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

[Steve Dolloff]

I have a similar issue with Sipura using compact headers, but not with
regular headers.  I am working on reproducing with the latest CVS.
Maybe you are using compact SIP headers on your ATA186?

http://bugs.digium.com/bug_view_page.php?bug_id=0001843

Stephen 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Florian Overkamp
> Sent: Wednesday, June 16, 2004 12:20 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] ATA186 v3.1 SIP - Attended transfer: NO JOY
> 
> 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
> 
> 
> 
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