| Summary: | ASTERISK-29074: SIP-Reregister timeout based on absolute time | ||
| Reporter: | Marcel Dischinger (marceldischinger) | Labels: | |
| Date Opened: | 2020-09-10 02:39:31 | Date Closed: | |
| Priority: | Minor | Regression? | Yes |
| Status: | Open/New | Components: | Channels/chan_sip/Registration |
| Versions: | 16.15.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | The timeout for re-registrations at a VoIP-Server seem to be using absolute time instead of relative time.
This leads to the following behavior: # Device boots, time is t (taken from RTC). # Asterisk registers SIP with server # NTP updates time to a past time (e.g., (t-2h)) or time is corrected manually by the user # Asterisk sends no re-registers until the time is (t+20s). As a result, the registration times out at the server, while asterisk claims it is still registered. While I cannot check, it is likely that other Asterisk versions are affected as well. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2020-09-10 02:39:33.602-0500 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: Benjamin Keith Ford (bford) 2020-09-10 09:55:15.764-0500 The chan_sip channel driver is in 'extended' support status and is supported only by community members. Your issue is in the queue. Your patience is appreciated as a community developer may work the issue when time and resources become available. Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1] If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way. [1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process [2]: http://www.asterisk.org/community/discuss [3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties | ||