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

[Florian Overkamp]

Hi, 

Response below.

In the meantime: I would REALLY appreciate comments from an ATA186 SIP user
who can tell me:

- How to transfer a call without using #-transfer
- Preferably more or less like how we are used to transferring in a classic
pbx system

Noteworthy:
- Which Asterisk version (CVS/CVS-HEAD/...)
- Which ATA186 firmware

Thanks, 

Florian

> -----Original Message-----
> 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

I have not found any setting on the ATA that can make such a difference in
approach.

> > -----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/0
18388.html
> > 
> > Is this:
> >
> http://lists.digium.com/pipermail/asterisk-users/2003-August/0
18414.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