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

[John Klimek]

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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