[Asterisk-Users] Call park on SIP phones

[Andrew Joakimsen]

Because otherwise the parked extension will not be announced.


> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Brian West
> Sent: Tuesday, October 07, 2003 9:03 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Call park on SIP phones
> 
> Yes but you can't do native sip tranfers to parking.  Thats what I
want.
> And thats what I was talking about.  You can't say use a Cisco 7960
and
> hit transfer then dial 700 then transfer.  WONT WORK.
> 
> bkw
> 
> On Tue, 7 Oct 2003, Andrew Joakimsen wrote:
> 
> > You need to enable transfer:
> >
> > Dial
> > Dialing Application - Place an call and connect to the current
channel
> >
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][
> > |URL]): Requests one or more channels and places specified outgoing
> > calls on them. As soon as a channel answers, the Dial app will
answer
> > the originating channel (if it needs to be answered) and will bridge
a
> > call with the channel which first answered. All other calls placed
by
> > the Dial app will be hunp up If a timeout is not specified, the Dial
> > application will wait indefinitely until either one of the called
> > channels answers, the user hangs up, or all channels return busy or
> > error. In general, the dialler will return 0 if it was unable to
place
> > the call, or the timeout expired. However, if all channels were
busy,
> > and there exists an extension with priority n+101 (where n is the
> > priority of the dialler instance), then it will be the next executed
> > extension (this allows you to setup different behavior on busy from
> > no-answer). This application returns -1 if the originating channel
hangs
> > up, or if the call is bridged and either of the parties in the
bridge
> > terminate the call. The option string may contain zero or more of
the
> > following characters:
> > ***'t' -- allow the called user transfer the calling user*** OR
> >
> > ***'T' -- to allow the calling user to transfer the call.***
> >
> > 'r' -- indicate ringing to the calling party, pass no audio until
> > answered.
> >
> > 'm' -- provide hold music to the calling party until answered.
> >
> > 'd' -- data-quality (modem) call (minimum delay).
> >
> > 'c' -- clear-channel data call (PRI-PRI only).
> >
> > 'H' -- allow caller to hang up by hitting *.
> >
> > 'C' -- reset call detail record for this call.
> >
> > 'P[(x)]' -- privacy mode, using 'x' as database if provided.
> > In addition to transferring the call, a call may be parked and then
> > picked up by another user. The optionnal URL will be sent to the
called
> > party if the channel supports it.
> >
> >
> >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-
> > > admin at lists.digium.com] On Behalf Of Juan J. Sierralta P.
> > > Sent: Tuesday, October 07, 2003 6:46 PM
> > > To: asterisk-users at lists.digium.com
> > > Subject: RE: [Asterisk-Users] Call park on SIP phones
> > >
> > > On Tue, 2003-10-07 at 18:23, Andrew Joakimsen wrote:
> > > > How are you transfering to 700? You dial # while in a call and
then
> > it
> > > > says "transfer" and you then dial 700, or are you using a
different
> > > > method?
> > >
> > > 	If I dial # while in a call nothing happens. I was transfering
> > using
> > > the 7960 transfer function which gives me a dial tone and then I
dial
> > > 700 which gives me a busy tone I also tried to dial #700 but as
soon
> > as
> > > you push # on a 7960 it dials since # its used to signal the end
of
> > the
> > > dial string.
> > >
> > > > >
> > > > > 	I still cannot park calls on my 7960, I have:
> > > > >
> > > > > ----- extensions.conf -------
> > > > > [demo]
> > > > > ; Juanjo
> > > > > exten => 8991,1,Dial(SIP/8991,20)|t
> > > > > exten => 8991,2,Voicemail2(u8991 at demo)
> > > > > exten => 8991,102,Voicemail2(b8991 at demo)
> > > > > exten => 8991,103,Hangup
> > > > >
> > > > > [local]
> > > > > ;
> > > > > ; Master context for local, toll-free, and iaxtel calls only
> > > > > ;
> > > > > ignorepat => 9
> > > > > include => default
> > > > > include => parkedcalls
> > > > > include => trunklocal
> > > > > include => cell
> > > > > include => iaxtel700
> > > > > include => trunktollfree
> > > > > include => iaxprovider
> > > > >
> > > > > ------ parking.conf -----------
> > > > >
> > > > > [general]
> > > > > parkext => 700              ; What ext. to dial to park
> > > > > parkpos => 701-720          ; What extensions to park calls on
> > > > > context => parkedcalls          ; Which context parked calls
are
> > in
> > > > >
> > > > > ----- sip.conf ----------------
> > > > > [8991]
> > > > > type=friend
> > > > > username=8991
> > > > > secret=secret
> > > > > nat=no                  ; This phone may be natted
> > > > > host=dynamic
> > > > > canreinvite=no                  ; Cisco poops on reinvite
> > sometimes
> > > > > qualify=500                     ; Qualify peer is no more than
> > 200ms
> > > > > context=local
> > > > > mailbox=8991 at demo
> > > > >
> > > > >
> > > > >
> > > > > 	If I dial 700 I got busy tone (440 Not Found) the same
happens
> > > > if I
> > > > > dial #700 which I had to configure in dialplan.xml of the
phone
> > > > > (rewriting 700 as #700).
> > > > >
> > > > > Any suggestions ?
> > > > >
> > > > > --
> > > > > Juanjo sin .sig
> > > > >
> > > > > _______________________________________________
> > > > > Asterisk-Users mailing list
> > > > > Asterisk-Users at lists.digium.com
> > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > --
> > > Juanjo sin .sig
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users