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

Ruben Rögels ruben.roegels at jumping-frog.org
Thu Mar 5 08:56:25 CST 2015

Am 05.03.2015 um 15:09 schrieb James B. Byrne:
> On Thu, March 5, 2015 05:30, Ruben Rögels wrote:
>> Am 05.03.2015 um 01:09 schrieb James B. Byrne:
>>> I am trying to determine how the transfer button on the Snom-870
>>> works
>>> with Asterisk.  Is the ## special code employed as when it is
>>> entered
>>> through the handset or is the blind transfer through the phone
>>> function accomplished in a different fashion?
>> Hi,
>> I hope I understood your question correctly.
>> AFAIK, the transfer button sends a SIP message.
>> Entering "##" on the handset is recognized via DTMF by asterisk.
> I hope that I understood what I was asking for.  Sometimes I do not.
>   Yes, that is what I wanted to know.  Does the implementation of the
> transfer button feature on the Snomp-870 use exactly the same
> technique as the ## feature code entered through the dial pad and
> produce exactly the same SIP message that Asterisk produces when it
> gets the ## DTMF?
> The reason is that I wish to be able to detect a call transfer
> performed via either method (## or <Transfer-Button>) and process the
> result of both in the same fashion. If the button and DTMF transfers
> are not performed using the same switching techniques in Asterisk then
> I need to discover what those differences are.  If both are totally
> equivalent from a SIP message signalling point of view then the issue
> is far easier to handle.
> I searched, in vain, in the Snom-870 docs for specifics on this and
> either could not find or did not recognize anything that applied.  Do
> you know where I can locate these sorts of details.  My knowledge of
> SIP/RTP/VOIP etc. is cursory but, given an adequate reference, I can
> usually sort things out.

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 signaling,
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.


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