[asterisk-bugs] [Asterisk 0014215]: Asterisk crashes anytime a call is parked by any method.

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Jan 16 16:16:22 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14215 
====================================================================== 
Reported By:                waverly360
Assigned To:                otherwiseguy
====================================================================== 
Project:                    Asterisk
Issue ID:                   14215
Category:                   Applications/app_parkandannounce
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.0.3-rc1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-01-12 09:35 CST
Last Modified:              2009-01-16 16:16 CST
====================================================================== 
Summary:                    Asterisk crashes anytime a call is parked by any
method.
Description: 
I've attempted to park a call using a Polycom IP 330 (firmware version
3.0.1.0032) by using the softkey attended and blind transfer functions.  

My previous version of asterisk was 1.2.14 and I'm attempting the upgrade
to 1.6.  As such, some of my config files may be out of date.  I've already
went through several and updated them, so I'm aware the issue could be
configuration related...I just didn't think a misconfiguration should cause
a crash like this, so I'm posting this bug.


The attended transfer will announce the parked extension, and then
asterisk crashes.  Here's the CLI output.  I have the core file as
well...not sure how to attach it to this bug though.

    -- Accepting call from '5555551212' to '5554441212' on channel 0/1,
span 1
[Jan 12 08:51:07] WARNING[16135]: chan_dahdi.c:1673 dahdi_enable_ec:
Unable to enable echo cancellation on channel 1 (No such device)
    -- Executing [5554441212 at inbound:1] Set("DAHDI/1-1",
"NumberCalled=6155158725") in new stack
    -- Executing [5554441212 at inbound:2] Wait("DAHDI/1-1", "2") in new
stack
    -- Executing [5554441212 at inbound:3] AGI("DAHDI/1-1",
"agi://127.0.0.1") in new stack
AGI Tx >> agi_network: yes
<DAHDI/1-1>AGI Tx >> agi_request: agi://127.0.0.1
<DAHDI/1-1>AGI Tx >> agi_channel: DAHDI/1-1
<DAHDI/1-1>AGI Tx >> agi_language: en
<DAHDI/1-1>AGI Tx >> agi_type: DAHDI
<DAHDI/1-1>AGI Tx >> agi_uniqueid: 1231771867.2
<DAHDI/1-1>AGI Tx >> agi_version: 1.6.0.3
<DAHDI/1-1>AGI Tx >> agi_callerid: 5555551212
<DAHDI/1-1>AGI Tx >> agi_calleridname: BUSINESS
<DAHDI/1-1>AGI Tx >> agi_callingpres: 0
<DAHDI/1-1>AGI Tx >> agi_callingani2: 0
<DAHDI/1-1>AGI Tx >> agi_callington: 33
<DAHDI/1-1>AGI Tx >> agi_callingtns: 0
<DAHDI/1-1>AGI Tx >> agi_dnid: 5554441212
<DAHDI/1-1>AGI Tx >> agi_rdnis: unknown
<DAHDI/1-1>AGI Tx >> agi_context: inbound
<DAHDI/1-1>AGI Tx >> agi_extension: 5554441212
<DAHDI/1-1>AGI Tx >> agi_priority: 3
<DAHDI/1-1>AGI Tx >> agi_enhanced: 0.0
<DAHDI/1-1>AGI Tx >> agi_accountcode: 
<DAHDI/1-1>AGI Tx >> agi_threadid: -1217377392
<DAHDI/1-1>AGI Tx >> 
<DAHDI/1-1>AGI Rx << ANSWER ""
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Set "NumberCalled="
