[Asterisk-Users] Dummy account/extension - Workaround for attended call trabsfer to ATA186

[Dan]

Hi again,

I think I have now a workaround for call transfer on ATA 186.

This is the extension corresponding to the phone connected to an ATA186

exten => 103,1,Dial(SIP/103,20),Tt
exten => 103,2,Voicemail2(us101)
exten => 103,3,Hangup
exten => 103,102,Ringing
exten => 103,103,Wait(1)
exten => 103,104,Goto(1)

I can now to attended transfer a call to this phone too.

The strange thins is that if I call this extension when the phone in
off-hook but not in  a call, it rings for 1 second then exit with a busy
tone.
Why?

Thanks,
Dan

----- Original Message ----- 
From: "Dan" <dtoma at fx.ro>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, July 30, 2003 5:44 PM
Subject: Re: [Asterisk-Users] Dummy account/extension


> Hi,
>
> Thanks for the suggestion.
> I have change it like that:
>
> ;dummy extension
> exten => 199,1,Ringing
> exten => 199,2,Wait(60) ; give illusion we might pick up
> exten => 199,3,Hangup
>
> in order to hear the ring too.
>
> ..but now... how can I do to call this extension from a Dial command?
>
> What I want in the final is to have a workaround for ATA186 in order to
> prevent consider it busy during the attended transfer.
> More, I want to prevent been bussy when not in a call. The Call Waiting
does
> not function during the dialtone period, just during the call.
>
> There is any other way to do it?
>
> Thanks for your help,
> Dan
>
>
>
>
> ----- Original Message ----- 
> From: "Armand A. Verstappen" <armand at nl.envida.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Wednesday, July 30, 2003 5:06 PM
> Subject: Re: [Asterisk-Users] Dummy account/extension
>
>
>
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>