Recent content by Max

We can't reproduce the issue with build 5.2.2192 using your configs and supposed test case. So you should do the following: 1. Enable server logs INFO level as mentioned above 2. Enable RTSP port manager DEBUG logs as mentioned above 3. Let the server work for a few hours 4. Collect a report as...

Yes, if a stream is mixed at Asterisk side, WCS receives the mixed audio. So it should be solved at Asterisk side.

As mentioned above, we can't reproduce the issue in build 5.2.1921 on our staging server with your settings. Please update to 5.2.1921 and test recording again.

Just send a few hours of logs. We have to estimate state of RtspPortManager Also please temporary switch logs to info level if it is possible. log4j.rootLogger=info, stdout, fAppender It will give us more details.

Hello We assume that it is a kind of leak in RTSP port manager. Unfortunately RTSP ports are not listed in the Stats. So you have to gather logs. Please add a line into the config conf/log4j.properties It does not require server restart. log4j.logger.RtspPortManager=DEBUG The log4j.properties...

Hello. We believe that the first thing you should do is roll back — that is, downgrade to the version that was working for you. If version 5.2.1054 was working, then you should downgrade to that version. If, for some reason, the downgrade doesn’t help, then you need to restore from a backup...

When updating WCS from old version to a newer one, you should also care about WebSDK version update. For example, if you've updated WCS to 5.2.1921, you should also update WebSDK to 2.0.240, especially if you're still using 0.5.28.* builds. Please try to make a test calls via Phone Min example...

Seems like you've started two simultaneous SIP calls to the same callee namber in the same client session: So PBX declines the first one The client session was closed, and the second call was cancelled It seems like either client (web page) code issue or REST hook issue. You should not start...

We fixed the issue in build 5.2.2207.

We tested the build 5.2.1921 with your settings on our staging server. All the SIP calls are recodring successfully with the following settings rtp_force_synchronization=true rtp_activity_detecting=false,0 generate_av_for_ua=all stun_freshness_timeout=1000000000...

Seems like you're still using build 5.2.1054 on the server which we have access. Please update to 5.2.1921. You can also try to update to newer build 5.2.2183. In this build, SIP calls may be recorded to MKV container: SIP calls recording

Good day, Marc. Seems you've unpacked FlashphonerWebCallServer-5.2.1921.tar.gz archive to /usr/local folder directly. You should not do it. So you should do the following: 1. Backup the current installation 2. Uninstall WCS /usr/local/FlashphonerWebCallServer/bin/uninstall.sh 3. Download and...

This file seems to be used as background. It contains black square and silence So blank screen seems to be normal in this case between one inject stops and another inject starts. Also the background file bg.mp4 is too short, and loops frequently. Use at least 1 minute file.

You must decode the stream to apply watermark at server side. For example: 1. Set the following parameter in falshphoner.properties custom_watermark_filename=/media/13983.png 2. Play the stream with resolution set explicitly In this case the stream will be transcoded to the resolution set, and...

It was the bug. The watermark must be applied to transcoder output stream, not to the transcoder source stream. The bug was fixed, so you should change your flow: a viewers should play transcoder output stream, but not the original stream.