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Amit Patkar | ATPL

2013-Nov-12 13:56 UTC

[asterisk-users] Asterisk 1.8.20 crashing

Hi

I am experiencing Asterisk Crash. Log got stopped when asterisk crashed.
Please help me to identify the reason and fix this issue.

Asterisk: 1.8.20

I am using AMI and fastAGI to control the call. Some part of dial plan 
is also defined in extensions.conf

I am experiencing this crash when app_meetme conference functionality is 
used with more than 3 parties. I faced this issue with app_conference 
also. This is finally resulting in call drops.
Please help me to identify and resolve this issue.
Last few lines of log are copied in this email. I can email full log, if 
this does not give sufficient information of crash.

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


[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: = Looking for Call ID: 
09e2d5011044076a652a743a737d64a4 at 192.168.2.135:5060 (Checking To) --From 
tag as107b6076 --To-tag ff30d250
[Nov 12 16:53:02] DEBUG[23332] manager.c: Running action 'Redirect'
[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: Stopping retransmission on 
'09e2d5011044076a652a743a737d64a4 at 192.168.2.135:5060' of Request 103:
Match Found
[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: Destroying SIP dialog 
09e2d5011044076a652a743a737d64a4 at 192.168.2.135:5060
[Nov 12 16:53:02] DEBUG[3582] rtp_engine.c: Destroyed RTP instance 
'0x8985958'
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: No provider found, checking 
channel drivers for SIP - 1003
[Nov 12 16:53:02] DEBUG[3573] chan_sip.c: Checking device state for peer 
1003
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: Changing state for SIP/1003 
- state 1 (Not in use)
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: device 'SIP/1003' state
'1'
[Nov 12 16:53:02] DEBUG[23332] manager.c: Running action 'Redirect'
[Nov 12 16:53:02] DEBUG[3612] app_queue.c: Device 'SIP/1003' changed to 
state '1' (Not in use) but we don't care because they're not a
member of
any queue.
[Nov 12 16:53:02] VERBOSE[27410] pbx.c:   == Spawn extension (avhan, 
100, 1) exited non-zero on 'DAHDI/i1/110-8f'
[Nov 12 16:53:02] DEBUG[27410] channel.c: Soft-Hanging up channel 
'DAHDI/i1/110-8f'
[Nov 12 16:53:02] DEBUG[27410] channel.c: Hanging up channel 
'DAHDI/i1/110-8f'
[Nov 12 16:53:02] DEBUG[27410] chan_dahdi.c: dahdi_hangup(DAHDI/i1/110-8f)
[Nov 12 16:53:02] DEBUG[27410] chan_dahdi.c: Set option AUDIO MODE, 
value: ON(1) on DAHDI/i1/110-8f
[Nov 12 16:53:02] DEBUG[27410] sig_pri.c: sig_pri_hangup 1
[Nov 12 16:53:02] DEBUG[27410] sig_pri.c: Not yet hungup...  Calling 
hangup with cause -1, and clearing call
[Nov 12 16:53:02] DEBUG[27410] chan_dahdi.c: Disabled echo cancellation 
on channel 1
[Nov 12 16:53:02] DEBUG[27410] chan_dahdi.c: Set option TDD MODE, value: 
OFF(0) on DAHDI/i1/110-8f
[Nov 12 16:53:02] DEBUG[27410] chan_dahdi.c: Updated conferencing on 1, 
with 0 conference users
[Nov 12 16:53:02] DEBUG[27410] chan_dahdi.c: Set option AUDIO MODE, 
value: OFF(0) on DAHDI/i1/110-8f
[Nov 12 16:53:02] VERBOSE[27410] chan_dahdi.c:     -- Hungup 
'DAHDI/i1/110-8f'
[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: DAHDI/i1/110-8f
Uniqueid: 1384275182.917
CallerIDNum: 110
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Cause: 0
Cause-txt: Unknown


