| Summary: | ASTERISK-25345: pjsip:0 <?>: tsx0xb3fe5d1c ...Failed to send Request msg INVITE/cseq=28532 (tdta0xb6bb49c0)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT)) | ||||
| Reporter: | Marek Cervenka (cervajs) | Labels: | |||
| Date Opened: | 2015-08-26 04:01:02 | Date Closed: | 2015-08-26 05:33:26 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | |||
| Versions: | 13.5.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | centos6,pjproject 2.4.5, simpl5 webrtc client, chrome 44 | Attachments: | ( 0) console.log ( 1) pjsip.conf.txt | ||
| Description: | call from webrtc to webrtc fail with Unsupported transport (PJSIP_EUNSUPTRANSPORT)
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| Comments: | By: Asterisk Team (asteriskteam) 2015-08-26 04:01:05.068-0500 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) 2015-08-26 04:59:17.572-0500 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 Also please do not sanitize or trim your log. The small amount you have provided shows that it failed, but not the steps leading up to it. By: Marek Cervenka (cervajs) 2015-08-26 05:28:29.384-0500 please close this issue it's duplicate to https://issues.asterisk.org/jira/browse/ASTERISK-24602 my apologies | ||||