[asterisk-bugs] [Asterisk 0017021]: On omitting the T flag from Dial() the caller can still make a blind transfer

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17021 
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Reported By:                kovzol
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17021
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.29.1 
JIRA:                       SWP-1090 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-03-13 11:18 CST
Last Modified:              2010-03-16 06:48 CDT
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Summary:                    On omitting the T flag from Dial() the caller can
still make a blind transfer
Description: 
I use allowtransfer=yes in sip.conf. I think if I omit the T flag from
Dial() in a dialplan extension, no blind transfer should be made by the
caller. But the caller can still make a blind transfer if he presses the
TRANSFER key.

A partial workaround is if I set allowtransfer=no in sip.conf, but this
will disable the blind transfer initiated by the caller side as well (which
is not what I would like to).
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 (0119434) davidw (reporter) - 2010-03-16 06:48
 https://issues.asterisk.org/view.php?id=17021#c119434 
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I think this is a feature request.

We would want the ability to do SIP transfers to be enable-able, even if
DTMF transfers were not enabled, as the latter force retention of the RTP
stream, and one reason for disabling them is to allow re-invites out.

Also "the caller presses TRANSFER (DTMF)", doesn't seem to make sense - if
you use the actual phone TRANSFER button, the digits will not be sent as
DTMF, even if the phone generates DTMF as confidence tones.  They will
actually be sent in a SIP URL in a REFER message.

Formally, REFER needs to be accepted by the original other party, so
adding a feature to reject REFER would not conflict with the SIP
specification.  Note that this can be a preconfigured consultation with the
user.

Note that, attended transfers will still get as far as the enquiry, as
this is indistinguishable from a normal outgoing call, and that the phone
may direct the REFER to either end, at its discretion. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-16 06:48 davidw         Note Added: 0119434                          
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