[Asterisk-Users] Call transfer on ATA186

[Dan]

Hi,

It works, bot ONLY when I try to transfer the call to another type of phone,
like X-Lite or Cisco 7960.
If the destination is an ATA too, it does not work because hanging-up is
considered as a closed call only after 1 second in ATA (if less than 1s, the
it is a flash function), but the transfer function in Asterisk tries to
recall the first extension in less than 1 second, so during this short
period of time, ATA based phone is bussy and cannot accept calls, so the
call is redirected to the voicemail.
One way to make this attended transfer work with ATA too, is to enter a
minimum delay of 1 second in th transer function, but I don't know how to do
it.

Look at the ATA186 specification for extended SIP functions, at the address:
http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataadmn/ata88sip/supp.pdf
or as HTML ast:
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html

It is stated that the attended transfer is done like that:

Step 1   Press the flash button on the telephone handset to put the existing
party on hold and get a dial tone.
Step 2   Dial the telephone number to which the existing party is being
transferred.
Step 3   When the callee answers the phone, you may consult with the callee
and then transfer the existing party by hanging up your telephone handset.

It works for me on ATA if the final destination is not an ATA too.

Best regards,
Dan
P.S. I'm interested in the attended transfer. The unattended one works
perfect.


----- Original Message ----- 
From: "Michael Ulitskiy" <mulitskiy at acedsl.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, July 28, 2003 7:41 PM
Subject: Re: [Asterisk-Users] Call transfer on ATA186


> On Monday 28 July 2003 12:24 pm, Dan wrote:
> > Hi Iain,
> >
> > > The basic call transfer functions, set with the T and t options to the
> > dial
> > > application and triggered by pressing a # work fine for me.
> > I have T and t options in dial application, but how can '#' be used for
> > transfer.
> > Escuse my ignorance...
> >
> > > Make sure that
> > > you have set the DialPlan on the ATA 186 so as not to grab the # (ie
look
> > > for any ># character pairs and change the second character or remove
it).
> > Where to do that? In the extensions.conf file?
> >
> > Now I have used Flash key to put the other part on hold and then dial to
the
> > new extension and after this one answer, I close the phone.
> > It works in that way only if the last party is anything else, but not
> > another ATA186.
>
> Does it really work this way for you? I thought asterisk cannot bridge
together
> 2 channels if originating party hangs up. I mean if I press flash button
to put
> one party on hold, then dial another extension and then hang up the two
other
> extensions do not get connected but both calls get dropped. Only "blind
transfer"
> with # key works for me.
> If it really works for you, would you mind to show your configuration?
> Thanks.
>
> Michael
>
> > Thanks for your support,
> > Dan
>
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