Hello, is Asterisk on a private IP and RTPEngine has to do bridging between public and private networks?
Getting ngrep or pcap with sip traffic on kamailio server for a call that doesn't work can help us figure out if something is not done for a proper rtp relaying. Cheers, Daniel On 15.09.17 05:44, Isravel Raja Thangamani wrote: > Hi > > > Thanks in advance if anyone can point me in the correct direction . > > I have kamailio running in front of some asterisk. SIP endpoint > register to their asterisk PBX via Kamailio dispatcher module. I'm running > rtpengine with a Wan and private interface to bridge audio stream between > these endpoints on the WAN to asterisk PBX running on LAN IP behind > Kamailio. > > Calls from ext to ext work fine audio both directions , calls outbound to > PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio > both directions. But incoming calls via SIP provider no audio from > external caller to the asterisk ext neither asterisk to external caller > > I reckon I have something wrong in my Kamailio.cfg . if I register an ext > direct to asterisk I get audio both ways on incoming calls. And rtp logs > > I think my mistake in somewhere in the cfg below. > > Do I need to handle invites from the backend asterisk servers different that > invites from sip endpoints? > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users@lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com
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