Summary:	ASTERISK-28386: res_pjsip does not follow DNS SRV priority values
Reporter:	Andy Chi (achi)	Labels:	pjsip
Date Opened:	2019-04-17 20:35:17	Date Closed:	2019-05-02 12:00:02
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Resources/res_pjsip
Versions:	13.25.0 13.26.0 16.3.0	Frequency of Occurrence	Frequent
Related Issues:
Environment:	Sangoma Linux release 7.5.1805 (Core)	Attachments:
Description:	Using a provider that supports DNS SRV with the following output: _sip._udp.testing.st.sip.global. 299 IN   SRV     20 100 6000 dllstx37-access-sbc2.sip.global. _sip._udp.testing.st.sip.global. 299 IN   SRV     10 100 6000 plmomn-access-sbc2.sip.global. My expectation would be for registration and calls to always use the priority 10 server, however when running a simple channel originate script, I see calls round robin between the two entries.
Comments:	By: Asterisk Team (asteriskteam) 2019-04-17 20:35:18.250-0500 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]. 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: Joshua C. Colp (jcolp) 2019-04-18 04:24:59.359-0500 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 particular if under Asterisk 16 you could enable debug logging and do tests, that would show what the resolver is doing. In the case of 13 we have no control over it or can see debug, as PJSIP itself does the resolution so getting data is difficult. By: Asterisk Team (asteriskteam) 2019-05-02 12:00:01.453-0500 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