Losing RTP on Call forward From external to another External #

Trying to get a rollover call forward working on a very old asterisk 1.6 server. Going though a Linux router gateway to a private WAN the providers SIP server is on. Normal calls in and out are fine. Call forwarding from an internal extension out is fine. But if a call goes from external > internal > external I get no audio. Signaling is working as the call rings but no sound. I’ve tried various combinations of nat, nonat, etc. Not in a position to upgrade at this moment as that would require new hardware, OS etc for this heartbeat cluster PBX.

I had this problem very often years ago with older versions of asterisk. I never really found out why but what did the trick was to answer the incoming call with Answer in the dialplan and play a little sound file with Playback, even if it’s just one second of silence. These two steps before the outgoing Dial is done.

It seems that answering the call and sending a little sound then makes the audio go through.

As I had been banging my head on this since yesterday morning, just after I posted I found the problem. Had [ t,T ] (allow transfer) in the Dial command for the call forwarding to the on call cell phone. This disables REINVITEs, hence why the RTP traffic wasn’t rerouted. Tested OK, hope this holds.

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.