| Summary: | ASTERISK-26370: bridge_softmix: softmix_bridge_write_voice doesn't handle dsp_talking_threshold correctly | ||
| Reporter: | Chris Cox (chris_and_jolene@hotmail.com) | Labels: | |
| Date Opened: | 2016-09-13 18:55:23 | Date Closed: | 2020-01-14 11:13:37.000-0600 |
| Priority: | Trivial | Regression? | |
| Status: | Closed/Complete | Components: | Bridges/bridge_softmix |
| Versions: | 13.9.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Any | Attachments: | |
| Description: | The implementation currently (13.9.1) bases the return to talking state on whether or not the time being silent is less than the silence_threshold, meaning that changes to the dsp_talking_threshold in confbridge.conf have no effect here and the conference talking event gets sent too soon. I'm looking at using these talk events to send a special push-to-talk signal (voxing ptt) to a transmitter, so I am adjusting the talking threshold to try to prevent this voxing-ptt from being too sensitive (Is there a talking energy threshold somewhere that I can adjust as well?) | ||
| Comments: | By: Asterisk Team (asteriskteam) 2016-09-13 18:55:23.730-0500 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: Chris Cox (chris_and_jolene@hotmail.com) 2016-09-14 18:14:31.015-0500 I forgot to mention how impressed I am by Asterisk. It is a great product and I am amazed by all it can do, and so easily from the user's perspective. Comparing it to some other software pbxs is like night and day. By: Rusty Newton (rnewton) 2016-09-15 17:44:34.818-0500 First, thanks for the compliment on the project. I'll relay that to the core team! :) Second, I think you want to run this request/question by the development list. We don't get all the eyeballs here on JIRA, that is not everyone monitors it. http://lists.digium.com/mailman/listinfo/asterisk-dev I'm not sure this a bug, but I can't answer this question specifically and your question is best served by getting it in front of all the developers before we run it through triage. The one or two I quickly asked were not sure about it. I'll leave the issue in Waiting on Feedback for a couple of weeks so you can come back with a response from the feedback you get on the list. Thanks! By: Asterisk Team (asteriskteam) 2016-09-30 12:00:00.663-0500 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 | ||