asterisk users - Oct 2004 - Slim Devices Sqeezebox Asterisk voicemail plugin.

Hi everybody,

For those of you that are interested I have written a plugin for the 
Slim Devices Squeeze box to monitor, read and play back voice mail 
messages created by an Asterisk PBX server.

You can find it at the address 
http://www.dinplug.com/vmplugin/dev_overview.html

Ian.

cool, just installed it

where can I choose which voicebox should be scanned for new messages,
I have more asterisk users than slimp3 assigned to them... ?




On Sat, 09 Oct 2004 16:03:14 +0200, Ian Hailey <asterisk@dinplug.com>
wrote:
> Hi everybody,
> 
> For those of you that are interested I have written a plugin for the
> Slim Devices Squeeze box to monitor, read and play back voice mail
> messages created by an Asterisk PBX server.
> 
> You can find it at the address
> http://www.dinplug.com/vmplugin/dev_overview.html
> 
> Ian.
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hi Ian,

Thank you! This is very cool. I did not patch voicemail.c because the  
slimserver and asterisk are already on the same box running with the  
same user id. I also did not patch the Source.pm file. Everything seems  
to work just fine.

[It would be nice to separate new and old (already played) voicemail  
messages.]

The slimserver is already used for Music on Hold at my home *  
installation. I wrote a little how-to on the wiki:
http://www.voip-info.org/tiki-index.php? 
page=Using+Slimserver+for+Music+on+Hold

Joris.


On 9-okt-04, at 16:03, Ian Hailey wrote:
> Hi everybody,
>
> For those of you that are interested I have written a plugin for the  
> Slim Devices Squeeze box to monitor, read and play back voice mail  
> messages created by an Asterisk PBX server.
>
> You can find it at the address  
> http://www.dinplug.com/vmplugin/dev_overview.html
>
> Ian.
>

On Sun, 2004-10-10 at 11:17 +0100, Bill Seddon wrote:
> So can anyone recommend a hardware sipphone that can
> be programmed to call an extension when it is lifted off-hook?
> 
> I can see that the Grandstream Budgetone supports "Early dial"
but it
> appears that the user still has to send at least one character.
> 
Look a bit closer and you will find:-

Off hook autodial:


-- 
Dave Cotton <dcotton@linuxautrement.com>

The GrandStream phone does support that feature. I dont think that Early
Dial is exactly that. Early Dial is some other feature. The BudgeTone 102
will work just fine.

Thank you,
Steve Maroney

On Sun, 10 Oct 2004, Bill Seddon wrote:
> Looking in the draft handbook I can see there is an example in the sample
> zapata.conf showing how to setup an fxs channel to behave as a lobby phone
> (the attached phone rings an operator as soon as the phone is lifted
> off-hook).
>
> There seems to be no equivalent SIP option presumably because a siphone has
> to initiate the call.  So can anyone recommend a hardware sipphone that can
> be programmed to call an extension when it is lifted off-hook?
>
> I can see that the Grandstream Budgetone supports "Early dial"
but it
> appears that the user still has to send at least one character.
>
> We could, of course, go and buy an fxs port for asterisk to solve the issue
> but since we use sip/iax for our telephony needs currently, that seems a
> retrograde step.
>
> Any insights welcome
>
> Bill Seddon
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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early dial is when asterisk checks after each digit is dialed and when a
match is found, connects the call.  this requires a good dialplan since for
expample is you have and extension 150 setup voicemail for exten 15 you will
always hit voicemail and never be able to get the 0 input.

i think GS calles it off hook autodial or something to that effect.


----- Original Message ----- 
From: "Steve Maroney" <steve@stevenet.net>
To: <bill.seddon@lyquidity.com>; "Asterisk Users Mailing List -
Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Sent: Sunday, October 10, 2004 12:00 PM
Subject: Re: [Asterisk-Users] SIP lobby phone

>
> The GrandStream phone does support that feature. I dont think that Early
> Dial is exactly that. Early Dial is some other feature. The BudgeTone 102
> will work just fine.
>
> Thank you,
> Steve Maroney
>
> On Sun, 10 Oct 2004, Bill Seddon wrote:
>
> > Looking in the draft handbook I can see there is an example in the
sample
> > zapata.conf showing how to setup an fxs channel to behave as a lobby
phone
> > (the attached phone rings an operator as soon as the phone is lifted
> > off-hook).
> >
> > There seems to be no equivalent SIP option presumably because a
siphone
has
> > to initiate the call.  So can anyone recommend a hardware sipphone
that
can
> > be programmed to call an extension when it is lifted off-hook?
> >
> > I can see that the Grandstream Budgetone supports "Early
dial" but it
> > appears that the user still has to send at least one character.
> >
> > We could, of course, go and buy an fxs port for asterisk to solve the
issue
> > but since we use sip/iax for our telephony needs currently, that seems
a
> > retrograde step.
> >
> > Any insights welcome
> >
> > Bill Seddon
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
>