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?

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