ASTERISK-23676: SIP Channel driver emits autodestruct WARNING with owner in place: channel reference leak

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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.