The options are not seperated by commas. exten => s,1,Dial(SIP/50,23,r,d) should be exten => s,1,Dial(SIP/50,23,rd)
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Klimek Sent: Thursday, June 22, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Can I enter an extension to dial whilevoicemail is playing? Any idea why it wouldn't work in my dial plan? On 6/22/06, Peter Antonacci <[EMAIL PROTECTED]> wrote: > d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for > the call to be answered and returns that value on the spot. This allows you > to dial a 1-digit exit extension while waiting for the call to be answered - > see also > > > On 6/22/06, John Klimek <[EMAIL PROTECTED]> wrote: > > Anybody have any more information on this Dial() "d" option for incoming > calls? > > > > On 6/19/06, John Klimek < [EMAIL PROTECTED]> wrote: > > > Thanks for the information... > > > > > > After doing some reading it looks like I can use the "d" option with > > > the Dial() command to be able to enter a 1-digit extension while the > > > other extension is ringing, but this doesn't seem to be working for me > > > either... > > > > > > Here is my new config: > > > > > > exten => s,1,Dial(SIP/50,23,r,d) > > > exten => s,2,VoiceMail( [EMAIL PROTECTED]) > > > exten => s,3,Playback(vm-goodbye) > > > exten => s,4,Hangup > > > > > > exten => 1,1,SayDigits(1) > > > exten => 2,1,SayDigits(2) > > > exten => 10,1,SayDigits(10) > > > > > > However, when my phone is ringing (eg. extension 50), I try entering > > > "1" or "2" (to be forwarded via the Dial "d" option), but it doesn't > > > do anything. > > > > > > What am I doing wrong? > > > > > > I like your solution above, but if I use that I'll need to wait 23 > > > seconds for Dial() to timeout before I can do anything. I'd like to > > > be immediately able to enter an extension (if possible, which maybe > > > it's not...) > > > > > > On 6/19/06, Leah Newmark <[EMAIL PROTECTED]> wrote: > > > > Using the Background command, you will be able to play the voicemail > > > > while still being allowed to enter digits. > > > > > > > > exten => s,1,Wait(2) > > > > exten => > 108,2,Background(voicemail/default/108/unavail) > > > > > > > > > > > > exten => s,1,Dial(SIP/50,23,r) > > > > exten => > s,2,Background(/voicemail/default/50/unavail) ;or whatever > the > > > > soundfile is called > > > > exten => s,3,Voicemail(s50) ;s will skip the greeting and just go to > the > > > > beep > > > > exten => s,4,Playback(vm-goodbye) > > > > exten => s,5,Hangup > > > > > > > > You can then put > > > > exten => 1, Dial(sip/me) > > > > exten => 2, Dial(sip/her) > > > > or whatever your dial statements look like. > > > > > > > > Leah Newmark > > > > Capalon VoIP > > > > > > > > > > > > [EMAIL PROTECTED] wrote: > > > > > > > > Message: 9 > > > > Date: Mon, 19 Jun 2006 14:18:22 -0400 > > > > From: "John Klimek" <[EMAIL PROTECTED]> > > > > Subject: [Asterisk-Users] Can I enter an extension to dial while > > > > voicemail is playing? > > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > <[email protected] > > > > > Message-ID: > > > > > <[EMAIL PROTECTED]> > > > > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > > > > > > I have a very, very simple Asterisk setup in my house. I have a > > > > Sipura 3000 with a PSTN line connected and one analog phone connected. > > > > > > > > The [incoming] context looks like this: > > > > > > > > exten => s,1,Dial(SIP/50,23,r) > > > > exten => s,2,VoiceMail([EMAIL PROTECTED]) > > > > exten => s,3,Playback(vm-goodbye) > > > > exten => s,4,Hangup > > > > > > > > As you can see, when somebody calls in if I don't answer in 23 seconds > > > > then they are forwarded to my voicemail. > > > > > > > > How can I make it so I can call an enter extensions either while the > > > > phone is ringing or while the voicemail message is playing? I want > > > > the system to be as seemless as possible so the wife is happy =) > > > > > > > > Right now it works great because my Sipura 3000 forwards to call to > > > > Asterisk and Asterisk rings my analog phone, but the incoming caller > > > > hears a steady dial-tone the whole time. I wouldn't want that to > > > > change. (so the caller isn't wondering what is going on) > > > > > > > > Any help is appriciated :) > > > > > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
