[Asterisk-Users] Re: Can I enter an extension to dial whilevoicemail is playing?

Thu Jun 22 13:59:25 MST 2006

Ahhh!  That fixed it!!!

However, it seems like I need to keep the Answer() in there.  This
causes incoming callers to here a stuttered ringing in the beginning.
Is there a way to fix/remove this?  I'm guessing there isn't because
Asterisk needs to answer to monitor the line for number presses...

However, I'm concerned this stuttered ringing will cause people to
call in to think we have a problem with our phones or something...

On 6/22/06, Tim Sharp <TSharp at infoviewsystems.com> wrote:
> 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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]On 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 <peter.antonacci at gmail.com> 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 <jklimek at gmail.com> wrote:
> > > Anybody have any more information on this Dial() "d" option for incoming
> > calls?
> > >
> > > On 6/19/06, John Klimek < jklimek at gmail.com> 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( u50 at default)
> > > > 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 <lnewmark at capalon.com> 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
> > > > >
> > > > >
> > > > > asterisk-users-request at lists.digium.com wrote:
> > > > >
> > > > > Message: 9
> > > > > Date: Mon, 19 Jun 2006 14:18:22 -0400
> > > > > From: "John Klimek" <jklimek at gmail.com>
> > > > > Subject: [Asterisk-Users] Can I enter an extension to dial while
> > > > >         voicemail       is playing?
> > > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > > >         <asterisk-users at lists.digium.com >
> > > > > Message-ID:
> > > > >
> > <c68396460606191118p12d6e5fcj144b5079995e11c2 at mail.gmail.com>
> > > > > 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(u50 at default)
> > > > > 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  :)
> > > > >
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