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: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P. > > Sent: Tuesday, October 07, 2003 6:46 PM > > To: [EMAIL PROTECTED] > > 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([EMAIL PROTECTED]) > > > > exten => 8991,102,Voicemail2([EMAIL PROTECTED]) > > > > 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 > > > > [EMAIL PROTECTED] > > > > > > > > > > > > > > > > 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 > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > Juanjo sin .sig > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users