    -- AGI Script Executing Application: (Set) Options: (NumberCalled=)
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Ringing ""
    -- AGI Script Executing Application: (Ringing) Options: ()
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Wait "3"
    -- AGI Script Executing Application: (Wait) Options: (3)
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Dial "SIP/1,15,rtT"
    -- AGI Script Executing Application: (Dial) Options: (SIP/1,15,rtT)
  == Using SIP RTP CoS mark 5
    -- Called 1
    -- SIP/1-08593298 is ringing
    -- SIP/1-08593298 answered DAHDI/1-1
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
  == Using SIP RTP CoS mark 5
    -- Executing [700 at phones:1] AGI("SIP/1-08598b38", "agi://127.0.0.1")
in new stack
AGI Tx >> agi_network: yes
<SIP/1-08598b38>AGI Tx >> agi_request: agi://127.0.0.1
<SIP/1-08598b38>AGI Tx >> agi_channel: SIP/1-08598b38
<SIP/1-08598b38>AGI Tx >> agi_language: en
<SIP/1-08598b38>AGI Tx >> agi_type: SIP
<SIP/1-08598b38>AGI Tx >> agi_uniqueid: 1231771883.4
<SIP/1-08598b38>AGI Tx >> agi_version: 1.6.0.3
<SIP/1-08598b38>AGI Tx >> agi_callerid: 125
<SIP/1-08598b38>AGI Tx >> agi_calleridname: Waverly
<SIP/1-08598b38>AGI Tx >> agi_callingpres: 0
<SIP/1-08598b38>AGI Tx >> agi_callingani2: 0
<SIP/1-08598b38>AGI Tx >> agi_callington: 0
<SIP/1-08598b38>AGI Tx >> agi_callingtns: 0
<SIP/1-08598b38>AGI Tx >> agi_dnid: 700
<SIP/1-08598b38>AGI Tx >> agi_rdnis: unknown
<SIP/1-08598b38>AGI Tx >> agi_context: phones
<SIP/1-08598b38>AGI Tx >> agi_extension: 700
<SIP/1-08598b38>AGI Tx >> agi_priority: 1
<SIP/1-08598b38>AGI Tx >> agi_enhanced: 0.0
<SIP/1-08598b38>AGI Tx >> agi_accountcode: 
<SIP/1-08598b38>AGI Tx >> agi_threadid: -1217873008
<SIP/1-08598b38>AGI Tx >> 
<SIP/1-08598b38>AGI Rx << NOOP "not inbound, skip to call routing"
<SIP/1-08598b38>AGI Tx >> 200 result=0
<SIP/1-08598b38>AGI Rx << EXEC Park ""
    -- AGI Script Executing Application: (Park) Options: ()
  == Parked SIP/1-08598b38 on 701 at parkedcalls. Will timeout back to
extension [phones] s, 1 in 45 seconds
    -- <SIP/1-08598b38> Playing 'digits/7.gsm' (language 'en')
    -- <SIP/1-08598b38> Playing 'digits/0.gsm' (language 'en')
    -- <SIP/1-08598b38> Playing 'digits/1.gsm' (language 'en')
localhost*CLI> 
    -- Added extension '701' priority 1 to parkedcalls (0xb7c2c7e0)
<SIP/1-08598b38>AGI Tx >> 200 result=10
<SIP/1-08598b38>AGI Rx << HANGUP
<SIP/1-08598b38>AGI Tx >> 511 Command Not Permitted on a dead channel
    -- <SIP/1-08598b38>AGI Script agi://127.0.0.1 completed, returning 10
[Jan 12 08:51:26] WARNING[18765]: pbx.c:3778 __ast_pbx_run: Channel
'SIP/1-08598b38' sent into invalid extension 's' in context 'phones', but
no invalid handler
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
Disconnected from Asterisk server
Executing last minute cleanups
[root at Asterisk 1.6 DCM: tmp]$ /data/appalachian/usr/sbin/safe_asterisk:
line 138: 16118 Segmentation fault      (core dumped) nice -n $PRIORITY
${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
cat: /data/appalachian/var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
mpg123: no process killed


