[asterisk-bugs] [Asterisk 0016759]: Attended transfers get incorrect voicemail.
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Feb 3 21:36:41 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16759
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Reported By: Herb
Assigned To:
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Project: Asterisk
Issue ID: 16759
Category: Resources/res_features
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.6.1.14
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: 2010-02-02 22:00 CST
Last Modified: 2010-02-03 21:36 CST
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Summary: Attended transfers get incorrect voicemail.
Description:
Callers who get automated transferred but fail to get answered end up
getting the voicemail of the transferer and not the callee.
Person A calls Person B
Person B xfers Person A to Person C
Person C does not answer.
Person A gets voicemail of Person B.
This is version 1.6.1.13. Fresh install/compile. Using my current config
files.
Blind transfers work correct. If you blind transfer the person will get
the correct voicemail.
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Relationships ID Summary
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related to 0015347 Unanswered attended transfers get the v...
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(0117688) Herb (reporter) - 2010-02-03 21:36
https://issues.asterisk.org/view.php?id=16759#c117688
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Ok,
I downloaded the most recent branch via SVN and still getting the same
transfer/voicemail problem.
svn checkout http://svn.digium.com/svn/asterisk/branches/1.6.1
asterisk-1.6.1
Checked out external at revision 234.
Checked out revision 696.
Checked out revision 244686.
#asterisk -r
Connected to Asterisk SVN-branch-1.6.1-r244553
trace:
Connected to Asterisk SVN-branch-1.6.1-r244553 currently running on
kelepona (pid = 29157)
kelepona*CLI> core set verbose 10
Verbosity was 0 and is now 10
-- Channel 0/15, span 1 got hangup request, cause 16
== Spawn extension (closed, s, 1) exited non-zero on 'DAHDI/15-1'
-- Hungup 'DAHDI/15-1'
-- Accepting overlap call from 'XXXXXXXXXX' to 'XXXX143' on channel
0/16, span 1
-- Starting simple switch on 'DAHDI/16-1'
-- Executing [XXXX143 at default:1] Macro("DAHDI/16-1", "vm,143") in new
stack
-- Executing [s at macro-vm:1] Dial("DAHDI/16-1", "SIP/143,15,tTkK") in
new stack
== Using SIP RTP CoS mark 5
-- Called 143
-- SIP/143-00000003 is ringing
-- SIP/143-00000003 answered DAHDI/16-1
-- Started music on hold, class 'default', on DAHDI/16-1
-- Stopped music on hold on DAHDI/16-1
== Using SIP RTP CoS mark 5
-- Executing [178 at default:1] Macro("SIP/143-00000004", "vm,178") in
new stack
-- Executing [s at macro-vm:1] Dial("SIP/143-00000004",
"SIP/178,15,tTkK") in new stack
== Using SIP RTP CoS mark 5
-- Called 178
-- SIP/178-00000005 is ringing
== Spawn extension (macro-vm, s, 1) exited non-zero on
'SIP/143-00000004<ZOMBIE>' in macro 'vm'
== Spawn extension (default, XXXX143, 1) exited non-zero on
'SIP/143-00000004<ZOMBIE>'
-- Nobody picked up in 15000 ms
-- Executing [s at macro-vm:2] Goto("DAHDI/16-1", "s-NOANSWER,1") in new
stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER at macro-vm:1] VoiceMail("DAHDI/16-1", "143,su")
in new stack
-- <DAHDI/16-1> Playing
'/var/spool/asterisk/voicemail/default/143/unavail.slin' (language 'en')
-- Channel 0/16, span 1 got hangup request, cause 16
== Spawn extension (macro-vm, s-NOANSWER, 1) exited non-zero on
'DAHDI/16-1' in macro 'vm'
== Spawn extension (default, 178, 1) exited non-zero on 'DAHDI/16-1'
-- Hungup 'DAHDI/16-1'
I am going through my config files and trying to update syntax/code where
I can. Maybe it's my old macros?
Thanks again for the help.
Issue History
Date Modified Username Field Change
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2010-02-03 21:36 Herb Note Added: 0117688
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