| Summary: | ASTERISK-27918: Transfer in U&D mode | ||
| Reporter: | Maxim Trofimov (maxws) | Labels: | pjsip |
| Date Opened: | 2018-06-14 07:07:25 | Date Closed: | 2020-01-14 11:13:28.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Core/Bridging |
| Versions: | 13.18.5 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | asterisk - 13.18.5 freepbx - 14.0.1.36 FreepbxDistro | Attachments: | |
| Description: | Good afternoon.
When transferring a call, the communication channel with the first user called is not completed and as a result the system does not treat this call as a transfer (there is no event in the queue_log ATTENDEDTRANSFER only COMPLITECALLER and it is created when the call with the user to whom the transfer was terminated ends). Transfer is made through *2 or ##. User&Device mode. I'm calling from number 6118 (external trunk for PBX) to the queue number 4777 the operator registered on the device 4920 at number 4987 (TEST) picks up the phone and transfers (* 2) to number 4988 talks and hangs up, call transfer number 6118 communicates with 4988 and ends the conversation. However, in the queues of the queue, this transfer is not processed as a transfer. Logs reflect that the conversation was between 6118 and 4987 and the duration of the conversation is equal to the total duration from when the 4987 accepted the call before the end of the conversation with 4988 1520274014 | 1520274013.270913 | 4777 | NONE | DID | 4777 1520274014 | 1520274013.270913 | 4777 | NONE | ENTERQUEUE || 6118 | 1 1520274021 | 1520274013.270913 | 4777 | TEST | CONNECT | 7 | 1520274014.270914 | 6 1520274061 | 720274013.270913 | 4777 | TEST | COMPLETEAGENT | 7 | 40 | 1 LogFile - http://pastebin.freepbx.org/view/0a163b01 | ||
| Comments: | By: Asterisk Team (asteriskteam) 2018-06-14 07:07:26.241-0500 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: George Joseph (gjoseph) 2018-06-14 09:04:03.098-0500 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., 'pjsip set logger on' if the issue involves chan_pjsip, 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: Asterisk Team (asteriskteam) 2018-06-28 12:00:01.814-0500 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 | ||