[asterisk-users] terminating the call, when transferer hangs up the call during attended transfer
Nikola Ciprich
nikola.ciprich at linuxbox.cz
Mon Nov 25 08:34:43 CST 2013
On Mon, Nov 25, 2013 at 03:23:05PM +0100, jg wrote:
> I think I got it now:
>
> While A is on hold,
> B dials C
> after a few seconds B wants to stop dialing C and hangs up,
> "the system" calls back and B is connected again with A
yes, I think this is what is required...
>
> Is that correct?
>
> If (yes) {
> I see a logical problem
I THINK this makes a bit of sense, (A party starts ringing
on B again), even though it's quite ridiculous request :(
in principle, I need to DISABLE blond transfer function
and force the Dial to C party to be terminated, so I can
ring back to B..
do You think there is some way to achieve this?
n.
> } else {
> please be more specific about the events
> }
>
> jg
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
-------------------------------------
Ing. Nikola CIPRICH
LinuxBox.cz, s.r.o.
28.rijna 168, 709 00 Ostrava
tel.: +420 591 166 214
fax: +420 596 621 273
mobil: +420 777 093 799
www.linuxbox.cz
mobil servis: +420 737 238 656
email servis: servis at linuxbox.cz
-------------------------------------
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 198 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131125/46740066/attachment.pgp>
More information about the asterisk-users
mailing list