[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:                lmadsen
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Project:                    Asterisk
Issue ID:                   17021
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
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:              2010-03-19 02:53 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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---------------------------------------------------------------------- 
 (0119599) kovzol (reporter) - 2010-03-19 02:53
 https://issues.asterisk.org/view.php?id=17021#c119599 
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Here is my sip.conf (which causes problems):

[general]
allowtransfer=yes
context=default
allowguest=yes
bindport=5060
srvlookup=no
domain=this.is.my.service.com
realm=this.is.my.service.com
autodomain=yes
pedantic=yes
t38pt_udptl=no
disallow=all
allow=alaw
allow=ulaw
allow=gsm
language=en
useragent=MyUserAgentString
dtmfmode=auto
amaflags=billing
qualify=yes
canreinvite=no
defaultexpiry=600
maxexpiry=3660
rtptimeout=180
rtpholdtimeout=600
tos_sip=cs3
tos_audio=ef
externip=my.external.ip

[12345678] ;sample client config
type=friend
secret=hissecret
username=12345678
host=dynamic
disallow=all
allow=alaw
allow=ulaw
allow=gsm
call-limit=1
nat=yes
callerid=12345678

I guess promiscredir is set to "no" by default, so this is what my system
assumes. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-19 02:53 kovzol         Note Added: 0119599                          
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