[Asterisk-Users] Call transfer on ATA186

[Dan]

Hi Michael,


> I meant that attended transfer doesn't work (at least for me) when I'm
trying
> to transfer call to a different device too.
It works for any other type of IP phones, except ATA186. Tested on Cisco
7960 and X-Lite.

> 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.
This is a normal behavior, as both calls where originated from ATA.

> What interesting is that asterisk
> redial my cell on itself in a second or so and then I get connected to
h323
> endpoint.
Unfortunatelly it does it in less than 1 second and  this is the reason that
ATA cannot handle attended transfers. It consider this delay as flash and
remain ini a busy state.

> 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.
This is normal. See before.

> 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.
If the redial dis done in more than 1s, then it can work on ATA too.
I think this is something very easy to be done by someone with greater
experience in the Asterisk source.

Best regards,
Dan


>
> 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
> > >
> > > _______________________________________________
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> > >
> > >
> >
> >
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