[Nov 12 16:53:02] DEBUG[3573] devicestate.c: No provider found, checking 
channel drivers for DAHDI - i1/110
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: Changing state for 
DAHDI/i1/110 - state 0 (Unknown)
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: device 'DAHDI/i1/110' state
'0'
[Nov 12 16:53:02] DEBUG[3612] app_queue.c: Device 'DAHDI/i1/110' changed
to state '0' (Unknown) but we don't care because they're not a
member of
any queue.
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: No provider found, checking 
channel drivers for SIP - 1003
[Nov 12 16:53:02] DEBUG[3573] chan_sip.c: Checking device state for peer 
1003
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: Changing state for SIP/1003 
- state 1 (Not in use)
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: device 'SIP/1003' state
'1'
[Nov 12 16:53:02] DEBUG[3612] app_queue.c: Device 'SIP/1003' changed to 
state '1' (Not in use) but we don't care because they're not a
member of
any queue.
[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: = Looking for Call ID: 
5145d97f517ba83776c613611e8ab7ad at 192.168.2.135:5060 (Checking To) --From 
tag as5a8db578 --To-tag 86988131
[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: Stopping retransmission on 
'5145d97f517ba83776c613611e8ab7ad at 192.168.2.135:5060' of Request 103:
Match Found
[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: Destroying SIP dialog 
5145d97f517ba83776c613611e8ab7ad at 192.168.2.135:5060
[Nov 12 16:53:02] DEBUG[3582] rtp_engine.c: Destroyed RTP instance 
'0x8987478'
[Nov 12 16:53:02] DEBUG[27368] pbx.c: Spawn extension 
(Conference,65001,1) exited non-zero on 'SIP/1001-000000b3'
[Nov 12 16:53:02] VERBOSE[27368] pbx.c:   == Spawn extension 
(Conference, 65001, 1) exited non-zero on 'SIP/1001-000000b3'
[Nov 12 16:53:02] DEBUG[27368] channel.c: Soft-Hanging up channel 
'SIP/1001-000000b3'
[Nov 12 16:53:02] DEBUG[27368] channel.c: Hanging up channel 
'SIP/1001-000000b3'
[Nov 12 16:53:02] DEBUG[27368] chan_sip.c: Hangup call 
SIP/1001-000000b3, SIP callid 
4bac78b176a9f8ff375b992a6dd42379 at 192.168.2.135:5060
[Nov 12 16:53:02] DEBUG[27368] chan_sip.c: Updating call counter for 
outgoing call
[Nov 12 16:53:02] DEBUG[27368] res_rtp_asterisk.c: Setting RTCP address 
on RTP instance '0x8984970'
[Nov 12 16:53:02] DEBUG[27368] netsock2.c: Splitting 
'192.168.2.111:2348' into...
[Nov 12 16:53:02] DEBUG[27368] netsock2.c: ...host '192.168.2.111' and 
port '2348'.
[Nov 12 16:53:02] DEBUG[27368] chan_sip.c: Trying to put 'BYE sip:100' 
onto UDP socket destined for 192.168.2.111:2348
[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: MeetmeLeave
Privilege: call,all
Channel: SIP/1001-000000b3
Uniqueid: 1384275118.914
Meetme: 65001
Usernum: 1
CallerIDNum: 110
CallerIDName: device
ConnectedLineNum: 110
ConnectedLineName: <unknown>
Duration: 64


[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/1001-000000b3
Variable: MEETMESECS
Value: 64
Uniqueid: 1384275118.914


[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/1001-000000b3
Variable: RTPAUDIOQOS
Value: 
ssrc=797521620;themssrc=278781419;lp=0;rxjitter=0.000489;rxcount=3885;txjitter=0.000000;txcount=3862;rlp=0;rtt=0.000000
Uniqueid: 1384275118.914


[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/1001-000000b3
Variable: RTPAUDIOQOSJITTER
Value: 
minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
Uniqueid: 1384275118.914


