[asterisk-bugs] [JIRA] Updated: (ASTERISK-20433) asterisk segementation fault on openwrt with uclibc / eglibc
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Wed Sep 19 19:52:27 CDT 2012
[ https://issues.asterisk.org/jira/browse/ASTERISK-20433?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-20433:
------------------------------------
Description:
sip.conf:
[8001]
type=friend
secret=123456
host=dynamic
context=default
[8002]
type=friend
secret=123456
host=dynamic
context=default
extensions.conf
exten => _8X.,1,dial(SIP/${EXTEN},40,Tt)
test follow:
make call from 8001 to 8002, after answer, press # to blind transfer, system will segementation fault after playback.
test result:
test1 segementation fault
test2 segementation fault
test3 segementation fault
test4 segementation fault
test5 works fine!!! not segfault!!!
background information: in uclibc 0.9.33.2 has an NTPL implement
i test with gdb in test2(x86):
Unknown RTP codec 126 received from '172.16.0.99:49580'
Unknown RTP codec 126 received from '172.16.0.99:49580'
Unknown RTP codec 126 received from '172.16.0.99:49580'
DTMF begin '#' received on SIP/8002-00000001
DTMF begin passthrough '#' on SIP/8002-00000001
DTMF end '#' received on SIP/8002-00000001, duration 120 ms
DTMF end accepted with begin '#' on SIP/8002-00000001
DTMF end passthrough '#' on SIP/8002-00000001
-- Started music on hold, class 'default', on SIP/8001-00000000
-- Stopped music on hold on SIP/8002-00000001
-- <SIP/8002-00000001> Playing 'pbx-transfer.alaw' (language 'en')
[New LWP 14314]
was:
sip.conf:
[8001]
type=friend
secret=123456
host=dynamic
context=default
[8002]
type=friend
secret=123456
host=dynamic
context=default
extensions.conf
exten => _8X.,1,dial(SIP/${EXTEN},40,Tt)
test follow:
make call from 8001 to 8002, after answer, press # to blind transfer, system will segementation fault after playback.
test result:
test1 segementation fault
test2 segementation fault
test3 segementation fault
test4 segementation fault
test5 works fine!!! not segfault!!!
background information: in uclibc 0.9.33.2 has an NTPL implement
i test with gdb in test2(x86):
Unknown RTP codec 126 received from '172.16.0.99:49580'
Unknown RTP codec 126 received from '172.16.0.99:49580'
Unknown RTP codec 126 received from '172.16.0.99:49580'
DTMF begin '#' received on SIP/8002-00000001
DTMF begin passthrough '#' on SIP/8002-00000001
DTMF end '#' received on SIP/8002-00000001, duration 120 ms
DTMF end accepted with begin '#' on SIP/8002-00000001
DTMF end passthrough '#' on SIP/8002-00000001
-- Started music on hold, class 'default', on SIP/8001-00000000
-- Stopped music on hold on SIP/8002-00000001
-- <SIP/8002-00000001> Playing 'pbx-transfer.alaw' (language 'en')
[New LWP 14314]
Program received signal SIGSEGV, Segmentation fault.
[Switching to LWP 14314]
0xb70adc4a in ?? () from /usr/lib/asterisk/modules/res_rtp_asterisk.so
(gdb) bt
#0 0xb70adc4a in ?? () from /usr/lib/asterisk/modules/res_rtp_asterisk.so
#1 0xb70b03d1 in ?? () from /usr/lib/asterisk/modules/res_rtp_asterisk.so
#2 0xb6e5d760 in ?? () from /usr/lib/asterisk/modules/chan_sip.so
#3 0x09550870 in ?? ()
#4 0x00000001 in ?? ()
#5 0xb6ee20bf in ?? () from /usr/lib/asterisk/modules/chan_sip.so
#6 0xb6ed1c89 in ?? () from /usr/lib/asterisk/modules/chan_sip.so
#7 0x0956cbe8 in ?? ()
#8 0x00000000 in ?? ()
(gdb) bt full
#0 0xb70adc4a in ?? () from /usr/lib/asterisk/modules/res_rtp_asterisk.so
No symbol table info available.
