| Summary: | ASTERISK-26965: Caller ID set to 'asterisk' during atxfer from bridged queue | ||
| Reporter: | David Wakelin (MacroMan) | Labels: | |
| Date Opened: | 2017-04-25 05:26:58 | Date Closed: | 2017-06-07 03:33:36 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Applications/app_queue Features |
| Versions: | 13.16.0 14.3.0 14.5.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Ubuntu 16.04 LTS Asterisk 14.3.0 | Attachments: | ( 0) extentions.conf.txt ( 1) features.conf.txt ( 2) queues.conf.txt ( 3) sip.conf.txt |
| Description: | Scenario:
A customer calls into a queue. Agent A's phone rings and can see the caller ID fine. Agent A picks up the call. Agent A starts an attended transfer to agent B using atxfer in features.conf Agent B's phone rings, but the caller ID is set to 'asterisk'. At this point I would expect the caller ID to be set to the phone number of agent A. I am using a Websocket based SIP phone, but the same problem occurs when using other SIP devices. Settings to recreate the issue: sip.conf: https://pastebin.com/LnD7Vepe extentions.conf: https://pastebin.com/wyhxBi0J queues.conf: https://pastebin.com/TQai9dKD features.conf: https://pastebin.com/2WLCZFfp | ||
| Comments: | By: Asterisk Team (asteriskteam) 2017-04-25 05:26:58.790-0500 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: Rusty Newton (rnewton) 2017-04-26 15:15:54.783-0500 David, Thanks for reporting this and providing the necessary configurations to reproduce the problem. As mentioned in our issue tracker guidelines[1], can you attach the configs or debug to the issue as txt extension files? See (More > Attach files) [1]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: David Wakelin (MacroMan) 2017-04-27 03:40:35.726-0500 Conf files to reproduce problem. By: David Wakelin (MacroMan) 2017-04-27 03:40:52.057-0500 Sorry for not attaching my config as txt files. Please find them attached. By: Benjamin Keith Ford (bford) 2017-06-06 11:21:03.253-0500 [~MacroMan], I noticed that in your sip.conf, you don't specify callerid for any of the sip users. Try adding callerid for each of the users and see if that fixes the problem. This worked for me. If it does not fix your problem, leave a comment with the results. By: David Wakelin (MacroMan) 2017-06-07 03:33:36.447-0500 Fixed the issue. Didn't see the comments in the sample spi.conf about the default being 'asterisk' | ||