[asterisk-bugs] [Asterisk 0015041]: Crash on attended transfer

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=15041 
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Reported By:                maxnuv
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15041
Category:                   General
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.24 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-05-06 04:47 CDT
Last Modified:              2009-05-07 23:37 CDT
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Summary:                    Crash on attended transfer
Description: 
Call incoming, first transfer (all ok), second transfer: call, the called
respond, the trasferer hangup, SEGFAULT (before playng beep).

Two backtrace available, one old and one new.

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---------------------------------------------------------------------- 
 (0104428) maxnuv (reporter) - 2009-05-07 23:37
 http://bugs.digium.com/view.php?id=15041#c104428 
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Hum, maybe. First, the "caller" was me :-) so i hear on the phone and see
on the console wath appens at the segfault.

The call was incoming on line and was answered from a SIP phone with a
queue, and transferred to a sip phone.

The call progress (number was 0121somethingnumber:

[from-telecom]

exten => (number),1,Noop(Numero ${EXTEN} entrata)
exten => (number),n,Set(CDR(accountcode)=entrata-gidi)
exten => (number),n,Set(CDR(userfield)=number)
exten => (number),n,Goto(e-standard|1

exten => e-standard,1,Noop(entrata standard)
exten => e-standard,n,Ringing()
exten => e-standard,n,Wait(3)
exten => e-standard,n,Queue(entrata|rtw)

so queue entrata is (queue.conf)

[entrata]
wrapuptime=1
retry = 5
strategy = ringall
member => SIP/211
member => SIP/218
member => SIP/210
member => SIP/217

So 210 answered.

210 pressed # (so this is why "maybe" this is attended transfer from the
asterisk side, not the SIP PHONE feature i think).
dialed 223, the SIP 223 answered, a little talk with 210, when 210 hangup
here the segfault, so you see the call between SIP 223 and SIP 210....

the extension on from-sip-u

[from-sip-u]
include=>from-sip-base
include=>from-sip-u-base

[from-sip-u-base]

... snip ...

exten => _2XX,1,Noop(Interno ${EXTEN})
exten => _2XX,n,Set(__CHIAMANTE=${CALLERID(num)})
exten => _2XX,n,Set(CDR(accountcode)=da ${CHIAMANTE})
exten => _2XX,n,Set(__INTERNO=${EXTEN})
exten => _2XX,n,Set(CDR(userfield)=${INTERNO})
exten => _2XX,n,Dial(SIP/${EXTEN}||rtTw)
exten => _2XX,n,Goto(interno-${DIALSTATUS}|1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

if you think it is necessary i can put here all the extensions.conf. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-07 23:37 maxnuv         Note Added: 0104428                          
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