[asterisk-bugs] [Asterisk 0019301]: Called channel stays in app_dial_gosub_virtual_context and can't transfer call properly.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19301 
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Reported By:                vmikhnevych
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19301
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.4 
JIRA:                        
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:             2011-05-16 06:38 CDT
Last Modified:              2011-05-17 07:28 CDT
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Summary:                    Called channel stays in
app_dial_gosub_virtual_context and can't transfer call properly.
Description: 
Upon calling Dial() with U() parameter, called channel seems to stay in
app_dial_gosub_virtual_context , because if called person tries to make a
transfer (using codes from features.conf , like
https://issues.asterisk.org/view.php?id=1), and is asked to enter
number to transfer call to, only one digit is read, as Asterisk immediately
looks for the extension to transfer to in app_dial_gosub_virtual_context.
If there is no U() used in Dial(), transfer works great.
So, it's impossible for called party to perform attended or blind transfer
in this way, because transfer number can't be entered.

Dialplan for called subroutine:
[SET_MOH_VOLUME]
exten => s,1,GotoIf($["${ARG1}" = ""]?not_moh)
same => n,Set(CHANNEL(musicclass)=${ARG1})
same => n(not_moh),Set(VOLUME(TX)=${ARG2})
same => n,Set(VOLUME(RX)=${ARG3})
same => n,Return

Asterisk log:
-- AGI Script Executing Application: (Dial) Options:
(SIP/2051,60,U(SET_MOH_VOLUME^^2^)tTkd)
-- Called 2051
-- SIP/2051-00000002 answered SIP/1997-00000001
    -- Executing [s at SET_MOH_VOLUME:1] GotoIf("SIP/2051-00000002",
"1?not_moh") in new stack
    -- Goto (SET_MOH_VOLUME,s,3)
    -- Executing [s at SET_MOH_VOLUME:3] Set("SIP/2051-00000002",
"VOLUME(TX)=2") in new stack
    -- Executing [s at SET_MOH_VOLUME:4] Set("SIP/2051-00000002",
"VOLUME(RX)=") in new stack
    -- Executing [s at SET_MOH_VOLUME:5] Return("SIP/2051-00000002", "") in
new stack
    -- Executing [s at app_dial_gosub_virtual_context:1]
NoOp("SIP/2051-00000002", "") in new stack
    -- Auto fallthrough, channel 'SIP/2051-00000002' status is 'UNKNOWN'
    -- Started music on hold, class 'default', on SIP/1997-00000001
    -- <SIP/2051-00000002> Playing 'pbx-transfer.gsm' (language 'en')
[May 16 14:20:49] WARNING[3079]: features.c:1626 builtin_atxfer: Extension
'2' does not exist in context 'app_dial_gosub_virtual_context'
    -- <SIP/2051-00000002> Playing 'pbx-invalid.gsm' (language 'en')
    -- Stopped music on hold on SIP/1997-00000001


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---------------------------------------------------------------------- 
 (0135026) lmadsen (administrator) - 2011-05-17 07:28
 https://issues.asterisk.org/view.php?id=19301#c135026 
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This will only be fixed on 1.8+.

~~~~~

Per the Asterisk maintenance timeline page at
http://www.asterisk.org/asterisk-versions maintenance (bug) support for the
1.4 and 1.6.x branches has ended. For continued maintenance support please
move to the 1.8 branch which is a long term support (LTS) branch.

For more information about branch support, please see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions 

Issue History 
Date Modified    Username       Field                    Change               
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2011-05-17 07:28 lmadsen        Note Added: 0135026                          
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