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Yes you were right about that, there were too many registration for that extension and after that it stopped working, I am wondering that why flashphoner SIP Pone demo page is not unregistered on Disconnect Botton where as it shows on the screen that it is disconnected... and on each connect it...

The WCS is causing the SIP user extensions to a locked situation, on abrupt closure of sip-phone webpage (from browser top cross button)... after that the user extension gets locked and unable to receive further calls, but the status shows as registered and out calls can also be done...

We would appreciate if you can inform us how to deregister that user from opensips

after the issue occurs ("extension gets locked") and when the user tries again to register - the status is REGISTERED again, but unable to receive calls, however it can make calls to other registered users... If the same extension is used with xLite softphone it has the same behaviour, not...

Hi there While testing the SIP phone demo we have noticed that the user extension gets locked and not unregistered on some situations.. after that we are unable to receive inbound call on this extension from other sip phones only outcall is working.. while we can register and connect as other...

How Can these calls be recorded? Thnx

Thanks

Hi there, We have successfully created the environment and installed OpenSIPS and it is up and running, we are also connecting the calls from the SIP Phone Demo page locally, but we are having issue in connecting it from the internet... port 5060 and 5061 is also opened in the CentOS firewall...

Hi there We are trying to make a connectivity with SIP API, we have successfully installed opensips on WCS and its running, kindly let us know how to define the sip extension on this and how to connect the web client? Thanks AB

Thanks... its resolved

Hi there, We tried to update the (Letsencrypt) SSL certificate on the server, after doing that we are unable to access the WCS main webpage on 8444 port neither from LAN nor from domain name ... kindly assist Thanks for assistance AB

Hi there We are having an issue in video chat application when 2 users are connected between WAN to LAN. We have a WCS server connected on global IP using router port forwading. Other users are sitting on LAN side. When user A connects from WAN side and user B connects from LAN side and starts...

We also need to reduce the mixer recorded file size... what is lowest possible setting for this? Thnx

Thanks Any entry is required in the flashphoner.properties file? AB

Hi, We have observed that in video chat if the caller is joining from mobile phone connection the picture gets stuck probably due to the low bandwidth, kindly assist how we can avoid this scenario.. our javascript settings for both sides are as following: conn.join({name...

we have installed the opensips-1.10.5-4.el7.x86_64 package on the same CentoOS machine, I hope this should not be an issue?

Please check the uploaded files

Sending you the credentials on the link with report file Thanks

Ok so ot means that WCS requires a registrar server to make tge user register? How about natting issue? Will it work for the browsers who are running from behind the routers?

Hi there We would like to connect peer to peer connectivity with the webphone to a sip phone and vice versa the clint sip phone is running on a LAN where WCS is running... The local sip phone is xlite and does not require any registration or user name... Kindly suggest what information we need...