Error chan_sip.c:4213 __sip_autodestruct: Autodestruct

14sudharsan · April 17, 2014, 5:21am

Hi Team,
This is my first post in this forum . I hope you friend will help me to sort out this issue . Thanks in advance

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@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 ~]$

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 Settings:

SIP address remapping: Disabled, no localnet list
Externhost:
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:
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

david55 · April 17, 2014, 7:13am

It’s a warning, not a full error, and the the important part is the with…still in place part.

I think this is normally a bug, but in theory it could be caused by something like a very long running “h” extension.

What other symptoms are you seeing that indicate that this is causing more than a warning message?

14sudharsan · April 18, 2014, 8:04am

Thanks For your swift response david .

We are facing issue unable to make outbound calls calls from the server once the error is generated autodestruct .

later it will reflect to all the users .,

Once asterisk service restarted then oly calls are going .

Is there any possibility to fix the error

david55 · April 18, 2014, 9:49am

All calls autodestruct.

I do not understand you description of what is going wrong at an end user level.

However, I’d repeat that the only configuration issue that might cause the warning is f you don’t let the channel go away, e.g. be having a long running h extension.

Otherwise you need to look at changelogs for later version, to see if any bugs have been fixed that might also cause the channel not to be released.

14sudharsan · April 22, 2014, 11:20am

Hi David,
Thanks for your reply…
Actually this server was installed and configured by third party vendor . Not done by me .

Few days before oly i have joined in this company.

Can you pls tell how to fix the error steps .

I have atteched links for my asterisk config files & screen shot of my error uploaded

pls help me to fix this issue

sendspace.com/filegroup/AEQu … nl42yWKEwA

david55 · April 22, 2014, 11:31am

Specifically, in this case, I don’t know why you are triggering this warning, and I’m not going to search the changelogs to see if some of the possible causes have been fixed.

Basically, it might be a problem with your dialplan, which isn’t part of Asterisk.

It might be a bug that is fixed in a later version, in which case upgrading may help.

It might be a bug that hasn’t been fixed.

Generally, you can’t expect step by step instructions in these forums, only hints as to where to look.

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Error chan_sip.c:4213 __sip_autodestruct: Autodestruct

Global Settings:

Network QoS Settings:

Network Settings:

Global Signalling Settings:

Default Settings:

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