WebRTC SIP Phone with Click2Dial

WebRTC SIP Phone is extension for send and receive calls using headset and work with any SIP operator who support WebRTC technology. (https://customer.sipde.net/AsteriskWebRTC)
Features:
☆ You can connect to any service provider
☆ Making and receiving phone calls
☆ Shown call rate before making (https://customer.sipde.net/APICheckers)
☆ Shown own account balance directly in Extension (https://customer.sipde.net/APICheckers)
☆ Click to call phone numbers on any web page (available soon)
☆ Receive alerts for incoming phone calls: know instantly who is calling, answer or decline the call
☆ Forward your phone calls to your voicemail or any phone number when you are unavailable or out of place
☆ Call History
☆ Blind Transfer
☆ Attended Transfer
☆ Multi-Party Conference
Updates:
v0.1.4
- MWI and VoiceMail control added
- Initiating of SIP calls is fast now
v0.1.5
- Web services added (custom balance checker and custom rate checker)
- Sending custom http message on answer (https://customer.sipde.net/how-to-CRM)
- Attended transfer works now
v0.1.6
- More convenient and useful settings with prompts
v0.1.7
- Little tweaks
v0.1.8
- Page of feedback and statistics collection added
- Protection from endless registration, so as not to get banned
v0.1.9
- Bug fixed in displaying the answer button on an incoming call
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NOTES
* The notice you see about having access to your history and website data is automatically generated because WebRTC SIP Phone runs on every tab. But it does NOT actually monitor your browsing history or require your personal information to work properly.