Hi Daniel

 

Thanks for your help on this , changing direction attr on rtpengine_offer fixed 
my issues

 

 

Best Regards

 

Gerry Kernan

 

From: sr-users [mailto:[email protected]] On Behalf Of 
Daniel-Constantin Mierla
Sent: 19 April 2017 10:28
To: Kamailio (SER) - Users Mailing List <[email protected]>
Subject: Re: [SR-Users] Kamailio rtpengine sdp

 

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

 



 

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