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Summary:ASTERISK-28988: Asterisk is crashing when we receive incoming calls
Reporter:Raghu (Raghu)Labels:
Date Opened:2020-07-13 12:27:55Date Closed:2020-08-11 12:00:01
Priority:MinorRegression?
Status:Closed/CompleteComponents:pjproject/pjsip
Versions:17.5.1 Frequency of
Occurrence
Related
Issues:
Environment:CentOS Linux release 8.2.2004 (Core)Attachments:( 0) Asterisk.txt
Description:We migrated from asterisk 11.2 to asterisk 17.5.1, I was moving the sip trunk to pjsip (with flowroute) but wanting to keep the endpoints on sip. I can register the trunk and make outbound calls, however when someone is trying to call our DID's, it is not working and the asterisk service is restarted when we are getting inbound calls.
Comments:By: Asterisk Team (asteriskteam) 2020-07-13 12:27:56.363-0500

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
By: Asterisk Team (asteriskteam) 2020-07-13 12:27:57.462-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
By: Asterisk Team (asteriskteam) 2020-07-13 12:30:49.159-0500

This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable.
By: Benjamin Keith Ford (bford) 2020-07-13 13:03:04.944-0500

A couple of questions:
# Are you saying your endpoints are chan_sip, and your trunk is chan_pjsip?
# What do you mean the Asterisk service "restarts"? Is it crashing? If so, can you get us a backtrace?
By: Raghu (Raghu) 2020-07-13 13:12:45.880-0500

Hi Benjamin,

1. yes, the endpoints are on chan_sip and the trunk is configured on chan_pjsip. CHAN_SIP is on port 5060 and PJSIP on 5080. Calls come in from the carrier on PJSIP and then routed to the endpoints via CHAN_SIP.
2. Yes, asterisk is crashing when we are receiving incoming calls. I have attached log file.
By: Benjamin Keith Ford (bford) 2020-07-13 13:16:59.122-0500

The log file doesn't have any information in it regarding the crash - we will need a backtrace in order to determine what's causing the crash. The link [1] below will guide you on how to get one.

If you have a more complete log file (full.txt), that may also be useful too, but the most important thing is the backtrace.

[1]: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
By: Raghu (Raghu) 2020-07-27 14:35:41.818-0500

Hi Benjamin, when I was looking into the errors "Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)" is the common message that I can see
By: Joshua C. Colp (jcolp) 2020-07-27 15:02:11.689-0500

If using chan_pjsip then dialplan also has to be updated to use PJSIP instead. That alone would not cause a crash or restart of Asterisk, though. The information Ben mentioned is what is needed.
By: Asterisk Team (asteriskteam) 2020-08-11 12:00:00.859-0500

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines