Search results
Results from the WOW.Com Content Network
Hi All, I wanted to share my webphone that I'm working on using a modified version of ctxSip. It includes an updated version (0.15.6) of Sipjs which...
I have made a customer WebRTC phone using Browser-Phone for FusionPBX where you just need to assign a user to the extension and click on the webphone which will be available on fusionpbx menu. You won't need to register/enter any information about the extension, the system will fetch everything and register automatically.
Hi all. Just testing this now. The webphone loads but is not able to send/receive calls. Not seeing anything in the CLI. I have an SSL cert and TLS is enabled/working. Under SIP status wss is showing as being on the private side IP not public. Is there a variable to ad such as ext_sip_ip to...
About us. Building a community of users to advance their knowledge and understanding of voip through sharing, learning and supporting each other.
Dear Amit I follow your github artical and everything is working great . I am testing call center feature using webrtc i create two users name (agent1 and agent2) and assign extension but they are part of agent group i also allow agent group in the phone option Menu manager .
I am facing one serious issue with the Fusion pbx webphone. When we are disconnecting a call from the webphone the call leg is not been cleared. When we check on active calls it shows active. Most of the time on the Web phone, this issue was happening. Can you help me to fix this issue? The softphone is working fine.
In this tutorial we are going to enable WebRTC on FusionPBX to use with an external webphone, in my case i use Saraphone. 1. First step login on your FusionPBX server and go to Menu->Advanced->Sip Profiles. 2. Click on "internal", then modify this. liberal-dtmf true true. send-message-query-on-register true true. send-presence-on-register true ...
1) Auto login with embedded webrtc webphone when signed in. 2) Agents able to login/logout of Queues. 3) Click-to-call for recent calls & abandoned calls widgets. 4) call center queues dashboards with supervisory features (if perm is supervisor) 5) single "workspace" view for agents and supervisor. Borrowed works from -
Good Morning, Short Story: I want to connect MS Teams client like most other SIP Softphones to our FusionPBX system; literally the settings involved are domain, extension, creds, FQDN , port# and a few others.
Hi, I am having problem when trying to use video over SRTP. I keep getting SRTP_READ_ERROR in freeswitch. I have tried linphone (PC, and mobile), Jami, Fanvil phone X7A and they all have the same issue. I am using the latest Freeswitch 1.10.6. I have read about something like this in...