Transfer using the blind option on the Polycom produced this CLI output:

localhost*CLI> 
    -- Accepting call from '5555551212' to '5554441212' on channel 0/1,
span 1
[Jan 12 08:52:52] WARNING[18798]: chan_dahdi.c:1673 dahdi_enable_ec:
Unable to enable echo cancellation on channel 1 (No such device)
    -- Executing [5554441212 at inbound:1] Set("DAHDI/1-1",
"NumberCalled=5554441212") in new stack
    -- Executing [5554441212 at inbound:2] Wait("DAHDI/1-1", "2") in new
stack
    -- Executing [5554441212 at inbound:3] AGI("DAHDI/1-1",
"agi://127.0.0.1") in new stack
AGI Tx >> agi_network: yes
<DAHDI/1-1>AGI Tx >> agi_request: agi://127.0.0.1
<DAHDI/1-1>AGI Tx >> agi_channel: DAHDI/1-1
<DAHDI/1-1>AGI Tx >> agi_language: en
<DAHDI/1-1>AGI Tx >> agi_type: DAHDI
<DAHDI/1-1>AGI Tx >> agi_uniqueid: 1231771972.0
<DAHDI/1-1>AGI Tx >> agi_version: 1.6.0.3
<DAHDI/1-1>AGI Tx >> agi_callerid: 5555551212
<DAHDI/1-1>AGI Tx >> agi_calleridname: BUSINESS
<DAHDI/1-1>AGI Tx >> agi_callingpres: 0
<DAHDI/1-1>AGI Tx >> agi_callingani2: 0
<DAHDI/1-1>AGI Tx >> agi_callington: 33
<DAHDI/1-1>AGI Tx >> agi_callingtns: 0
<DAHDI/1-1>AGI Tx >> agi_dnid: 5554441212
<DAHDI/1-1>AGI Tx >> agi_rdnis: unknown
<DAHDI/1-1>AGI Tx >> agi_context: inbound
<DAHDI/1-1>AGI Tx >> agi_extension: 5554441212
<DAHDI/1-1>AGI Tx >> agi_priority: 3
<DAHDI/1-1>AGI Tx >> agi_enhanced: 0.0
<DAHDI/1-1>AGI Tx >> agi_accountcode: 
<DAHDI/1-1>AGI Tx >> agi_threadid: -1217868912
<DAHDI/1-1>AGI Tx >> 
<DAHDI/1-1>AGI Rx << ANSWER ""
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Set "NumberCalled="
    -- AGI Script Executing Application: (Set) Options: (NumberCalled=)
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Ringing ""
    -- AGI Script Executing Application: (Ringing) Options: ()
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Wait "3"
    -- AGI Script Executing Application: (Wait) Options: (3)
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Dial "SIP/1,15,rtT"
    -- AGI Script Executing Application: (Dial) Options: (SIP/1,15,rtT)
  == Using SIP RTP CoS mark 5
    -- Called 1
    -- SIP/1-097e7ff0 is ringing
    -- SIP/1-097e7ff0 answered DAHDI/1-1
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
<DAHDI/1-1>AGI Tx >> 200 result=-1
  == Parked DAHDI/1-1 on 701 at parkedcalls. Will timeout back to extension
[inbound] 5554441212, 3 in 45 seconds
<Parking/DAHDI/1-1<ZOMBIE>>AGI Rx << HANGUP
<Parking/DAHDI/1-1<ZOMBIE>>AGI Tx >> 511 Command Not Permitted on a dead
channel
    -- <SIP/1-097e7ff0> Playing 'digits/7.gsm' (language 'en')
    -- <Parking/DAHDI/1-1<ZOMBIE>>AGI Script agi://127.0.0.1 completed,
returning -1
    -- <SIP/1-097e7ff0> Playing 'digits/0.gsm' (language 'en')
    -- <SIP/1-097e7ff0> Playing 'digits/1.gsm' (language 'en')
localhost*CLI> *** glibc detected *** /data/appalachian/usr/sbin/asterisk:
free(): invalid pointer: 0x097ef50c ***
======= Backtrace: =========
/lib/libc.so.6[0x6f78b6]
/lib/libc.so.6(cfree+0x90)[0x6fae00]
/data/appalachian/usr/lib/asterisk/modules/chan_sip.so[0x1efe1d]
/data/appalachian/usr/sbin/asterisk[0x812f10b]
/lib/libpthread.so.0[0x80743b]
/lib/libc.so.6(clone+0x5e)[0x75efde]
======= Memory map: ========
00110000-00116000 r-xp 00000000 08:07 6620967   
/data/appalachian/usr/lib/asterisk/modules/res_monitor.so
00116000-00117000 rwxp 00005000 08:07 6620967   
/data/appalachian/usr/lib/asterisk/modules/res_monitor.so