[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/1001-000000b3
Variable: RTPAUDIOQOSLOSS
Value: 
minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
Uniqueid: 1384275118.914


[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/1001-000000b3
Variable: RTPAUDIOQOSRTT
Value: minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
Uniqueid: 1384275118.914


[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/1001-000000b3
Variable: RTPAUDIOQOS
Value: 
ssrc=797521620;themssrc=278781419;lp=0;rxjitter=0.000489;rxcount=3885;txjitter=0.000000;txcount=3862;rlp=0;rtt=0.000000
Uniqueid: 1384275118.914


[Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: SIP/1001-000000b3
Uniqueid: 1384275118.914
CallerIDNum: 110
CallerIDName: device
ConnectedLineNum: 110
ConnectedLineName: <unknown>
Cause: 0
Cause-txt: Unknown


[Nov 12 16:53:02] DEBUG[3573] devicestate.c: No provider found, checking 
channel drivers for SIP - 1001
[Nov 12 16:53:02] DEBUG[3573] chan_sip.c: Checking device state for peer 
1001
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: Changing state for SIP/1001 
- state 1 (Not in use)
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: device 'SIP/1001' state
'1'
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: No provider found, checking 
channel drivers for SIP - 1001
[Nov 12 16:53:02] DEBUG[3573] chan_sip.c: Checking device state for peer 
1001
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: Changing state for SIP/1001 
- state 1 (Not in use)
[Nov 12 16:53:02] DEBUG[3573] devicestate.c: device 'SIP/1001' state
'1'
[Nov 12 16:53:02] DEBUG[3612] app_queue.c: Device 'SIP/1001' changed to 
state '1' (Not in use) but we don't care because they're not a
member of
any queue.
[Nov 12 16:53:02] DEBUG[3612] app_queue.c: Device 'SIP/1001' changed to 
state '1' (Not in use) but we don't care because they're not a
member of
any queue.
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: ** SIP timers: Rescheduling 
retransmission 2 to 202 ms (t1 101 ms (Retrans id #12871))
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: Trying to put 'BYE sip:100' 
onto UDP socket destined for 192.168.2.111:2348
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: = Looking for Call ID: 
4bac78b176a9f8ff375b992a6dd42379 at 192.168.2.135:5060 (Checking To) --From 
tag as5104e8fc --To-tag ae6f4708
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: Stopping retransmission on 
'4bac78b176a9f8ff375b992a6dd42379 at 192.168.2.135:5060' of Request 103:
Match Found
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: Destroying SIP dialog 
4bac78b176a9f8ff375b992a6dd42379 at 192.168.2.135:5060
[Nov 12 16:53:03] DEBUG[3582] rtp_engine.c: Destroyed RTP instance 
'0x8984970'
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: = Looking for Call ID: 
4bac78b176a9f8ff375b992a6dd42379 at 192.168.2.135:5060 (Checking To) --From 
tag as5104e8fc --To-tag ae6f4708
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: That's odd... Got a response 
on a call we don't know about. Callid 
4bac78b176a9f8ff375b992a6dd42379 at 192.168.2.135:5060
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: Invalid SIP message - rejected 
, no callid, len 413
[Nov 12 16:53:03] DEBUG[27367] devicestate.c: device 'meetme:65001' 
state '1'
[Nov 12 16:53:03] DEBUG[27367] channel.c: Hanging up channel 
'DAHDI/pseudo-1038151397'
[Nov 12 16:53:03] DEBUG[27367] chan_dahdi.c: 
dahdi_hangup(DAHDI/pseudo-1038151397)
[Nov 12 16:53:03] DEBUG[27367] chan_dahdi.c: Hangup: channel: -2 index = 
0, normal = 170, callwait = -1, thirdcall = -1
[Nov 12 16:53:03] DEBUG[27367] chan_dahdi.c: Set option TDD MODE, value: 
OFF(0) on DAHDI/pseudo-1038151397
[Nov 12 16:53:03] DEBUG[27367] chan_dahdi.c: Updated conferencing on -2, 
with 0 conference users
[Nov 12 16:53:03] VERBOSE[27367] chan_dahdi.c:     -- Hungup 
'DAHDI/pseudo-1038151397'
[Nov 12 16:53:03] DEBUG[27367] pbx.c: Spawn extension (avhan,100,1) 
exited non-zero on 'Bridge/SIP/1004-000000b4<ZOMBIE>'
[Nov 12 16:53:03] VERBOSE[27367] pbx.c:   == Spawn extension (avhan, 
100, 1) exited non-zero on 'Bridge/SIP/1004-000000b4<ZOMBIE>'
[Nov 12 16:53:03] DEBUG[27367] channel.c: Soft-Hanging up channel 
'Bridge/SIP/1004-000000b4<ZOMBIE>'
[Nov 12 16:53:03] DEBUG[23332] manager.c: Examining event:
Event: MeetmeLeave
Privilege: call,all
Channel: Bridge/SIP/1004-000000b4<ZOMBIE>
Uniqueid: 1384275118.913
Meetme: 65001
Usernum: 2
CallerIDNum: <unknown>
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Duration: 45


