Direct communication

I have searched widely for this information without success.

I want to get an ATA to place a call to another ATA (both SIP registered), and then have the call go directly between the 2 ATAs.

I have been told to set REINVITE=NO to do this. But I have done that both in Asterisk (I am running Trixbox) and the ATAs (PAP2s and HT386/488s).

I have to two ATAs on the same subnet as the server and a system setup with Ethereal, so I know that the RTP flow remains handled by the server.

What do I need to do this. I cannot afford for 2 remote people to have to route their calls through my server and WAN link.

According to this: … anreinvite

its canreinvite=yes

on the cli, you can do a ‘rtp debug’ to confirm the media-strean flow

It is not working.

I see the canreinvite=yes with SIP SHOW PEER nnnn

But I do not see it in the actual SIP exchange.

Further your method of checking: RTP DEBUG should not work, as the goal here is for the server to NOT receive any RTP packets.

Ethereal shows the exchange as a 3-way.

show us the Dial string you’re using. any transfer options or call recording, and Asterisk won’t hand the call off.