Hi

Thanks in advance if anyone can point me in the correct direction .

I have kamailio running in front of some asterisk VM’s.  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 I only get audio on stream from asterisk registered ext to external caller , no audio from external caller to the asterisk ext.

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 from rtpenegine show it as trying to send the rtp to the private address of the sip endpoint rather that its WAN address.

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?

Gerry Kernan

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