Media
5 items
Description
Click2Call SIP Dialer to make calls from any web page. Indispensable for call centers and any other business. WebRTC SIP Phone lets you make and receive calls with your headset through any SIP provider or PBX that supports WebRTC. (https://customer.sipde.net/AsteriskWebRTC) Features: ☆ Connect to any SIP provider or PBX with WebRTC support ☆ Make and receive phone calls ☆ See the call rate before you dial (https://customer.sipde.net/APICheckers) ☆ Check your SIP account balance right in the extension (https://customer.sipde.net/APICheckers) ☆ Click to call phone numbers on web pages (available soon) ☆ Get incoming-call notifications with caller details and quick Answer or Decline actions ☆ Forward calls to voicemail or another phone number when you are unavailable ☆ View incoming, outgoing, and missed calls in the call history ☆ Message Waiting Indicator (MWI) and voicemail controls ☆ Blind call transfer ☆ Attended call transfer ☆ Multi-party conferencing ☆ Send a custom HTTP notification when a call is answered, for CRM integration (https://customer.sipde.net/how-to-CRM) What’s new v0.1.10 - Final Manifest V2 release - Bug fixes and stability improvements v0.2.1 - Moved to Manifest V3 and the Chrome Side Panel - Improved microphone access checks and handling when no microphone is available - Added Answer and Decline actions to incoming-call notifications - Improved call-status and missed-call notifications - Added toolbar icons that reflect the current phone status - Improved settings opening, Side Panel recovery, and call-history updates Data and privacy - Feature availability depends on the configuration of your SIP provider or PBX. - The current version of WebRTC SIP Phone does not track browsing history or inspect the content of open web pages. - SIP account settings, including sign-in details, the last dialed number, and call history are stored locally in the extension’s Chrome storage. - When registering or setting up a call, the extension sends SIP account details, signaling data, and phone numbers to the SIP provider or PBX configured by the user. - Microphone access is used only for calls. During an active call, audio is sent over WebRTC to the media endpoint provided by the SIP provider or PBX. - If balance or call-rate checks are enabled, the data needed for the request is sent to the web-service address configured by the user. (https://customer.sipde.net/APICheckers) - If CRM integration is enabled, the configured HTTP notification and the call-event data included in it are sent to the address configured by the user. (https://customer.sipde.net/how-to-CRM) - Connections to additional servers are made only for SIP services, balance or rate checks, and CRM integrations configured by the user. - Click-to-call on web pages is planned but is not available yet.