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Thank you. I'll update and try it out.

Thank you for reply. To explain in more detail, 1. When a caller makes a call, an Bot answers the call. 2. The Bot analyzes the inquiry and responds in a human-like manner. The problem is that the Bot's voice can sometimes be audible to the caller, but sometimes it can be distorted. Typical...

Hello. I'm developing a web-based SIP service. ----Version info----- wcs_version=5.3.243-b287c9cf9f9fb400f97398296810c5ca214b5199 wcs_client_version=2.0.269-7ca52229c68b6b769e3e67e29c7e4cf887237822 When connecting via WCS in a web browser, the audio tempo fluctuates between faster and slower...

Thank you Max After updating to 5.3.243, the 8444 website does not loading. I installed jdk17 version. (JDK 21 didn't work either.) This is the startup.log. Is it a permission issue? I did run set-permissions. Loading now. It happened because wss.jks was missing after the new installation...

Hello? I am configuring SIP functionality on an Amazon public IP. I installed Asterisk 22.7 (pjsip) on another server to connect between public IPs, but it is failing. (Attached logs) -----Version info----- wcs_version=5.2.2309-be82b7474f2bb543491f92db3c7b2ab5bca9d49d...

Thank you Max. I have a question. Can you tell me when it will be released? I'm preparing a service demo using this feature. I was planning to use it next week—when will the updated version be available?

Thank you for your reply. As I mentioned in my initial post, I tested reg_expires and sip_force_session_expires before asking the question, as they seemed related. I also confirmed that the expired= value in the SIP header was changing. However, it still remained in the contacts list. I...

Thank you for your response. I understand the SIP registration settings during the createSession process. Our service must unregister all incoming calls after using them once SIP registration is established. Specifically, we want to remove calls from the list shown by “pjsip show contacts”...

Hello. SIP disconnect was performed, but SIP unregistration failed. I integrated with Asterisk following this guide. SIP call connections work fine. : https://docs.flashphoner.com/static/WCS53/WebRTC-SIP_gateway_functions/SIP_calls_in_a_WebRTC-compatible_browser/ ----- MY WCS, Version...

Thank you for your reply. Referring to the article, I updated to version 5.3.165, but I'm still experiencing the same issue (486 busy). I didn't install a separate certificate for Asterisk. Could this be related? I've attached the logs for WCS and Asterisk. Is there anything else you'd like to...

Hello. I'm testing SIP functionality, and when I make a call, it returns "Busy." - https://docs.flashphoner.com/display/WCS52EN/A+call+between+two+browsers+made+via+the+SIP+server - The "Without an external SIP server. SIP and RTP media are processed by WCS" section in the documentation works...

oh, I misunderstood it as possible in my previous reply. We plan to migrate the service developed with roomApp to the sfuApp platform. We'll try using the bridge option (not recommended for production) and wait for support on ticket 4591. Looking forward to good news. Thank you.

I have another question. I'm trying to use Mixer, but I can't find the stream name without the bridge option. When I use the SFU client, outgoing and ingoing information is collected in /sfu/stats, but the stream information required for mixing is missing. I'm looking for a way to go live (RTMP)...

Oh, that's right. Another WCS server also doesn't have a /usr/local/FlashphonerWebCallServer/conf/zclient folder. Instead, it has a /usr/local/FlashphonerWebCallServer/conf/conf folder structure (conf 2) and zclient*.yml files under it.

More good news. After running ./uninstall.sh, I reinstalled version 5.2.2301, and the sfu client worked fine even in the 5.2 version. Thanks.

No. The test server was first installed on AWS Marketplace (EC2) in 2022 and has been updated since then. The current version is 5.2.2301. I added the sfu app to this server, and during sfu client testing, a server error was logged. I applied two methods to this server: 1. Installed 5.3.126 -...

Good news! I installed the new WCS 5.3.126 version using the link below, and the sfu client now works properly. : https://docs.flashphoner.com/static/WCS53/WCS_update_to_5.3/#updating_52_to_53_or_one_53_build_to_another Then, I reinstalled the previous version(5.2.2301), and the problem...

Thank you for your reply. 1. It works fine on demo wcs. 2. Even after updating my wcs to 5.2.2301(First installed in 2022), the sfu client does not work. Similarly, the server log shows a web socket error log. 3. The newly installed WCS has not yet been tested. It would be helpful if you could...

Hello. I have a question while testing sfu functions with simulcast. : https://docs.flashphoner.com/display/WCS52EN/SFU+functions+with+Simulcast When I open the sfu client sample page, the cam is added, but message send fails (datachannel). Other users cannot join either. Looking at the server...

Thanks for your reply. Additionally, how do I do a YouTube Live (Re-Publish) in this situation? I need to include both the selected video and audio, but I'm not sure how to do that. Should I re-publish after mixing the stream? Thank you.