[asterisk-users] OT - How does the blind transfer function work on Snom-870?

[John Kiniston]

Take a look at two variables you can set on your SIP peer.

TRANSFER_CONTEXT  and FORWARD_CONTEXT

You should be able to use this syntax in your sip.conf

setvar=_TRANSFER_CONTEXT=kiniston-xfr

You can then create the logic you need in your dialplan to change the ring
using something like

exten => _XXX,1,SIPAddHeader(Alert-Info: Ring1)
exten => _XXX,1,Goto(INTERNAL-EXTENSIONS,${EXTEN},1)

Or you could modify your extension macro and have a test there to see if
the call has been blind transferred.with

ExecIf($[${LEN(${BLINDTRANSFER})} > 0 ]?SIPAddHeader(Alert-Info: Ring1))


On Thu, Mar 5, 2015 at 8:43 AM, James B. Byrne <byrnejb at harte-lyne.ca>
wrote:

>
> On Thu, March 5, 2015 09:56, Ruben Rögels wrote:
>
> >
> > Hi again,
> >
> > I'm glad to hear that I provided a somehow useful answer.
> >
> > Unfortunatelly, I don't know these details.
> > If you wasn't lucky consulting the snom docs, maybe the snom support
> > can be helpful with information about the exact implementation
> > details.
> >
> > You also could use "sip debug" on asterisk to check what's going on
> > when pressing the transfer button vs. what's happening when using
> > "##" via DTMF.
> >
> > Are you forced to get the transfer information from the SIP
> > signalling, or can you use AMI events for example? I think
> > this would be possible if asterisk is configured to stay in
> > the media path, so re-inviting is handled over asterisk itself
> > and therefore detectable with AMI events.
> >
>
> I am working with a FreePBX12/Asterisk11 setup.  Asterisk stays on the
> path (B2B) and there are no peer-to-peer re-invites.
>
> What I am trying to do is to get our Snom870s to use a distinctive
> ring tone when external calls are transferred internally.  I have an
> extension context override that detects the origin of calls and
> assigns a distinctive ring to each based on "${CallerIDNum}".
>
> But when a call is transferred then the tone does not change since the
> CallerIDNum does not.  An external original call always rings as if it
> were coming from the outside (which it is but transferred calls have a
> different handling procedure than unanswered calls).  I need some way
> to distinguish when the call has already been answered at least once
> without changing the CallerID.
>
> I am not worried about attended transfers since then the internal ring
> tone is what should be used and that is what happens now.  I just need
> to deal with blind transfers.
>
> What I have now is:
>
> 1. Outside call => ring1
> 2. Internal call => ring2
> 3. Transferred call => ring1 || ring2 (depending on 1 or 2)
>
>
> What I want is:
>
>
> 1. Outside call => ring1
> 2. Internal call => ring2
> 3. Transferred call => ring3 (regardless of 1 or 2)
>
>
> If everything went though ## then that would be simple enough.  The
> trick is that most (all) users employ the transfer button and the
> touch screen to forward calls using blind transfer.  But whatever
> method they use to transfer I want the transfer ring tone to be the
> same, albeit different from the one used for a new incoming call.
>
> If the transfer is done using a sip message then that should be doable
> as well.  I just have to discover what the message is.  If someone
> already knows and would care to share the information then that would
> be helpful.  Otherwise wireshark and debug will eventually reveal it.
>
> I may not know what I am doing. But, at least I know that I do not
> know what I am doing.
>
> --
> ***          E-Mail is NOT a SECURE channel          ***
> James B. Byrne                mailto:ByrneJB at Harte-Lyne.ca
> Harte & Lyne Limited          http://www.harte-lyne.ca
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>
>
> --
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