[Asterisk-Users] Call park on SIP phones

[Brian West]

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
>