Currently When I make calls all my rtp traffic is going through the asterisk
server as a B2BUA.
Is it possible to turn off this feature and have all my calls RTP traffic going
directly to the SIP
On Jan 5, 2008 4:40 AM, ameel [EMAIL PROTECTED] wrote:
I am trying to setup asterisk as a registrar and sip server only.
Currently When I make calls all my rtp traffic is going through the
asterisk server as a B2BUA.
Is it possible to turn off this feature and have all my calls RTP traffic
Hey Srs.
I have a little problem with the next scenario:
Internal Phone(801)--Asterisk(public IP) --INTERNET--ADSL
Router--Budgetone(716)
|--ADSL Router--Budgetone(717)
My sip.conf is the next:
[general]
port = 5060 ; Port to bind to
bindaddr =
