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Thanks

OK ... by setting it to sip_add_contact_id = false solved this issue.. Can you kindly assist in how can we restrict WCS to use a fixed SIP port, currently it is using dynamic SIP ports from 30001 to onwards and on every restart it changes it... we have tried to apply the following setting but...

If above configuration is not possible then how can I know the additional number that WCS is adding via the API sip:108-102877877 ?

Thanks, the update is now working for Disconnection, but I have noticed that if I login as 108 the WCS registration is adding some number after the 108 and making it as sip:108-102877877@192.168.0.223:30004 I want to get register only as sip:108@192.168.0.223:30004 how can I configure this...

Or use the same registration of last connect, purpose is not to get locked after sometimes since the users will be logging in and in several times. thnx

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...