00117000-0011b000 r-xp 00000000 08:07 6620962   
/data/appalachian/usr/lib/asterisk/modules/res_config_curl.so
0011b000-0011c000 rwxp 00003000 08:07 6620962   
/data/appalachian/usr/lib/asterisk/modules/res_config_curl.so
0011c000-00157000 r-xp 00000000 08:01 233683    
/usr/lib/libcurl.so.3.0.0
00157000-00158000 rwxp 0003b000 08:01 233683    
/usr/lib/libcurl.so.3.0.0
00158000-00188000 r-xp 00000000 08:01 235121    
/usr/lib/libidn.so.11.5.19
00188000-00189000 rwxp 0002f000 08:01 235121    
/usr/lib/libidn.so.11.5.19
00189000-0018b000 r-xp 00000000 08:07 6620925   
/data/appalachian/usr/lib/asterisk/modules/func_cut.so
0018b000-0018c000 rwxp 00001000 08:07 6620925   
/data/appalachian/usr/lib/asterisk/modules/func_cut.so
0018c000-0018e000 r-xp 00000000 08:07 6620916   
/data/appalachian/usr/lib/asterisk/modules/format_vox.so
0018e000-0018f000 rwxp 00001000 08:07 6620916   
/data/appalachian/usr/lib/asterisk/modules/format_vox.so
0018f000-00191000 r-xp 00000000 08:07 6620883   
/data/appalachian/usr/lib/asterisk/modules/cdr_custom.so
00191000-00192000 rwxp 00001000 08:07 6620883   
/data/appalachian/usr/lib/asterisk/modules/cdr_custom.so
00192000-00193000 rwxp 00192000 00:00 0 
00193000-001af000 r-xp 00000000 08:01 243731    
/usr/lib/libvorbis.so.0.3.1
001af000-001bd000 rwxp 0001c000 08:01 243731    
/usr/lib/libvorbis.so.0.3.1
001bd000-001c0000 r-xp 00000000 08:07 6620809   
/data/appalachian/usr/lib/asterisk/modules/app_authenticate.so
001c0000-001c1000 rwxp 00002000 08:07 6620809   
/data/appalachian/usr/lib/asterisk/modules/app_authenticate.so
001c1000-0023f000 r-xp 00000000 08:07 6620892   
/data/appalachian/usr/lib/asterisk/modules/chan_sip.so
0023f000-00241000 rwxp 0007d000 08:07 6620892   
/data/appalachian/usr/lib/asterisk/modules/chan_sip.so
00241000-00242000 rwxp 00241000 00:00 0 
00242000-0024b000 r-xp 00000000 08:01 654121     /lib/libnss_files-2.5.so
0024b000-0024c000 r-xp 00008000 08:01 654121     /lib/libnss_files-2.5.so
0024c000-0024d000 rwxp 00009000 08:01 654121     /lib/libnss_files-2.5.so
0024d000-0024f000 r-xp 00000000 08:07 6620896   
/data/appalachian/usr/lib/asterisk/modules/codec_adpcm.so
0024f000-00250000 rwxp 00001000 08:07 6620896   
/data/appalachian/usr/lib/asterisk/modules/codec_adpcm.so
00250000-00251000 r-xp 00000000 08:07 6620948   
/data/appalachian/usr/lib/asterisk/modules/func_uri.so
00251000-00252000 rwxp 00000000 08:07 6620948   
/data/appalachian/usr/lib/asterisk/modules/func_uri.so
00252000-00255000 r-xp 00000000 08:07 6620936   
/data/appalachian/usr/lib/asterisk/modules/func_lock.so
00255000-00256000 rwxp 00002000 08:07 6620936   
/data/appalachian/usr/lib/asterisk/modules/func_lock.so
00256000-00262000 r-xp 00000000 08:07 6620806   
/data/appalachian/usr/lib/asterisk/modules/app_adsiprog.so
00262000-00263000 rwxp 0000b000 08:07 6620806   
/data/appalachian/usr/lib/asterisk/modules/app_adsiprog.so
00263000-00265000 r-xp 00000000 08:07 6620873   
/data/appalachian/usr/lib/asterisk/modules/app_url.so
00265000-00266000 rwxp 00001000 08:07 6620873   
/data/appalachian/usr/lib/asterisk/modules/app_url.so
00266000-00268000 r-xp 00000000 08:07 6620884   
/data/appalachian/usr/lib/asterisk/modules/cdr_manager.so
00268000-00269000 rwxp 00001000 08:07 6620884   
/data/appalachian/usr/lib/asterisk/modules/cdr_manager.so
00269000-0026b000 r-xp 00000000 08:07 6620874   
/data/appalachian/usr/lib/asterisk/modules/app_userevent.so


