Hi All,
I want to prevent transfer on based of user,
means we can disable any user or peer to transfer calls in asterisk.
Can any one helps how can we prevent transfer feature.
I am using asterisk 1.4 branch.
Thanks,
Max Alex
Voip Developer
___
-- Bandwi
--- bee-beeep <[EMAIL PROTECTED]> wrote:
> It works fine in every case, with disabling transfer
> in Dial() options
>
> 2008/4/25 Grey Man <[EMAIL PROTECTED]>:
>
> > > > > Thanks to your answers, but i found more
> beautiful way to do this -
> > > > > there is some system variable
> __TRANSFER_
It works fine in every case, with disabling transfer in Dial() options
2008/4/25 Grey Man <[EMAIL PROTECTED]>:
> > > > Thanks to your answers, but i found more beautiful way to do this -
> > > > there is some system variable __TRANSFER_CONTEXT, which defines
> context
> > > > to handle the transf
> > > Thanks to your answers, but i found more beautiful way to do this -
> > > there is some system variable __TRANSFER_CONTEXT, which defines context
> > > to handle the transfered number, so you can create a new context and
> > > there you can do anything with transfered call - i just hang it up
Most times it's easier to find something in google, than in your own
computer :)
2008/4/25, Eric Wieling <[EMAIL PROTECTED]>:
>
> In 1.2 it is documented in /path/to/src/asterisk/doc/README.variables,
> in 1.4 the file is called /path/to/src/asterisk/doc/channelvariables.txt
>
> The "doc" director
In 1.2 it is documented in /path/to/src/asterisk/doc/README.variables,
in 1.4 the file is called /path/to/src/asterisk/doc/channelvariables.txt
The "doc" directory is the only official source of documentation for
Asterisk that I am aware of. Read it.
[EMAIL PROTECTED] wrote:
> Dinesh Nair пише
Dinesh Nair пишет:
> On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
>
>> The best option is to put a SIP Proxy in front of your Asterisk sever
>> and block REFER requests.
>>
>
> or just comment out the block in chan_sip.c which handles the refers.
>
>
Thanks to your answers, but
On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
> The best option is to put a SIP Proxy in front of your Asterisk sever
> and block REFER requests.
or just comment out the block in chan_sip.c which handles the refers.
--
Regards, /\_/\ "All dogs go to heaven."
[E
Hi Danila,
You can't turn them transfers off with Asterisk.
The best option is to put a SIP Proxy in front of your Asterisk sever
and block REFER requests.
Regards,
Greyman.
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a
Hi folks,
I have some asterisk 1.2 box with self-made billing, and I need to
disable call transfer on all calls and directions.
I turned it off in features.conf and there is no 'tT' option in all my
Dial() commands, but users still able to transfer call using "transfer"
function in ip of softph
Hi!
> of the above scenario using 'Dial', and confirmed that it works). however,
> i can't seem to figure out how to disable transfer for outgoing calls that
> are initiated through the Manager interface (or through .call files, for that
> matter).
Interesting problem.
Try to use a "Local" chan
hi there. i'm developing a kind of collaborative phone system in which
groups of users need to navigate phone menus together, and run into
problems with the '#' character.
here's a scenario:
1) person_a dials into my asterisk server.
2) asterisk parks the call, and initiates an outgoing call to pe
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