Summary: | ASTERISK-30412: chan_unistim: RTP Bleedover | ||
Reporter: | John Panzer (jpanzer) | Labels: | |
Date Opened: | 2023-01-31 21:35:10.000-0600 | Date Closed: | 2023-02-16 12:00:18.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_unistim |
Versions: | 16.28.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | VM running on Debian 10 "Buster" | Attachments: | |
Description: | I am experiencing an issue with multiple lines on the Nortel i2004 and i2002 handsets where RTP is bleeding through on held lines when using multiple extensions. When holding a call from the Nortel phone, the RTP audio will bleed through to the second channel on the Nortel handset.
For example - I place a call on hold on the Nortel handset. If I then pickup the second channel, I can still hear the remote end of the original call. However, they hear my MOH channel. If I start a second call, I still hear the audio from the original call and don't hear any audio from the second call until I release the first call. It is as if the RTP channel is not being released or paused correctly and I am still hearing the original RTP channel. | ||
Comments: | By: Asterisk Team (asteriskteam) 2023-01-31 21:35:14.353-0600 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Igor Goncharovsky (igorg) 2023-02-01 12:12:28.744-0600 Please attach output of CLI commands "unistim show devices" and "unistim show info" for the phones used to reproduce issue. Also please provide CLI output while issue reproduced with unistim debug enabled - "unistim set debug". By: Asterisk Team (asteriskteam) 2023-02-16 12:00:17.915-0600 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 |