0026b000-00
Disconnected from Asterisk server
Executing last minute cleanups
/data/appalachian/usr/sbin/safe_asterisk: line 138: 18780 Aborted         
       (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} >&/dev/${TTY} </dev/${TTY}
[root at Asterisk 1.6 DCM: tmp]$ Asterisk ended with exit status 134
Asterisk exited on signal 6.
cat: /data/appalachian/var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
mpg123: no process killed

I have the core for this file too if you need it.


The last method I tried was setting *5 to blind transfer in the
features.conf file.  This one acts weird.  It looks like the call makes it
into the parking lot, but then falls immediately through to the person's
voicemail.  When I try and hang up the call from the source, asterisk
crashes.  I realize this may have something to do with how I'm processing
calls, but I wouldn't think even a misconfiguration should cause an
asterisk crash.  Here's the CLI output:

localhost*CLI> 
    -- Executing [5554441212 at inbound:3] AGI("DAHDI/1-1",
"agi://127.0.0.1") in new stack
AGI Tx >> agi_network: yes
<DAHDI/1-1>AGI Tx >> agi_request: agi://127.0.0.1
<DAHDI/1-1>AGI Tx >> agi_channel: DAHDI/1-1
<DAHDI/1-1>AGI Tx >> agi_language: en
<DAHDI/1-1>AGI Tx >> agi_type: DAHDI
<DAHDI/1-1>AGI Tx >> agi_uniqueid: 1231772770.4
<DAHDI/1-1>AGI Tx >> agi_version: 1.6.0.3
<DAHDI/1-1>AGI Tx >> agi_callerid: 5555551212
<DAHDI/1-1>AGI Tx >> agi_calleridname: BUSINESS
<DAHDI/1-1>AGI Tx >> agi_callingpres: 0
<DAHDI/1-1>AGI Tx >> agi_callingani2: 0
<DAHDI/1-1>AGI Tx >> agi_callington: 33
<DAHDI/1-1>AGI Tx >> agi_callingtns: 0
<DAHDI/1-1>AGI Tx >> agi_dnid: 5554441212
<DAHDI/1-1>AGI Tx >> agi_rdnis: unknown
<DAHDI/1-1>AGI Tx >> agi_context: inbound
<DAHDI/1-1>AGI Tx >> agi_extension: 5554441212
<DAHDI/1-1>AGI Tx >> agi_priority: 3
<DAHDI/1-1>AGI Tx >> agi_enhanced: 0.0
<DAHDI/1-1>AGI Tx >> agi_accountcode: 
<DAHDI/1-1>AGI Tx >> agi_threadid: -1217311856
<DAHDI/1-1>AGI Tx >> 
<DAHDI/1-1>AGI Rx << ANSWER ""
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Set "NumberCalled="
    -- AGI Script Executing Application: (Set) Options: (NumberCalled=)
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Ringing ""
    -- AGI Script Executing Application: (Ringing) Options: ()
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Wait "3"
    -- AGI Script Executing Application: (Wait) Options: (3)
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
localhost*CLI> 
<DAHDI/1-1>AGI Tx >> 200 result=0
<DAHDI/1-1>AGI Rx << EXEC Dial "SIP/1,15,rtT"
    -- AGI Script Executing Application: (Dial) Options: (SIP/1,15,rtT)
  == Using SIP RTP CoS mark 5
    -- Called 1
    -- SIP/1-0a165780 is ringing
    -- SIP/1-0a165780 answered DAHDI/1-1
    -- <SIP/1-0a165780> Playing 'pbx-transfer.gsm' (language 'en')
  == Parked DAHDI/1-1 on 701 at parkedcalls. Will timeout back to extension
[inbound] 5554441212, 3 in 45 seconds
    -- <SIP/1-0a165780> Playing 'digits/7.gsm' (language 'en')
    -- <SIP/1-0a165780> Playing 'digits/0.gsm' (language 'en')