#1 0xb70b03d1 in ?? () from /usr/lib/asterisk/modules/res_rtp_asterisk.so
No symbol table info available.
#2 0xb6e5d760 in ?? () from /usr/lib/asterisk/modules/chan_sip.so
No symbol table info available.
#3 0x09550870 in ?? ()
No symbol table info available.
#4 0x00000001 in ?? ()
No symbol table info available.
#5 0xb6ee20bf in ?? () from /usr/lib/asterisk/modules/chan_sip.so
No symbol table info available.
#6 0xb6ed1c89 in ?? () from /usr/lib/asterisk/modules/chan_sip.so
No symbol table info available.
#7 0x0956cbe8 in ?? ()
No symbol table info available.
#8 0x00000000 in ?? ()
No symbol table info available.
(gdb) thread apply all bt
Thread 1 (LWP 14314):
#0 0xb70adc4a in ?? () from /usr/lib/asterisk/modules/res_rtp_asterisk.so
#1 0xb70b03d1 in ?? () from /usr/lib/asterisk/modules/res_rtp_asterisk.so
#2 0xb6e5d760 in ?? () from /usr/lib/asterisk/modules/chan_sip.so
#3 0x09550870 in ?? ()
#4 0x00000001 in ?? ()
#5 0xb6ee20bf in ?? () from /usr/lib/asterisk/modules/chan_sip.so
#6 0xb6ed1c89 in ?? () from /usr/lib/asterisk/modules/chan_sip.so
#7 0x0956cbe8 in ?? ()
#8 0x00000000 in ?? ()
(gdb)
> asterisk segementation fault on openwrt with uclibc / eglibc
> ------------------------------------------------------------
>
> Key: ASTERISK-20433
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-20433
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Resources/res_features, Resources/res_rtp_asterisk
> Affects Versions: 1.8.10.1
> Environment: test1: openwrt r33444, ramips, kernel 3.3.8, uclibc 0.9.33.2, asterisk-1.8.10.1
> test2: openwrt r33444, x86(virtualbox), kernel 3.3.8, eglibc 2.1.5, asterisk-1.8.10.1
> test3: openwrt r33444, ramips, kernel 3.3.8, uclibc 0.9.33.2, asterisk-1.8.15.1
> test4: openwrt r33444, ramips, kernel 2.6.35, uclibc 0.9.33.2, asterisk-1.8.4.4
> test5: openwrt 10.3.1, brcm63xx, kernel 2.6.32.10, uclibc 0.9.30.1, asterisk-1.6.2
> Reporter: sun bing
> Assignee: sun bing
> Attachments: backtrace.txt
>
>
> sip.conf:
> [8001]
> type=friend
> secret=123456
> host=dynamic
> context=default
> [8002]
> type=friend
> secret=123456
> host=dynamic
> context=default
> extensions.conf
> exten => _8X.,1,dial(SIP/${EXTEN},40,Tt)
> test follow:
> make call from 8001 to 8002, after answer, press # to blind transfer, system will segementation fault after playback.
> test result:
> test1 segementation fault
> test2 segementation fault
> test3 segementation fault
> test4 segementation fault
> test5 works fine!!! not segfault!!!
> background information: in uclibc 0.9.33.2 has an NTPL implement
> i test with gdb in test2(x86):
> Unknown RTP codec 126 received from '172.16.0.99:49580'
> Unknown RTP codec 126 received from '172.16.0.99:49580'
> Unknown RTP codec 126 received from '172.16.0.99:49580'
> DTMF begin '#' received on SIP/8002-00000001
> DTMF begin passthrough '#' on SIP/8002-00000001
> DTMF end '#' received on SIP/8002-00000001, duration 120 ms
> DTMF end accepted with begin '#' on SIP/8002-00000001
> DTMF end passthrough '#' on SIP/8002-00000001
> -- Started music on hold, class 'default', on SIP/8001-00000000
> -- Stopped music on hold on SIP/8002-00000001
> -- <SIP/8002-00000001> Playing 'pbx-transfer.alaw' (language 'en')
> [New LWP 14314]
--
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