Hello,

you have to instruct rtpengine to do bridging between the two network
interfaces.

Can you show the INVITE and the 200ok with all SDPs, both sides
(incoming and outgoing to/from kamailio),  for the case you get audio
problem? Then we can confirm if the SDP has been updated properly for
bridging.

Cheers,
Daniel


On 18.04.17 17:23, gerry kernan wrote:
>
> 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*
>
>  
>
> cid:[email protected]
>
>  
>
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