[asterisk-users] Asterisk Dial Plan to Play Message

[Steve Totaro]

which is exactly what I said if you read the whole thread  :-)

Eric "ManxPower" Wieling wrote:
> You can do it one of two ways:
>
> 1) make the SIP device dial a predefined number when the user picks up 
> the phone.  You do this in the SIP device.  Check the manual for that 
> device for detail on how to do this.  It's normally called "hotline". 
> In extensions.conf have Asterisk run Authenticate before the Dial() line.
>
> 2) Let the SIP device dial as normal, but in the dialplan execute 
> Authenticate before the Dial line.
>
> Steve Totaro wrote:
>> You could put the phone in a context such as context=restricted in 
>> sip.conf
>>
>> In extensions.conf put a context
>> [restricted]
>> exten => _.,1,Answer
>> exten => _.,2,Authenticate(8675301)
>> exten => _.,3,Goto(whateverdialcontext,whateverexten,whateverpriority)
>>
>> replace Allison's recording for authenticate with your own.
>> Unless I am totally missing what you are trying to do.
>>
>> Thanks,
>> Steve
>>
>> Eric "ManxPower" Wieling wrote:
>>> "[9507]" is the incoming User ID.  "user=8407" is the outgoing User 
>>> ID.  Do you really want them to be different?
>>>
>>> Dial() will stop processing of the dialplan until the call ends.  Do 
>>> you really want this?
>>>
>>> "r" option to Dial will force a ringing sound to the caller, even if 
>>> the caller should be hearing a "all circuits are busy", or "your 
>>> call cannot be completed as dialed" or similar message.  Do you 
>>> really want that?
>>>
>>> broadbandvoice at comcast.net wrote:
>>>> Thanks for the response, its looks logical, for some reason the 
>>>> authentication is not working for me, I'm using a Linksys Phone 
>>>> adapter and here is a sample dial plan in extensions.conf and also 
>>>> SIP channels.
>>>>
>>>> exten => 8407,1,Dial(SIP/8407,80,rt)      ; permit transfer
>>>> exten => 8407,n,Authenticate(9461)                      exten => 
>>>> 8407,n,Playback(pbx-invalid)
>>>> exten => 8407,n,Hangup()
>>>>
>>>> and in sip.conf
>>>>
>>>> [9507]
>>>> type=friend
>>>> user=8407
>>>> secret=xxxxxxxxxx
>>>> ;context=from-sip
>>>> callerid=8407
>>>> host=dynamic
>>>> nat=yes
>>>> qualify=yes
>>>> canreinvite=no
>>>> dtmfmode=rfc2833
>>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>