-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Danny,
Thanks for your replay. Jep, that would be a possibility. But then the user has to wait until my dialtime is over. If he/she is that inpatient, then with my solution he/she can end the dialing whenever needed. But, I'll try your successtion, looks interesting. chris... Danny Nicholas schrieb: > This is what you "Really" want; It should work with SIP or Zap > > exten => _X.,1,Dial(${DIALNUM},${ARG2},tT) > exten => _X.-NOANSWER,1,background(press5tocallback) > exten => -X.-NOANSWER,2,waitexten(5) > exten => 5,1,goto(callback,s,1) > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Atis Lezdins > Sent: Wednesday, April 15, 2009 9:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Exit Dial Application > > On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller > <fuch_li...@kurtkrenn.com> wrote: >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Hi Danny, >> >> Danny Nicholas schrieb: >>> Here's how core show application dial says you should do it: >>> Change your dial to >>> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback) >> I'm not sure if this is correct. core show application dial says: >> Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]) >> If I configure what you wrote, then callback is passed as URL to the > called party. >> "The optional URL will be sent to the called party if the channel supports > it." >> I don't think that's what I want. >> What I want is: If A dials B and B doesn't answer, A can press 5 and place > an automatic >> callback. If B is back and places or takes a call, the automatic callback > to A should be >> started. >> >> I've found a possibility to do this via answering the call before the > dial. But ... that's >> not an ideal solution. I would prefer not to answer the call in the > dialplan. Does the >> option 'd' implies an answered channel? Or is this a Bug? >> > > I think the limitation could be by analogous Zap phones, as they > probably don't support sending DTMF on unanswered channel. You could > try it opposite way - Dial from SIP phone to Zap. > > Regards, > Atis > - -- commpany dialog solutions gmbh Dipl.-Ing.(FH) Christoph Fürstaller IP-Communications Ischlerbahnstraße 14, 5301 Eugendorf Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 commpany (26667269) Email: c.fuerstal...@commpany.at -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAknmDIAACgkQR0exH8dhr/ZySQCfSAJ+ir0memNLKF5q0M219XPP f3AAn0PYw580wN2xWZOUgdSJNIPq/ZBd =5TkD -----END PGP SIGNATURE----- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users