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 <568...@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 <d4rks...@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/
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