[Asterisk-Users] Call park on SIP phones

[Andrew Joakimsen]

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
> 
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