VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
amit at avhan.com Guest
|
Posted: Tue Nov 12, 2013 8:55 am Post subject: [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@192.168.2.135:5060 ([email]09e2d5011044076a652a743a737d64a4@192.168.2.135:5060[/email]) (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@192.168.2.135:5060 ([email]09e2d5011044076a652a743a737d64a4@192.168.2.135:5060[/email])' of Request 103: Match Found
[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: Destroying SIP dialog 09e2d5011044076a652a743a737d64a4@192.168.2.135:5060 ([email]09e2d5011044076a652a743a737d64a4@192.168.2.135:5060[/email])
[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@192.168.2.135:5060 ([email]5145d97f517ba83776c613611e8ab7ad@192.168.2.135:5060[/email]) (Checking To) --From tag as5a8db578 --To-tag 86988131
[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: Stopping retransmission on '5145d97f517ba83776c613611e8ab7ad@192.168.2.135:5060 ([email]5145d97f517ba83776c613611e8ab7ad@192.168.2.135:5060[/email])' of Request 103: Match Found
[Nov 12 16:53:02] DEBUG[3582] chan_sip.c: Destroying SIP dialog 5145d97f517ba83776c613611e8ab7ad@192.168.2.135:5060 ([email]5145d97f517ba83776c613611e8ab7ad@192.168.2.135:5060[/email])
[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@192.168.2.135:5060 ([email]4bac78b176a9f8ff375b992a6dd42379@192.168.2.135:5060[/email])
[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 [url=sip:100]sip:100[/url]' 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 [url=sip:100]sip:100[/url]' 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@192.168.2.135:5060 ([email]4bac78b176a9f8ff375b992a6dd42379@192.168.2.135:5060[/email]) (Checking To) --From tag as5104e8fc --To-tag ae6f4708
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: Stopping retransmission on '4bac78b176a9f8ff375b992a6dd42379@192.168.2.135:5060 ([email]4bac78b176a9f8ff375b992a6dd42379@192.168.2.135:5060[/email])' of Request 103: Match Found
[Nov 12 16:53:03] DEBUG[3582] chan_sip.c: Destroying SIP dialog 4bac78b176a9f8ff375b992a6dd42379@192.168.2.135:5060 ([email]4bac78b176a9f8ff375b992a6dd42379@192.168.2.135:5060[/email])
[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@192.168.2.135:5060 ([email]4bac78b176a9f8ff375b992a6dd42379@192.168.2.135:5060[/email]) (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@192.168.2.135:5060 ([email]4bac78b176a9f8ff375b992a6dd42379@192.168.2.135:5060[/email])
[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 |
|
Back to top |
|
|
mjordan at digium.com Guest
|
Posted: Tue Nov 12, 2013 10:13 am Post subject: [asterisk-users] Asterisk 1.8.20 crashing |
|
|
On Tue, Nov 12, 2013 at 7:56 AM, Amit Patkar | ATPL <amit@avhan.com (amit@avhan.com)> wrote:
Quote: |
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 |
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|