12 Aug
2015
12 Aug
'15
12:43 a.m.
Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already
included. Also you can use LCR for routing calls to different providers, a
simple guide can be found here
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/
2015-08-12 0:41 GMT+03:00 Alexandru Covalschi <568691(a)gmail.com>om>:
First of all I'd suggest to use
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
guide in combination with
http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html
But, assuming your platform is behind NAT, you need:
1st. Use rtpengine instead of rtpproxy. You can read about how to
advertise your external public adress on rtpengine git page.
2nd. In Kamailio configuration when you define listen, you should use
listen - advertise construction (
http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen).
3d. Be sure to leave "secret" column empty on asterisk database, otherwise
all users registered on asterisks won't have OK status, what can cause
problems with queues etc.
2015-08-12 0:19 GMT+03:00 Bruno <d4rkstar(a)gmail.com>om>:
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
Hello,
i'm on my first try with kamailio. I need to build a SIP balancer that
should keep SIP
registration from VoIP provider and route the calls to the asterisk boxes
where an IVR
will take care to answer.
Here's my network topology:
+---> [asterisk1]
[public_ip] | 10.50.10.131
[router] <---NAT---> [kamailio] <---+
10.50.10.1 10.50.10.120 |
+---> [asterisk2]
10.50.10.132
In my setup i planned to use UAC and DISPATCHER modules. I started from
the
"kamailio-basic.cfg" and added some extra lines to handle UAC and
DISPATCHER.
All is working fine when i do a test call from a softphone inside network
10.50.10.0/24.
When a call is coming from the sip carrier, troubles occurs because
asterisk boxes
are sending their internal ip in SDP.
I understand that i need to rewrite SDP in that case, but i actually
don't know how/where.
I've attached kamailio configuration and a sip trace taken with sngrep
where the problem
is visible.
For security reasons, i would like to force the RTP through RTPProxy.
I'm missing something, and need your help me to understand my errors.
Best Regards,
Bruno
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--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/