|Summary:||ASTERISK-19891: Realtime queue problem with joinempty option|
|Reporter:||Michał Schielmann (schielmann)||Labels:|
|Date Opened:||2012-05-21 09:11:43||Date Closed:||2012-06-29 17:58:42|
|Description:||When a client joins a queue without any members and after a while a member does login to the queue - by adding a row in Database - Asterisk would not create a connection to the new member. This situation lasts until another client joins the queue or until the queue show 'queuename' command. Those two events refresh the state of queue members and in effect a connection to the queue member is created. |
This problem may cause a situation in which a client would wait for a really long time, thinking that he is first in the queue while there is active, free member waiting for a call.
|Comments:||By: Rusty Newton (rnewton) 2012-05-23 17:26:55.119-0500|
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. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. 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
3. the behavior you actually encountered (in as much detail as possible).
This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks!
Please be sure to include the Asterisk version this was found on, any additional versions tested against, OS distribution, kernel version, architecture, a full log with at least DEBUG 5 covering the time in which this issue happened and if database connectivity is involved any application configuration files for the database, i.e. res_odbc.conf, etc. Please sanitize any private information.
By: Rusty Newton (rnewton) 2012-06-29 17:58:53.758-0500
Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines