WARNING[12980] chan_sip.c: Autodestruct on dialog

Asterisk Asterisk SIP
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WARNING[12980] chan_sip.c: Autodestruct on dialog ‘3516a25e311be0854700ca5e2de6e356@10.0.91.32:5060’ with owner SIP/serverA9132-0000b338 in place (Method: BYE). Rescheduling destruction for 10000 ms.
Calls not patch, when we get this on Our Server

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witch version of asterisk are you using ?
Asterisk 18.18.0 / 18.X.X or something even older

also you may want to change from chan_sip to chan_pjsip as that is what is supported going forward

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asterisk 13

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It means that SOMETHING is blocking the channel. There have been issues filed for this a few times for chan_sip. Some were resolved, others weren’t.

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What should I do

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Upgrade to a supported version of Asterisk and switch to chan_pjsip.

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What it mean supported version in this context

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Asterisk 13 went end of life in 2021. Current supported major versions are 18 and 20. Dates and what things mean are on the wiki[1].

[1] Asterisk Versions - Asterisk Project - Asterisk Project Wiki

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I am using doubango sipml5 will it work with chan_pjsip

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The chan_pjsip module implements SIP and supports WebRTC.

1 Like
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For practical purposes, and in this context, “supported version” means one
which people here are likely to be able/inclined to help you with.

Antony

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you can start with upgrading to Asterisk 18 as that one still have chan_sip
and when that is working change to chan_pjsip

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Any instant solution without upgrading ?

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This message can be caused by h extensions and hangup handlers that take a long time to complete, so you should make sure that you are not doing that.

However, if it is an actual deadlock, you are not going to get helped on an obsolete version.

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I too have occasionally wondered about this problem in the past, and I wonder
whether anyone can give a definition of “a long time”?

Thanks,

Antony.

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I think it is in the initial scheduling message, but just look at the timestamp of the SIP channel being hung up and the first extension. It looks like it is 32 seconds.

However, I would have said the intent was << 1 second.

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I have upgraded to Asterisk 18.17.1
I am getting a warning
chan_sip.c:8153 sip_indicate: Don’t know how to indicate condition 36

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That would be a topology change which chan_sip doesn’t support and would have no effect on things. If you’re actually having a problem, you’d need to state what the problem is.

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What I Need to change if I want to switch from chan_sip to chan_pjsip

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https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
should help you with this.

https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships
also explains more detail about how PJSIP works.

Antony.