[asterisk-bugs] [JIRA] (ASTERISK-22056) Asterisk 11.4.0 : Queue RINGNOANSWER wrong ring time when one of the peer becomes unresponsive
Nikita Zogas (JIRA)
noreply at issues.asterisk.org
Tue Jul 16 02:58:03 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22056?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Nikita Zogas updated ASTERISK-22056:
------------------------------------
Description:
When Queue application dials to one agent (peer) which becomes unresponsive during call, another agent's phone rings for remaining time of queue_timeout but still logs full timeout to second agent.
To illustrate, I attach screenshots of MySQL database and full log extract. Here is what is seen in the log:
[Jun 21 09:53:09] VERBOSE[14467][C-00004b41] pbx.c: -- Executing [2 at sodra-ivr-work:2] Queue("SIP/testlink-00005981", "pasalpos") in new stack
[Jun 21 09:53:09] VERBOSE[14467][C-00004b41] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/testlink-00005981
[Jun 21 09:53:09] WARNING[14467][C-00004b41] translate.c: no samples for alawtolin
[Jun 21 09:53:09] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:1] NoOp("Local/316 at agents-00003418;2", "Dial to agent 316") in new stack
[Jun 21 09:53:09] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:2] Dial("Local/316 at agents-00003418;2", "SIP/316") in new stack
[Jun 21 09:53:09] VERBOSE[14469][C-00004b41] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 21 09:53:09] VERBOSE[14469][C-00004b41] app_dial.c: -- Called SIP/316
SIP/316 is called and then got message in the log:
[Jun 21 09:53:15] WARNING[5650] chan_sip.c: Retransmission timeout reached on transmission 1e3ed80f5a8fd5976b56602868f6501a at 192.168.88.2:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Jun 21 09:53:15] WARNING[5650] chan_sip.c: Hanging up call 1e3ed80f5a8fd5976b56602868f6501a at 192.168.88.2:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Jun 21 09:53:15] VERBOSE[14469][C-00004b41] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[Jun 21 09:53:15] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:3] Hangup("Local/316 at agents-00003418;2", "") in new stack
[Jun 21 09:53:15] VERBOSE[14469][C-00004b41] pbx.c: == Spawn extension (agents, 316, 3) exited non-zero on 'Local/316 at agents-00003418;2'
[Jun 21 09:53:15] VERBOSE[14467][C-00004b41] app_queue.c: -- Nobody picked up in 6000 ms
then starts new call, as you see, call start and end difference is 2 seconds whereas Asterisk states "Nobody picked up in 8000 ms"
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] pbx.c: -- Executing [330 at agents:1] NoOp("Local/330 at agents-00003419;2", "Dial to agent 330") in new stack
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] pbx.c: -- Executing [330 at agents:2] Dial("Local/330 at agents-00003419;2", "SIP/330") in new stack
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] app_dial.c: -- Called SIP/330
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] app_dial.c: -- SIP/330-00005983 is ringing
[Jun 21 09:53:15] VERBOSE[14467][C-00004b41] app_queue.c: -- Local/330 at agents-00003419;1 is ringing
[Jun 21 09:53:16] VERBOSE[14449][C-00004b40] pbx.c: -- Executing [s at sodra-ivr-work:10] BackGround("OOH323/avaya1-10151", "sodra4") in new stack
[Jun 21 09:53:16] VERBOSE[14449][C-00004b40] file.c: -- <OOH323/avaya1-10151> Playing 'sodra4.slin' (language 'en')
[Jun 21 09:53:17] VERBOSE[14467][C-00004b41] app_queue.c: -- Nobody picked up in 8000 ms
[Jun 21 09:53:17] VERBOSE[14475][C-00004b41] pbx.c: == Spawn extension (agents, 330, 2) exited non-zero on 'Local/330 at agents-00003419;2'
[Jun 21 09:53:20] VERBOSE[14437][C-00004b39] res_musiconhold.c: -- Started music on hold, class 'default', on Local/320 at agents-00003414;2
Steps to reproduce (likely):
1. Use Queue() application and 2 logged agents
2. Force one agent to become unresponsive (e.g. unplug Ethernet cable of VoIP phone if HW phone is used)
was:
When Queue application dials to one agent (peer) which becomes unresponsive during call, another agent's phone rings for remaining time of queue_timeout but still logs full timeout to second agent.
