[asterisk-bugs] [JIRA] (ASTERISK-23676) autodestruct error
SAN (JIRA)
noreply at issues.asterisk.org
Sat Apr 26 06:39:18 CDT 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-23676?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
SAN updated ASTERISK-23676:
---------------------------
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
[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
I have atteched links for my asterisk config files & screen shot of my error uploaded for you reference
http://www.sendspace.com/filegroup/AEQuBKXg8Qr4nl42yWKEwA
was:
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
> 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
> I have atteched links for my asterisk config files & screen shot of my error uploaded for you reference
> http://www.sendspace.com/filegroup/AEQuBKXg8Qr4nl42yWKEwA
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