Summary: | ASTERISK-23676: SIP Channel driver emits autodestruct WARNING with owner in place: channel reference leak | ||
Reporter: | SAN (sanindia) | Labels: | |
Date Opened: | 2014-04-26 06:35:36 | Date Closed: | 2014-05-19 08:54:27 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | 1.8.23.1 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | centos 6.5 Digium pri quard card 4 ports | Attachments: | |
Description: | i'm struggling to fix the error in asterisk.
Note : I have not installed vicidial pls dont mistake me im unable find solution that's why posting error . I hope you guys kindly help me to fix the error * Intel(R) Xeon(R) CPU E5-2403 0 @ 1.80GHz * HDD = 500 Gb * Ram = 16 GB * asterisk version = Asterisk 1.8.23.1 In this configuration how agents could able to use for concurrent calling may i know . I have a doubt is that happening bcoz of overloading server ??? Below have mentioned error what we are facing in our asterisk server very often Asterisk 1.8.23.1 {noformat} [Apr 17 10:36:49] WARNING[19994]: chan_sip.c:4213 __sip_autodestruct: Autodestruct on dialog '0efcead16431b4b36227baa3122c0767@192.168.11.2:5060' with owner SIP/6024-0000002a in place (Method: BYE). Rescheduling destruction for 10000 ms [Apr 17 10:36:51] WARNING[19994]: chan_sip.c:4213 __sip_autodestruct: Autodestruct on dialog '350c94e22bc0687d179e2eac23d59b12@192.168.11.2:5060' with owner SIP/4144-00000024 in place (Method: BYE). Rescheduling destruction for 10000 ms == Manager 'tevatel' logged on from 192.168.11.2 == Using SIP RTP CoS mark 5 == Manager 'tevatel' logged on from 192.168.11.2 == Using SIP RTP CoS mark 5 [Apr 17 10:36:54] WARNING[19994]: chan_sip.c:4213 __sip_autodestruct: Autodestruct on dialog '69496cf70eb35c057d9b1b3a0654f750@192.168.11.2:5060' with owner SIP/4135-0000002b in place (Method: BYE). Rescheduling destruction for 10000 ms -- Got SIP response 486 "Busy Here" back from 192.168.10.43:5060 > Channel SIP/4144-0000002d was never answered. > Channel SIP/4144-0000002c was answered. -- Executing [009400596492@default:1] Set("SIP/4144-0000002c", "CHANNEL(userfield)=ETP") in new stack -- Executing [009400596492@default:2] CELGenUserEvent("SIP/4144-0000002c", "LOCATION,10501") in new stack -- Executing [009400596492@default:3] CELGenUserEvent("SIP/4144-0000002c", "RECORD,17042014-1397711213.90559") in new stack -- Executing [009400596492@default:4] CELGenUserEvent("SIP/4144-0000002c", "EMPID,1873") in new stack -- Executing [009400596492@default:5] CELGenUserEvent("SIP/4144-0000002c", "MATRIMONY,E2321594") in new stack -- Executing [009400596492@default:6] CELGenUserEvent("SIP/4144-0000002c", "CHANNEL,2") in new stack -- Executing [009400596492@default:7] CELGenUserEvent("SIP/4144-0000002c", "BRANCH,5") in new stack -- Executing [009400596492@default:8] MixMonitor("SIP/4144-0000002c", "10501-1873-E2321594-17042014-1397711213.90559.wav,a") in new stack == Begin MixMonitor Recording SIP/4144-0000002c -- Executing [009400596492@default:9] Dial("SIP/4144-0000002c", "DAHDI/g12/09400596492,,TtoR") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g12/09400596492 -- DAHDI/i2/09400596492-2b is proceeding passing it to SIP/4144-0000002c -- DAHDI/i2/09400596492-2b is making progress passing it to SIP/4144-0000002c [Apr 17 10:36:56] WARNING[19994]: chan_sip.c:4213 __sip_autodestruct: Autodestruct on dialog '0efcead16431b4b36227baa3122c0767@192.168.11.2:5060' with owner SIP/6024-0000002a in place (Method: BYE). Rescheduling destruction for 10000 ms -- DAHDI/i2/09400596492-2b is ringing bmchn2asterisk*CLI> exit Disconnected from Asterisk server Executing last minute cleanups [sysadmin@bmchn2asterisk ~]$ {noformat} For your reference have pasted below sip settings {noformat} Global Settings: ---------------- UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: Disabled TLS SIP Bindaddress: Disabled Videosupport: No Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: No Allow promisc. redir: No Enable call counters: No SIP domain support: No Realm. auth: No Our auth realm 10501 Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk PBX 1.8.23.1 SDP Session Name: Asterisk PBX 1.8.23.1 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Legacy userfield parse: No Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: Off Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 60000 ms Q.850 Reason header: No Store SIP_CAUSE: No Network QoS Settings: --------------------------- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: No Network Settings: --------------------------- SIP address remapping: Disabled, no localnet list Externhost: <none> Externaddr: (null) Externrefresh: 10 Global Signalling Settings: --------------------------- Codecs: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) Codec Order: none Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 60 RTP Hold Timeout: 300 MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: Yes Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Include CID: No Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy: <not set> Session Timers: Originate Session Refresher: uas Session Expires: 60 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B: 32000 No premature media: Yes Max forwards: 70 Default Settings: ----------------- Allowed transports: UDP Outbound transport: UDP Context: default Force rport: Yes DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk 14sudharsan Posts: 32 Joined: Fri Oct 01, 2010 3:45 pm {noformat} I have atteched links for my asterisk config files & screen shot of my error uploaded for you reference http://www.sendspace.com/filegroup/AEQuBKXg8Qr4nl42yWKEwA | ||
Comments: | By: Matt Jordan (mjordan) 2014-04-29 16:10:03.081-0500 Please don't post to an external location for your configuration files. External locations have a tendency to go away; the files need to be attached to this issue so that a developer can look at them when they try to fix this issue. Autodestruct errors typically indicate that a reference to a channel is still hanging around, preventing its destruction. In such a case, two things will need to be provided: # A scenario that reproduces the error in a consistent manner. # Ideally, a REF_DEBUG file generated during the error state. If you are using the latest Asterisk 1.8 (from the branch), you should be able to generate a REF_DEBUG file by selecting REF_DEBUG in menuselect, re-compiling, and re-producing the problem. Otherwise, you will need to follow the instructions on the wiki here: https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging By: Rusty Newton (rnewton) 2014-05-19 08:54:00.533-0500 Thank you for the bug report. However I am unable to reproduce this issue. We are now going to close this report - please feel free to reopen when you have more information at hand. |