To illustrate, I attach screenshots of MySQL database and full log extract. Here is what is seen in the log:
[Jun 21 09:53:09] VERBOSE[14467][C-00004b41] pbx.c: -- Executing [2 at sodra-ivr-work:2] Queue("SIP/testlink-00005981", "pasalpos") in new stack
[Jun 21 09:53:09] VERBOSE[14467][C-00004b41] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/testlink-00005981
[Jun 21 09:53:09] WARNING[14467][C-00004b41] translate.c: no samples for alawtolin
[Jun 21 09:53:09] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:1] NoOp("Local/316 at agents-00003418;2", "Dial to agent 316") in new stack
[Jun 21 09:53:09] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:2] Dial("Local/316 at agents-00003418;2", "SIP/316") in new stack
[Jun 21 09:53:09] VERBOSE[14469][C-00004b41] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 21 09:53:09] VERBOSE[14469][C-00004b41] app_dial.c: -- Called SIP/316
<b> SIP/316 is called and then got message in the log: </b>
[Jun 21 09:53:15] WARNING[5650] chan_sip.c: Retransmission timeout reached on transmission 1e3ed80f5a8fd5976b56602868f6501a at 192.168.88.2:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Jun 21 09:53:15] WARNING[5650] chan_sip.c: Hanging up call 1e3ed80f5a8fd5976b56602868f6501a at 192.168.88.2:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Jun 21 09:53:15] VERBOSE[14469][C-00004b41] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[Jun 21 09:53:15] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:3] Hangup("Local/316 at agents-00003418;2", "") in new stack
[Jun 21 09:53:15] VERBOSE[14469][C-00004b41] pbx.c: == Spawn extension (agents, 316, 3) exited non-zero on 'Local/316 at agents-00003418;2'
[Jun 21 09:53:15] VERBOSE[14467][C-00004b41] app_queue.c: -- Nobody picked up in 6000 ms
then starts new call, as you see, call start and end difference is 2 seconds whereas Asterisk states "Nobody picked up in 8000 ms"
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] pbx.c: -- Executing [330 at agents:1] NoOp("Local/330 at agents-00003419;2", "Dial to agent 330") in new stack
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] pbx.c: -- Executing [330 at agents:2] Dial("Local/330 at agents-00003419;2", "SIP/330") in new stack
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] app_dial.c: -- Called SIP/330
[Jun 21 09:53:15] VERBOSE[14475][C-00004b41] app_dial.c: -- SIP/330-00005983 is ringing
[Jun 21 09:53:15] VERBOSE[14467][C-00004b41] app_queue.c: -- Local/330 at agents-00003419;1 is ringing
[Jun 21 09:53:16] VERBOSE[14449][C-00004b40] pbx.c: -- Executing [s at sodra-ivr-work:10] BackGround("OOH323/avaya1-10151", "sodra4") in new stack
[Jun 21 09:53:16] VERBOSE[14449][C-00004b40] file.c: -- <OOH323/avaya1-10151> Playing 'sodra4.slin' (language 'en')
[Jun 21 09:53:17] VERBOSE[14467][C-00004b41] app_queue.c: -- Nobody picked up in 8000 ms
[Jun 21 09:53:17] VERBOSE[14475][C-00004b41] pbx.c: == Spawn extension (agents, 330, 2) exited non-zero on 'Local/330 at agents-00003419;2'
[Jun 21 09:53:20] VERBOSE[14437][C-00004b39] res_musiconhold.c: -- Started music on hold, class 'default', on Local/320 at agents-00003414;2
Steps to reproduce (likely):
1. Use Queue() application and 2 logged agents
2. Force one agent to become unresponsive (e.g. unplug Ethernet cable of VoIP phone if HW phone is used)
> Asterisk 11.4.0 : Queue RINGNOANSWER wrong ring time when one of the peer becomes unresponsive
> ----------------------------------------------------------------------------------------------
>
> Key: ASTERISK-22056
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-22056
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Applications/app_queue
> Affects Versions: 11.4.0
> Environment: CentOS release 6.4 (Final), CPU info: Intel(R) Xeon(R) CPU E31220 @ 3.10GHz
> Reporter: Nikita Zogas
> Assignee: Nikita Zogas
> Severity: Critical
> Attachments: database_screen_2013-06-21_09-53-09.jpg, extensions_agents.conf, extensions.conf, log_extract.jpg, queues_settings_mysql1.txt
>
>
> When Queue application dials to one agent (peer) which becomes unresponsive during call, another agent's phone rings for remaining time of queue_timeout but still logs full timeout to second agent.
