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
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@gmail.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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@gmail.com:
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@gmail.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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
Thankyou Alexandru for your suggestions. I'll give it a try tomorrow and will report my progress here. It seems that i'm not so far from the result! Bruno
Il giorno mar 11 ago 2015 alle 23:44 Alexandru Covalschi 568691@gmail.com ha scritto:
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@gmail.com:
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@gmail.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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Alexandru Covalschi -
Bruno -
Bruno Salzano