[Asterisk-Users] Call transfer on ATA186

[Michael Ulitskiy]

I meant that attended transfer doesn't work (at least for me) when I'm trying
to transfer call to a different device too.
Let's say I dial from ata186 another h323 endpoint. Put it on hold. Then dial my
cell phone (ata -> asterisk chan_h323 -> h323/pstn gateway). Then if I hang
up on ata the call to my cell phone drops. What interesting is that asterisk 
redial my cell on itself in a second or so and then I get connected to h323 
endpoint. If before hanging up I press flash on ata186 to have 3way conference
call, it works fine - 3 phones get connected, but then if I hang up on ata then
2 other parties don't stay connected but both get dropped. In the latter case
asterisk doesn't redial either phone.
I think I've seen in the development maillist that asterisk doesn't support
attended call transfer yet (at least on voip channels). It would be nice
if someone of the gurus confirm (or better disprove ;-)) this.

Michael

On Monday 28 July 2003 01:00 pm, Dan wrote:
> 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
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> 
> 
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