Summary:	ASTERISK-26266: SIP Outbound Registration is not working correct
Reporter:	Antonis Psaras (apsaras)	Labels:
Date Opened:	2016-08-03 17:29:34	Date Closed:	2020-01-14 11:13:40.000-0600
Priority:	Critical	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/Registration
Versions:	11.23.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	Centos 6 64bit	Attachments:
Description:	It seems that Asterisk when acting as user to a sip provider does not respect the expiration proposed by sip provider. Moreover, even if expiration offered is within the ranges of the provider, instead of using the registration specific expiration configured at the registration string with ~expiry, is using the defaultexpiry general sip configuration. To re-produce it, create a registration string with expiration ie 600 register => 123:123@1.1.1.1~600 and set defaultexpiry to 3600. On the sip trace you will see that Asterisk offers 600 expiration but when you run sip show registry you see that refresh is at 3600 which can be check that is true from the sip trace which has no re-registration with in 10 minutes.
Comments:	By: Asterisk Team (asteriskteam) 2016-08-03 17:29:34.537-0500 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: Joshua C. Colp (jcolp) 2016-08-04 05:10:14.131-0500 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 In this case the complete SIP interaction with the remote server would be needed. By: Asterisk Team (asteriskteam) 2016-08-18 12:00:01.667-0500 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