[Nov 12 16:53:03] DEBUG[23332] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: Bridge/SIP/1004-000000b4<ZOMBIE>
Variable: MEETMESECS
Value: 45
Uniqueid: 1384275118.913


[Nov 12 16:53:03] DEBUG[23332] manager.c: Examining event:
Event: MeetmeEnd
Privilege: call,all
Meetme: 65001


[Nov 12 16:53:03] DEBUG[23332] manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: DAHDI/pseudo-1038151397
Uniqueid: 1384275118.915
CallerIDNum: <unknown>
CallerIDName: <unknown>
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
Cause: 0
Cause-txt: Unknown


[Nov 12 16:53:03] DEBUG[3573] devicestate.c: No provider found, checking 
channel drivers for DAHDI - pseudo
[Nov 12 16:53:03] DEBUG[3573] devicestate.c: Changing state for 
DAHDI/pseudo - state 0 (Unknown)
[Nov 12 16:53:03] DEBUG[3573] devicestate.c: device 'DAHDI/pseudo' state
'0'
[Nov 12 16:53:03] DEBUG[3612] app_queue.c: Device 'meetme:65001' changed
to state '1' (Not in use) but we don't care because they're not
a member
of any queue.
[Nov 12 16:53:03] DEBUG[3612] app_queue.c: Device 'DAHDI/pseudo' changed
to state '0' (Unknown) but we don't care because they're not a
member of
any queue.
===========================================================================================================================================================================================
Thanks & Regards,
Amit Patkar







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Matthew Jordan

2013-Nov-12 15:13 UTC

head link

[asterisk-users] Asterisk 1.8.20 crashing

On Tue, Nov 12, 2013 at 7:56 AM, Amit Patkar | ATPL <amit at avhan.com>
wrote:
>
>  Hi
>
> I am experiencing Asterisk Crash. Log got stopped when asterisk crashed.
> Please help me to identify the reason and fix this issue.
>
> Asterisk: 1.8.20
>
> I am using AMI and fastAGI to control the call. Some part of dial plan is
> also defined in extensions.conf
>
> I am experiencing this crash when app_meetme conference functionality is
> used with more than 3 parties. I faced this issue with app_conference also.
> This is finally resulting in call drops.
> Please help me to identify and resolve this issue.
> Last few lines of log are copied in this email. I can email full log, if
> this does not give sufficient information of crash.
>
><snip>

While the log will be helpful, I'd recommend getting a backtrace from the
core produced when Asterisk crashed. Instructions on generating a proper
backtrace can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Please make sure Asterisk and its modules are compiled with DONT_OPTIMIZE
and BETTER_BACKTRACES. When you have a backtrace, please file an issue in
the Asterisk issue tracker - attach the log as well, as it may be useful in
analyzing how the system got into that state.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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