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 <568...@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 <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/ > _______________________________________________ > 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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