Max's latest activity

Good day. We checked the report. If you're using a custom AWS image, please use Amazon Linux 2023 or Ubuntu 22.04 or 24.04 as base OS...

WCS does not refer to any SIP server configs There was probably an actie SIP session between WCS and Asterisk. By default, session is...

Good day. Please make sure UDP ports are available for both sides: Port routing checking If all the ports group are available, and the...

Good day. We fixed a new builds uploading, now both WCS and Web SDK should be available.

Good day. WCS does not support pulling SRT streams from external source. Only SRT publishing is supported. We raised the ticket WCS-4825...

На прошлой неделе проходил аудит лицензий. Возможно Ваша лицензия была помечена как expired. Пришлите пожалуйста на...

В прикрепленном логе видно, что истекла лицензия: Убедитесь, что: - сервер лицензий https://my.flashphoner.com/ доступен с вашего...

Добрый день. Сервер присылает сигналинговое сообщение getVersion с текущей версией сервера, это документировано. Однако на стороне Web...

You can change the buffer size settings: rtmp_out_buffer_enabled=true rtmp_out_buffer_polling_time=20 rtmp_out_buffer_start_size=10000...

Good day. No. The only way to add a delay still is RTMP output buffer. So you should continue with this workaround.

We fixed the issue in build 5.3.254. You should update and set client_timeout=5000 reg_expires=60 sip_force_session_expires=60 In this...

Please gather a traffic pcap dump between WCS and PBX TTS. tcpdump -s 4096 -w log.pcap Analyze this dump in Wireshark and make sure...

Good day. Please clarify the case: 1. Caller establishes audio only call from browser to the TTS 2. Caller hears audio from the TTS...

Builds downloading issue is fixed now.

We checked the server and tested a SIP call. The problem is between browser and WCS, not between WCS and 3CX: This occurs at callee...