[asterisk-bugs] [JIRA] (ASTERISK-23676) autodestruct error

Sat Apr 26 06:39:18 CDT 2014

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23676?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217631#comment-217631 ] 

SAN commented on ASTERISK-23676:
--------------------------------

I have atteched links for my asterisk config files & screen shot of my error uploaded 
http://www.sendspace.com/filegroup/AEQuBKXg8Qr4nl42yWKEwA

> autodestruct error 
> -------------------
>
>                 Key: ASTERISK-23676
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23676
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>    Affects Versions: 1.8.23.1
>         Environment: centos 6.5 Digium pri quard card 4 ports
>            Reporter: SAN
>
> 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
> [Apr 17 10:36:49] WARNING[19994]: chan_sip.c:4213 __sip_autodestruct: Autodestruct on dialog '0efcead16431b4b36227baa3122c0767 at 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 at 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 at 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 at default:1] Set("SIP/4144-0000002c", "CHANNEL(userfield)=ETP") in new stack
> -- Executing [009400596492 at default:2] CELGenUserEvent("SIP/4144-0000002c", "LOCATION,10501") in new stack
> -- Executing [009400596492 at default:3] CELGenUserEvent("SIP/4144-0000002c", "RECORD,17042014-1397711213.90559") in new stack
> -- Executing [009400596492 at default:4] CELGenUserEvent("SIP/4144-0000002c", "EMPID,1873") in new stack
> -- Executing [009400596492 at default:5] CELGenUserEvent("SIP/4144-0000002c", "MATRIMONY,E2321594") in new stack
> -- Executing [009400596492 at default:6] CELGenUserEvent("SIP/4144-0000002c", "CHANNEL,2") in new stack
> -- Executing [009400596492 at default:7] CELGenUserEvent("SIP/4144-0000002c", "BRANCH,5") in new stack
> -- Executing [009400596492 at 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 at 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 at 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 at bmchn2asterisk ~]$
> For your reference have pasted below sip settings 
> 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


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