[asterisk-bugs] [Asterisk 0016759]: Attended transfers get incorrect voicemail.

[Asterisk Bug Tracker]

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 17:19 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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 (0117680) Herb (reporter) - 2010-02-03 17:19
 https://issues.asterisk.org/view.php?id=16759#c117680 
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Aloha,

We are using the attended transfer function on our phones.  Cisco
7940s/7960s with SIP firmware 8.11.  I have the attended transfer feature
code enabled but it does not work for us.

Both the blind transfer function on the phones, and the feature code works
fine.

I have attached a debug output and here is a trace:

    -- Accepting overlap call from 'XXXXXXXXXX' to 'XXXX143' on channel
0/1, span 1
    -- Starting simple switch on 'DAHDI/1-1'
    -- Executing [XXXX143 at default:1] Macro("DAHDI/1-1", "vm,143") in new
stack
    -- Executing [s at macro-vm:1] Dial("DAHDI/1-1", "SIP/143,15,tTkK") in
new stack
  == Using SIP RTP CoS mark 5
    -- Called 143
    -- SIP/143-0000012d is ringing
    -- SIP/143-0000012d answered DAHDI/1-1
    -- Started music on hold, class 'default', on DAHDI/1-1
  == Using SIP RTP CoS mark 5
    -- Executing [178 at default:1] Macro("SIP/143-0000012e", "vm,178") in
new stack
    -- Executing [s at macro-vm:1] Dial("SIP/143-0000012e",
"SIP/178,15,tTkK") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 178
    -- SIP/178-0000012f is ringing
    -- Stopped music on hold on DAHDI/1-1
  == Spawn extension (macro-vm, s, 1) exited non-zero on
'SIP/143-0000012e<ZOMBIE>' in macro 'vm'
  == Spawn extension (default, XXXX143, 1) exited non-zero on
'SIP/143-0000012e<ZOMBIE>'
    -- Nobody picked up in 15000 ms
    -- Executing [s at macro-vm:2] Goto("DAHDI/1-1", "s-NOANSWER,1") in new
stack
    -- Goto (macro-vm,s-NOANSWER,1)
    -- Executing [s-NOANSWER at macro-vm:1] VoiceMail("DAHDI/1-1", "143,su")
in new stack
    -- <DAHDI/1-1> Playing
'/var/spool/asterisk/voicemail/default/143/unavail.slin' (language 'en')
    -- Channel 0/1, span 1 got hangup request, cause 16
  == Spawn extension (macro-vm, s-NOANSWER, 1) exited non-zero on
'DAHDI/1-1' in macro 'vm'
  == Spawn extension (default, 178, 1) exited non-zero on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'

Here is our voicemail macro:
[macro-vm]
include => default
exten => s,1,Dial(SIP/${ARG1},15,tTkK)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,VoiceMail(${ARG1},su)
exten => s-NOANSWER,n,Goto(default,s,1)
exten => s-BUSY,1,VoiceMail(${ARG1},sb)
exten => s-BUSY,n,Goto(default,s,1)
exten => _s-.,1,Goto(s-NOANSWER,1) 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-03 17:19 Herb           Note Added: 0117680                          
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