Hi, It is possible to create a dummy account (SIP or IAX type) in order to be used in a "dummy" extension? I want to be able to use it as a normal extension (as an IP phone connected to it), but without the need to answer or call from that extension. I want that when I call that extension to hear the ring, and after the defined period of time to enter in the Voicemail system. I don't want to use a real phone (hardware or software) for this purpose. It is possible to do this in a simple way? Thanks, Dan
On Wed, 2003-07-30 at 15:55, Dan wrote:> It is possible to create a dummy account (SIP or IAX type) in order to be > used in a "dummy" extension? > I want to be able to use it as a normal extension (as an IP phone connected > to it), but without the need to answer or call from that extension. > I want that when I call that extension to hear the ring, and after the > defined period of time to enter in the Voicemail system. > I don't want to use a real phone (hardware or software) for this purpose. > > It is possible to do this in a simple way?doesn't: [globals] WAITTIME=10 MAILBOX=1234 [dummy] exten => 1234,1,Wait(${WAITTIME}) ; give illusion we might pick up exten => 1234,2,VoiceMail2(${MAILBOX}) ; then kick into voicemail exten => 1234,3,Hangup do the trick? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 armand@nl.envida.net 3531 AH Utrecht tel: +31 (0)30 298 2255 Postbus 19127 fax: +31 (0)30 298 2111 3501 DC Utrecht -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030730/24465f9c/attachment.pgp
Hi, Thanks for the suggestion. I have change it like that: ;dummy extension exten => 199,1,Ringing exten => 199,2,Wait(60) ; give illusion we might pick up exten => 199,3,Hangup in order to hear the ring too. ..but now... how can I do to call this extension from a Dial command? What I want in the final is to have a workaround for ATA186 in order to prevent consider it busy during the attended transfer. More, I want to prevent been bussy when not in a call. The Call Waiting does not function during the dialtone period, just during the call. There is any other way to do it? Thanks for your help, Dan ----- Original Message ----- From: "Armand A. Verstappen" <armand@nl.envida.net> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 5:06 PM Subject: Re: [Asterisk-Users] Dummy account/extension
Dan
2003-Jul-30 07:58 UTC
[Asterisk-Users] Dummy account/extension - Workaround for attended call trabsfer to ATA186
Hi again, I think I have now a workaround for call transfer on ATA 186. This is the extension corresponding to the phone connected to an ATA186 exten => 103,1,Dial(SIP/103,20),Tt exten => 103,2,Voicemail2(us101) exten => 103,3,Hangup exten => 103,102,Ringing exten => 103,103,Wait(1) exten => 103,104,Goto(1) I can now to attended transfer a call to this phone too. The strange thins is that if I call this extension when the phone in off-hook but not in a call, it rings for 1 second then exit with a busy tone. Why? Thanks, Dan ----- Original Message ----- From: "Dan" <dtoma@fx.ro> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 5:44 PM Subject: Re: [Asterisk-Users] Dummy account/extension> Hi, > > Thanks for the suggestion. > I have change it like that: > > ;dummy extension > exten => 199,1,Ringing > exten => 199,2,Wait(60) ; give illusion we might pick up > exten => 199,3,Hangup > > in order to hear the ring too. > > ..but now... how can I do to call this extension from a Dial command? > > What I want in the final is to have a workaround for ATA186 in order to > prevent consider it busy during the attended transfer. > More, I want to prevent been bussy when not in a call. The Call Waitingdoes> not function during the dialtone period, just during the call. > > There is any other way to do it? > > Thanks for your help, > Dan > > > > > ----- Original Message ----- > From: "Armand A. Verstappen" <armand@nl.envida.net> > To: <asterisk-users@lists.digium.com> > Sent: Wednesday, July 30, 2003 5:06 PM > Subject: Re: [Asterisk-Users] Dummy account/extension > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > >
On Wed, 2003-07-30 at 16:44, Dan wrote:> Thanks for the suggestion. > I have change it like that: > > ;dummy extension > exten => 199,1,Ringing > exten => 199,2,Wait(60) ; give illusion we might pick up > exten => 199,3,Hangup > > in order to hear the ring too.> > ..but now... how can I do to call this extension from a Dial command?Not sure what you are trying to do, but would the goto app be of any help? [other-ext] ... exten => 198,3,Goto(dummy,199,1) wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 armand@nl.envida.net 3531 AH Utrecht tel: +31 (0)30 298 2255 Postbus 19127 fax: +31 (0)30 298 2111 3501 DC Utrecht -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030730/5acfc5ba/attachment.pgp
Hi, It was one of the possible way to get a workaround with my ATA and attended call transfer. It was solved for now.. check one of my previous mails. Thanks, Dan ----- Original Message ----- From: "Armand A. Verstappen" <armand@nl.envida.net> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 6:11 PM Subject: Re: [Asterisk-Users] Dummy account/extension