Hi,
The tutorial you mentioned is for integration Asterisk as a media server
(for VM or conf services).
I suppose you want to receive all calls into OpenSIPS and then have them
forwarded to Asterisk ? If so:
1) be sure you get the call into OpenSIPS
2) for initial calls, identify the RURI you need to re-route to Asterisk
3) use t_relay() to send them over to asterisk.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 27.01.2015 14:32, mahan77 wrote:
Hi need some help’
I’m playing around with openSIPS and asterisk in same server. I was flowing
this link
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration
What I’m trying to do, wen I call the incoming sip number I will get the
calls in asterisk sip.conf internal context, then I will able to route the
call. But if I run openSIPS in front of asterisk I cant get any calls. Its
look like simple routing scripts in openSIPS but I can’t figure out how to
do it? Any help please.
openSIPS running 5060 port
asterisk running 5080 port
;this is my sip.conf
[general]
allowguest=yes
maxexpirey=3600
defaultexpirey=3600
port=5080
bindaddr=192.168.1.150
nat=no
disallow=all
allow=alaw,ulaw
context=internal
allowoverlap=no
language=en
dtmfmode=info
rtcachefriends=yes
Many thanks
Sathees
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