[Asterisk-Users] ATA186 v3.1 SIP - Attended transfer: NO JOY

[Florian Overkamp]

Hi,

I'm still hassling with the consultative/attended transfer stuff. Someone
please help me identify this

A lot has already been said about the ATA186. Some report it works fine,
others say it doesn't. Lets get clarity on this.

My scenario is reasonably simple (I think)
Phone A: SIP/video1
Phone B: SIP/werkkamer
Phone C: IAX2/provider

Phone A calls phone B, they chat:
*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data
SIP/werkkamer-91f5  (from-werkkamer              1   )      Up Bridged Call
SIP/video1-e2a0
SIP/video1-e2a0  (pbx        1202         1   )      Up Dial
SIP/swiss&SIP/snom&SIP/werkkamer&IAX2/florian&SIP/video1
2 active channel(s)

Phone B hits flash and gets a dialtone. Dials a number and connects to phone
C:
*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data
IAX2[172.28.8.8:4569]/7  (           s            1   )      Up Bridged Call
SIP/werkkamer-2507
SIP/werkkamer-2507  (pbx        4307076      2   )      Up Dial
IAX2/provider/4307076
SIP/werkkamer-91f5  (from-werkkamer              1   )      Up Bridged Call
SIP/video1-e2a0
SIP/video1-e2a0  (pbx        1202         1   )      Up Dial
SIP/swiss&SIP/snom&SIP/werkkamer&IAX2/florian&SIP/video1
4 active channel(s)

Phone A now hears music on hold. Phone B and C can chat.

Phone B now hits flash again. All phones end in a three-way conversation:
*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data
IAX2[172.28.8.8:4569]/7  (           s            1   )      Up Bridged Call
SIP/werkkamer-2507
SIP/werkkamer-2507  (pbx        4307076      2   )      Up Dial
IAX2/provider/4307076
SIP/werkkamer-91f5  (from-werkkamer              1   )      Up Bridged Call
SIP/video1-e2a0
SIP/video1-e2a0  (pbx        1202         1   )      Up Dial
SIP/swiss&SIP/snom&SIP/werkkamer&IAX2/florian&SIP/video1
4 active channel(s)

Now the misery starts: If Phone B wants to back out of the conversation, it
seems phones C and A are also disconnected. 

I've tried doing this with SIP firmwares, 2.15, 2.16, 3.0 and 3.1 and CVS
HEAD as of today.

Other people have claimed success:
http://lists.digium.com/pipermail/asterisk-users/2003-August/018388.html

Is this:
http://lists.digium.com/pipermail/asterisk-users/2003-August/018414.html
also related ?

By the way, canreinvite=no as suggested by Mark in one of the slightly
related conversations on bugs.digium.com does not help...

I would really _love_ to know why this is and to see it fixed somehow. A
bounty would be in order. Can anyone comment on this ??

On a related note: If the consultation ends in a failure (user unavailable
or unable to talk) the way to back out is hitting flash once if the remote
hung up (ata doesn't give any tone at that time??) or twice if you got
voicemail. The remote (phone A) briefly hears this, as the first flash opens
a three-way conversation with phones A, B and the voicemail. The second one
then disconnects the voicemail again. Not really elegant (albeit useable).
Is there a better way ?

Best regards,
Florian