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" > <asterisk-users@lists.digium.com> > 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 >
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