Summary: | ASTERISK-27815: REST API - channel_ids are getting stuck | ||
Reporter: | Bruno Duarte (bruno) | Labels: | pjsip |
Date Opened: | 2018-04-19 08:47:18 | Date Closed: | 2020-01-14 11:13:37.000-0600 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | Resources/res_ari_channels |
Versions: | 13.19.0 | Frequency of Occurrence | Frequent |
Related Issues: | |||
Environment: | Production | Attachments: | |
Description: | Hi,
We have implemented the REST API for our application, to monitor the sip extensions. Always when the extension originates or receives a call, it will be attributed to it a channel number: { "technology": "SIP", "resource": "0300", "state": "online", "channel_ids": [ "1968058526.14522", ] } When the call is finished, the asterisk API have to clear the channel id from the sip extension: { "technology": "SIP", "resource": "0300", "state": "online", "channel_ids": [ ] } But for some of my extensions, Asterisk is no more cleaning the channel_ids field, and the extension is even accumulating these channel numbers: { "technology": "SIP", "resource": "0301", "state": "online", "channel_ids": [ "1524058526.10923", "1523987775.8614", "1524082882.12616", "1523987766.8605" ] } Please, let me know could be done with this issue. Regards, Bruno Duarte. | ||
Comments: | By: Asterisk Team (asteriskteam) 2018-04-19 08:47:19.671-0500 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. 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]. By: Bruno Duarte (bruno) 2018-04-19 10:01:01.619-0500 As additional information, There is no call stuck on asterisk, and another thing that I realized, If I try to check the details of one of these stuck channels, it seems like it doesn't exist. http://localhost:8010/ari/channels/1524058526.10923 { "message": "Channel not found" } Obs: the port is 8010 instead of 8088 due to a ssh tunel for the application. By: Kevin Harwell (kharwell) 2018-04-19 15:27:14.623-0500 We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis. We also need protocol specific debug captured at the time of the issue. Please include the following: * Asterisk log files generated using the instructions on the Asterisk wiki [1], with the appropriate protocol debug options enabled, e.g. 'pjsip set logger on' if the issue involves the chan_pjsip channel driver. * Configuration information for the relevant channel driver, e.g. pjsip.conf. * A wireshark compatible packet capture, captured at the same time as the Asterisk log output. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Asterisk Team (asteriskteam) 2018-05-04 12:00:01.521-0500 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 |