    -- <SIP/1-0a165780> Playing 'digits/1.gsm' (language 'en')
    -- Added extension '701' priority 1 to parkedcalls (0xa149430)
<DAHDI/1-1>AGI Tx >> 200 result=10
<DAHDI/1-1>AGI Rx << EXEC Voicemail "125,u"
    -- AGI Script Executing Application: (Voicemail) Options: (125,u)
    -- <DAHDI/1-1> Playing
'/data/appalachian/var/spool/asterisk/voicemail/default/125/unavail.slin'
(language 'en')
    -- <DAHDI/1-1> Playing 'vm-intro.gsm' (language 'en')
    -- Channel 0/1, span 1 got hangup request, cause 16
<DAHDI/1-1>AGI Tx >> 200 result=-1
  == DAHDI/1-1 got tired of being parked
    -- Hungup 'DAHDI/1-1'
[Jan 12 09:06:36] WARNING[19097]: channel.c:1319 ast_channel_free: PBX may
not have been terminated properly on 'DAHDI/1-1'
<>AGI Rx << HANGUP
<>AGI Tx >> 511 Command Not Permitted on a dead channel

Disconnected from Asterisk server
/data/appalachian/usr/sbin/safe_asterisk: line 138: 19085 Segmentation
fault      (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f
${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}
Asterisk ended with exit status 139
Executing last minute cleanups
Asterisk exited on signal 11.
cat: /data/appalachian/var/run/asterisk.pid: No such file or directory
[root at Asterisk 1.6 DCM: tmp]$ Automatically restarting Asterisk.
mpg123: no process killed


I'll add all of my config files below..about the only thing I won't be
adding is the agi script I'm using.  If there's more information that you
need, please let me know.


====================================================================== 

---------------------------------------------------------------------- 
 (0098057) svnbot (reporter) - 2009-01-16 16:16
 http://bugs.digium.com/view.php?id=14215#c98057 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 168941

_U  trunk/
U   trunk/main/features.c

------------------------------------------------------------------------
r168941 | twilson | 2009-01-16 16:16:21 -0600 (Fri, 16 Jan 2009) | 19
lines

Merged revisions 168716 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12
lines
  
  Convert call to park_call_full to masq_park_call_announce
  
  Since we removed the AST_PBX_KEEPALIVE return value, we need to use
masqueraded
  parking, otherwise we will try to call ast_hangup() in __pbx_run() and
in
  do_parking_thread() and then promptly crash.
  (closes issue http://bugs.digium.com/view.php?id=14215)
  	Reported by: waverly360	
  	Tested by: otherwiseguy
  (closes issue http://bugs.digium.com/view.php?id=14228)
  	Reported by: kobaz
  	Tested by: otherwiseguy
........

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http://svn.digium.com/view/asterisk?view=rev&revision=168941 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-16 16:16 svnbot         Checkin                                      
2009-01-16 16:16 svnbot         Note Added: 0098057                          
======================================================================




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