<strong>d</strong>: 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
<br><br>
<div><span class="gmail_quote">On 6/22/06, <b class="gmail_sendername">John Klimek</b> <<a href="mailto:jklimek@gmail.com">jklimek@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Anybody have any more information on this Dial() "d" option for incoming calls?<br><br>On 6/19/06, John Klimek <
<a href="mailto:jklimek@gmail.com">jklimek@gmail.com</a>> wrote:<br>> Thanks for the information...<br>><br>> After doing some reading it looks like I can use the "d" option with<br>> the Dial() command to be able to enter a 1-digit extension while the
<br>> other extension is ringing, but this doesn't seem to be working for me<br>> either...<br>><br>> Here is my new config:<br>><br>> exten => s,1,Dial(SIP/50,23,r,d)<br>> exten => s,2,VoiceMail(
u50@default)<br>> exten => s,3,Playback(vm-goodbye)<br>> exten => s,4,Hangup<br>><br>> exten => 1,1,SayDigits(1)<br>> exten => 2,1,SayDigits(2)<br>> exten => 10,1,SayDigits(10)<br>><br>
> However, when my phone is ringing (eg. extension 50), I try entering<br>> "1" or "2" (to be forwarded via the Dial "d" option), but it doesn't<br>> do anything.<br>><br>> What am I doing wrong?
<br>><br>> I like your solution above, but if I use that I'll need to wait 23<br>> seconds for Dial() to timeout before I can do anything. I'd like to<br>> be immediately able to enter an extension (if possible, which maybe
<br>> it's not...)<br>><br>> On 6/19/06, Leah Newmark <<a href="mailto:lnewmark@capalon.com">lnewmark@capalon.com</a>> wrote:<br>> > Using the Background command, you will be able to play the voicemail
<br>> > while still being allowed to enter digits.<br>> ><br>> > exten => s,1,Wait(2)<br>> > exten => 108,2,Background(voicemail/default/108/unavail)<br>> ><br>> ><br>> > exten => s,1,Dial(SIP/50,23,r)
<br>> > exten => s,2,Background(/voicemail/default/50/unavail) ;or whatever the<br>> > soundfile is called<br>> > exten => s,3,Voicemail(s50) ;s will skip the greeting and just go to the<br>> > beep
<br>> > exten => s,4,Playback(vm-goodbye)<br>> > exten => s,5,Hangup<br>> ><br>> > You can then put<br>> > exten => 1, Dial(sip/me)<br>> > exten => 2, Dial(sip/her)<br>> > or whatever your dial statements look like.
<br>> ><br>> > Leah Newmark<br>> > Capalon VoIP<br>> ><br>> ><br>> > <a href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a> wrote:<br>> >
<br>> > Message: 9<br>> > Date: Mon, 19 Jun 2006 14:18:22 -0400<br>> > From: "John Klimek" <<a href="mailto:jklimek@gmail.com">jklimek@gmail.com</a>><br>> > Subject: [Asterisk-Users] Can I enter an extension to dial while
<br>> > voicemail is playing?<br>> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br>> > <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com
</a>><br>> > Message-ID:<br>> > <<a href="mailto:c68396460606191118p12d6e5fcj144b5079995e11c2@mail.gmail.com">c68396460606191118p12d6e5fcj144b5079995e11c2@mail.gmail.com</a>><br>> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed
<br>> ><br>> > I have a very, very simple Asterisk setup in my house. I have a<br>> > Sipura 3000 with a PSTN line connected and one analog phone connected.<br>> ><br>> > The [incoming] context looks like this:
<br>> ><br>> > exten => s,1,Dial(SIP/50,23,r)<br>> > exten => s,2,VoiceMail(u50@default)<br>> > exten => s,3,Playback(vm-goodbye)<br>> > exten => s,4,Hangup<br>> ><br>> > As you can see, when somebody calls in if I don't answer in 23 seconds
<br>> > then they are forwarded to my voicemail.<br>> ><br>> > How can I make it so I can call an enter extensions either while the<br>> > phone is ringing or while the voicemail message is playing? I want
<br>> > the system to be as seemless as possible so the wife is happy =)<br>> ><br>> > Right now it works great because my Sipura 3000 forwards to call to<br>> > Asterisk and Asterisk rings my analog phone, but the incoming caller
<br>> > hears a steady dial-tone the whole time. I wouldn't want that to<br>> > change. (so the caller isn't wondering what is going on)<br>> ><br>> > Any help is appriciated :)<br>> ><br>
> > _______________________________________________<br>> > --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>> ><br>> > Asterisk-Users mailing list<br>> > To UNSUBSCRIBE or update options visit:
<br>> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>> ><br>><br>_______________________________________________<br>--Bandwidth and Colocation provided by
<a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br></blockquote></div><br>