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[asterisk-users] Disable transfer on all calls

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bee.beeep at gmail.com
Guest

Posted: Mon Apr 21, 2008 4:30 pm Post subject: [asterisk-users] Disable transfer on all calls

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 softphones (AFAIK this function uses SIP method REFER), so this transfers are hard to trace in CDR and my users can make a free call using trick with transfer:) I've googled it, but didn't find anything about my problem Thanks, Danila

greymanvoip at gmail.com
Guest

Posted: Tue Apr 22, 2008 5:54 am Post subject: [asterisk-users] Disable transfer on all calls

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.

dinesh at alphaque.com
Guest

Posted: Tue Apr 22, 2008 6:08 am Post subject: [asterisk-users] Disable transfer on all calls

On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:

Quote:

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."
dinesh at alphaque.com (0 0) http://www.openmalaysiablog.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do |
| for b in clients employers associates relatives neighbours pets; do |
| echo "The opinions here in no way reflect the opinions of my $a $b." |
| done; done |
+=========================================================================+

bee.beeep at gmail.com
Guest

Posted: Thu Apr 24, 2008 10:52 am Post subject: [asterisk-users] Disable transfer on all calls

Dinesh Nair ?????:

Quote:

On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:

Quote:

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 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.

It's really strange that this is in fact undocumented function - you can
find it only in comments on wiki at voip-info.org. Man there said that
he found this variable while hacking source code of asterisk:

$ grep -R TRANSFER_CONTEXT /usr/src/asterisk-1.2.15/
/usr/src/asterisk-1.2.15/channels/chan_sip.c: *transfercontext =
pbx_builtin_getvar_helper(sip_pvt->owner, "TRANSFER_CONTEXT");
/usr/src/asterisk-1.2.15/doc/README.variables:${TRANSFER_CONTEXT}
Context for transferred calls
/usr/src/asterisk-1.2.15/ChangeLog: * channels/chan_sip.c: chan_sip did
not use the TRANSFER_CONTEXT
/usr/src/asterisk-1.2.15/res/res_features.c: if
(!(transferer_real_context = pbx_builtin_getvar_helper(transferee,
"TRANSFER_CONTEXT")) &&
/usr/src/asterisk-1.2.15/res/res_features.c: !(transferer_real_context =
pbx_builtin_getvar_helper(transferer, "TRANSFER_CONTEXT"))) {
/usr/src/asterisk-1.2.15/res/res_features.c: if
(!(transferer_real_context=pbx_builtin_getvar_helper(transferee,
"TRANSFER_CONTEXT")) &&
/usr/src/asterisk-1.2.15/res/res_features.c:
!(transferer_real_context=pbx_builtin_getvar_helper(transferer,
"TRANSFER_CONTEXT"))) {

eric at fnords.org
Guest

Posted: Thu Apr 24, 2008 3:43 pm Post subject: [asterisk-users] Disable transfer on all calls

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.

bee.beeep at gmail.com wrote:

Quote:

Dinesh Nair ?????:

Quote:

On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:

Quote:

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.

--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.

greymanvoip at gmail.com
Guest

Posted: Fri Apr 25, 2008 2:42 am Post subject: [asterisk-users] Disable transfer on all calls

Quote:

It's only relevant for blind transfers. For attended transfers that
mechanism won't work.

Regards,

Greyman.

rentorbuy at yahoo.com
Guest

Posted: Mon Apr 28, 2008 2:10 am Post subject: [asterisk-users] Disable transfer on all calls

--- bee-beeep <bee.beeep at gmail.com> wrote:

Quote:

It works fine in every case, with disabling transfer
in Dial() options

2008/4/25 Grey Man <greymanvoip at gmail.com>:

Quote:

Thanks to your answers, but i found more

beautiful way to do this -

Quote:

there is some system variable

__TRANSFER_CONTEXT, which defines

Quote:

context

Quote:

to handle the transfered number, so you can

create a new context and

Quote:

there you can do anything with transfered

call - i just hang it up.

Quote:

It's only relevant for blind transfers. For

attended transfers that

Quote:

mechanism won't work.

In my case I don't want to disable transfer in Dial()
because I want transfers (both blind and attended) to
work always except in just one case: when "src"
extension is not one of my "local" extensions and when
"dst" is an outbound trunk. Typical scenario is to
avoid external callers to call a "local" extension
which in turn transfers the call to another external
number.

Currently, what I do is simply check the BLINDTRANSFER
variable on outbound trunk contexts and that works
fine except for attended transfers, of course.

I can't disable T in Dial() for outbound calls because
I want my local extensions to be able to transfer an
external call to another local extension.

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