[Asterisk-Users] Call Transfer

[Andy Powell]

Foong

Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below

Channel: SIP/1000 at mysipcontext
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: mysipcontext2
Extension: 2000
Priority: 1

This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2..

All you need is a script to lookup in the database and generate the script file for you and it's done.

HTH

Andy


*********** REPLY SEPARATOR  ***********

On 30/07/2003 at 16:30 Chee Foong wrote:

>Hello Dan,
>
>Thanks for you reply.
>
>Base on you recomendation using the 'T' argument. I manage to do call
>transfer an it works really well.
>
>My problem comes when my boss comes out with a superb idea where the
>transfering process is automated without involving a human :(
>
>Say asterisk get 2 numbers (from database, text file, etc), one belongs
>party A and the other belongs to party B. Asterisk will calls both parties
>and do the tranfer automatically. In another words, asterisk is resposible
>to 'press' the '#' to do the transfer. I don't this can be achieve in the
>extension.conf not matter how you structure you dial plan.
>
>Perhaps, the only way is to write a apps and plug it into asterisk like all
>the asterisk modules such as Meetme.
>
>Any ideas?
>
>
>Foong
>
>----- Original Message -----
>From: "Dan" <dtoma at fx.ro>
>To: <asterisk-users at lists.digium.com>
>Sent: Wednesday, July 30, 2003 3:42 PM
>Subject: Re: [Asterisk-Users] Call Transfer
>
>
>> Hi,
>>
>> It works if you put the 'T' switch in the dial line.
>>
>> You can then transfer the call from the caller.
>> I have tested it in the folllowing configuration and it works:
>> Call from a Cisco 7960 to an ATA 186.
>> Select 'Transfer" on 7960
>> Call another extension (X-Lite)
>> Select again transfer on 7960.
>> The call remain between ATA and X-Lite.
>>
>> This is what you need?
>>
>> BR,
>> Dan
>>
>> ----- Original Message -----
>> From: "Chee Foong" <cheefoong at inovas.com>
>> To: <asterisk-users at lists.digium.com>
>> Sent: Wednesday, July 30, 2003 7:08 AM
>> Subject: [Asterisk-Users] Call Transfer
>>
>>
>> Hello all,
>>
>> I am in a situation where I need to use asterisk to call someone say
>Party
>> A. After the call to Party A got through, asterisk will put Party A on
>hold,
>> then asterisk will call Party B. If call to Party B got through, asterisk
>> will transfer Party A to Party B.
>>
>> I wonder if this features is implemented into asterisk. I have found a
>post
>> in asterisk mailing list:
>> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
>>
>> but that doesn't help much.
>>
>> If this features is not implemented, can anyone give me some point on how
>to
>> implement this in asterisk? Do I need to write an app like the Dial apps
>for
>> asterisk to load at start up?
>>
>>
>> thanks
>>
>> Foong
>>
>>
>> _______________________________________________
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>> Asterisk-Users at lists.digium.com
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>>
>
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