rgmhtt
1
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.
yusuf
2
According to this:
voip-info.org/wiki/index.php … anreinvite
its canreinvite=yes
on the cli, you can do a ‘rtp debug’ to confirm the media-strean flow
rgmhtt
3
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.