| Summary: | ASTERISK-25913: firends and peers cannot connect | ||
| Reporter: | Bill Neely (ceo_xantek) | Labels: | |
| Date Opened: | 2016-04-11 13:49:03 | Date Closed: | 2016-04-19 17:12:16 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | |
| Versions: | 13.6.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | OS= centos 7 running on cloud server at digitalocean NYC1 | Attachments: | |
| Description: | A few minutes after starting, Asterisk reports high ping times from qualify requests, and then sets all accounts to UNREACHABLE. We ran ping tests using sipping.py and found ping times to be normal. Restarting * reset the ping times to normal, but again. after a few minutes, ping times grew to UNREACHABLE. The effect was that phones and peers were unable to connect. In order to correct the problem, we had to downgrade to 13.2.0. Problem has not repeated since downgrade (now several days).
We are not sure whether this is a bug or some security hole. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2016-04-11 13:49:03.942-0500 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: Richard Mudgett (rmudgett) 2016-04-11 14:25:58.614-0500 You have not specified which SIP channel driver you are using. There was a regression that has since been fixed after v13.6 dealing with the scheduler. Why didn't you try v13.8 before returning to v13.2? By: Rusty Newton (rnewton) 2016-04-11 20:10:01.402-0500 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. The specific steps or actions you took that caused you to encounter the problem. 2. The behavior you expected and the location of documentation that led you to that expectation. 3. The behavior you actually encountered. To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines [2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Bill Neely (ceo_xantek) 2016-04-19 16:39:39.295-0500 This issue appears to have been resolved in 13.8 | ||