<div>Thankyou Alexandru for your suggestions. <br>I'll give it a try tomorrow and will report my progress here.<br>It seems that i'm not so far from the result! <br>Bruno<br></div><br><div class="gmail_quote"><div dir="ltr">Il giorno mar 11 ago 2015 alle 23:44 Alexandru Covalschi <<a href="mailto:568691@gmail.com">568691@gmail.com</a>> ha scritto:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">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 <a href="http://dopensource.com/least-cost-routing-with-kamailio-v4-1/" target="_blank">http://dopensource.com/least-cost-routing-with-kamailio-v4-1/</a><br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-08-12 0:41 GMT+03:00 Alexandru Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>First of all I'd suggest to use <a href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb" target="_blank">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a> guide in combination with <a href="http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html" target="_blank">http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html</a><br>But, assuming your platform is behind NAT, you need:<br></div>1st. Use rtpengine instead of rtpproxy. You can read about how to advertise your external public adress on rtpengine git page. <br></div>2nd. In Kamailio configuration when you define listen, you should use listen - advertise construction (<a href="http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen" target="_blank">http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen</a>).<br></div><div>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.<br></div><div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>2015-08-12 0:19 GMT+03:00 Bruno <span dir="ltr"><<a href="mailto:d4rkstar@gmail.com" target="_blank">d4rkstar@gmail.com</a>></span>:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><br clear="all">Hello,<br>i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP<br>registration from VoIP provider and route the calls to the asterisk boxes where an IVR<br>will take care to answer.<br><br>Here's my network topology: <br><br><span style="font-family:monospace,monospace"> +---> [asterisk1]<br>[public_ip] | 10.50.10.131<br> [router] <---NAT---> [kamailio] <---+ <br>10.50.10.1 10.50.10.120 | <br> +---> [asterisk2]<br> 10.50.10.132</span><br><br>In my setup i planned to use UAC and DISPATCHER modules. I started from the <br>"kamailio-basic.cfg" and added some extra lines to handle UAC and DISPATCHER.<br><br>All is working fine when i do a test call from a softphone inside network <a href="http://10.50.10.0/24" target="_blank">10.50.10.0/24</a>.<br><br>When a call is coming from the sip carrier, troubles occurs because asterisk boxes<br>are sending their internal ip in SDP.<br><br>I understand that i need to rewrite SDP in that case, but i actually don't know how/where.<br><br>I've attached kamailio configuration and a sip trace taken with sngrep where the problem<br>is visible.<br><br>For security reasons, i would like to force the RTP through RTPProxy. <br><br>I'm missing something, and need your help me to understand my errors.<br><br>Best Regards,<br>Bruno<br><br><br>
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<br></blockquote></div><span><font color="#888888"><br><br clear="all"><br>-- <br><div><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
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</blockquote></div><br><br clear="all"><br>-- <br><div><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: +37367398493<br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
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