> To illustrate, I attach screenshots of MySQL database and full log extract. Here is what is seen in the log:
> [Jun 21 09:53:09] VERBOSE[14467][C-00004b41] pbx.c: -- Executing [2 at sodra-ivr-work:2] Queue("SIP/testlink-00005981", "pasalpos") in new stack
> [Jun 21 09:53:09] VERBOSE[14467][C-00004b41] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/testlink-00005981
> [Jun 21 09:53:09] WARNING[14467][C-00004b41] translate.c: no samples for alawtolin
> [Jun 21 09:53:09] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:1] NoOp("Local/316 at agents-00003418;2", "Dial to agent 316") in new stack
> [Jun 21 09:53:09] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:2] Dial("Local/316 at agents-00003418;2", "SIP/316") in new stack
> [Jun 21 09:53:09] VERBOSE[14469][C-00004b41] netsock2.c: == Using SIP RTP CoS mark 5
> [Jun 21 09:53:09] VERBOSE[14469][C-00004b41] app_dial.c: -- Called SIP/316
> SIP/316 is called and then got message in the log:
> [Jun 21 09:53:15] WARNING[5650] chan_sip.c: Retransmission timeout reached on transmission 1e3ed80f5a8fd5976b56602868f6501a at 192.168.88.2:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 6399ms with no response
> [Jun 21 09:53:15] WARNING[5650] chan_sip.c: Hanging up call 1e3ed80f5a8fd5976b56602868f6501a at 192.168.88.2:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
> [Jun 21 09:53:15] VERBOSE[14469][C-00004b41] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
> [Jun 21 09:53:15] VERBOSE[14469][C-00004b41] pbx.c: -- Executing [316 at agents:3] Hangup("Local/316 at agents-00003418;2", "") in new stack
> [Jun 21 09:53:15] VERBOSE[14469][C-00004b41] pbx.c: == Spawn extension (agents, 316, 3) exited non-zero on 'Local/316 at agents-00003418;2'
> [Jun 21 09:53:15] VERBOSE[14467][C-00004b41] app_queue.c: -- Nobody picked up in 6000 ms
> then starts new call, as you see, call start and end difference is 2 seconds whereas Asterisk states "Nobody picked up in 8000 ms"
> [Jun 21 09:53:15] VERBOSE[14475][C-00004b41] pbx.c: -- Executing [330 at agents:1] NoOp("Local/330 at agents-00003419;2", "Dial to agent 330") in new stack
> [Jun 21 09:53:15] VERBOSE[14475][C-00004b41] pbx.c: -- Executing [330 at agents:2] Dial("Local/330 at agents-00003419;2", "SIP/330") in new stack
> [Jun 21 09:53:15] VERBOSE[14475][C-00004b41] netsock2.c: == Using SIP RTP CoS mark 5
> [Jun 21 09:53:15] VERBOSE[14475][C-00004b41] app_dial.c: -- Called SIP/330
> [Jun 21 09:53:15] VERBOSE[14475][C-00004b41] app_dial.c: -- SIP/330-00005983 is ringing
> [Jun 21 09:53:15] VERBOSE[14467][C-00004b41] app_queue.c: -- Local/330 at agents-00003419;1 is ringing
> [Jun 21 09:53:16] VERBOSE[14449][C-00004b40] pbx.c: -- Executing [s at sodra-ivr-work:10] BackGround("OOH323/avaya1-10151", "sodra4") in new stack
> [Jun 21 09:53:16] VERBOSE[14449][C-00004b40] file.c: -- <OOH323/avaya1-10151> Playing 'sodra4.slin' (language 'en')
> [Jun 21 09:53:17] VERBOSE[14467][C-00004b41] app_queue.c: -- Nobody picked up in 8000 ms
> [Jun 21 09:53:17] VERBOSE[14475][C-00004b41] pbx.c: == Spawn extension (agents, 330, 2) exited non-zero on 'Local/330 at agents-00003419;2'
> [Jun 21 09:53:20] VERBOSE[14437][C-00004b39] res_musiconhold.c: -- Started music on hold, class 'default', on Local/320 at agents-00003414;2
> Steps to reproduce (likely):
> 1. Use Queue() application and 2 logged agents
> 2. Force one agent to become unresponsive (e.g. unplug Ethernet cable of VoIP phone if HW phone is used)
--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira
More information about the asterisk-bugs
mailing list