WebRTC SIP VoIP Phone — Click-to-call & Conference Calls

Click2Call lets you dial from web and CRM phone links. Your webphone is an online phone with softphone controls and call history.

WebRTC SIP VoIP Phone turns Chrome into a WebRTC phone for everyday business calls. Make and receive calls through your WebRTC-compatible SIP provider or PBX, and bring callers together in local conferences. Keep your phone within reach in the Chrome Side Panel. (https://customer.sipde.net/AsteriskWebRTC) FEATURES ☆ Make and receive calls through your WebRTC-compatible SIP provider or PBX in the Chrome Side Panel, with caller details, status, and timers in view. ☆ Switch between call rooms L1, L2, L3, and beyond: hold one conversation, continue another, and answer new callers. ☆ Host local conferences without a PBX room: add outgoing callers before they answer, step out and return, and hold, disconnect, or reconnect participants. ☆ Tap a phone link on a website or CRM page with optional Click2Dial: the panel fills in the number, and you choose when to call. ☆ Reach favorite numbers in one tap with four speed dial buttons, each with its own name and number. ☆ Answer or reject from notifications with caller details; answer or hang up with a compatible headset’s Play/Pause button. Choose when the panel comes forward. ☆ Mute your microphone, use the keypad, or make blind and attended transfers from the main number field. ☆ Choose the microphone and audio output device that suit your setup. ☆ Forward calls to voicemail or another number when unavailable; check the Message Waiting Indicator (MWI) and use voicemail controls. ☆ Find incoming, outgoing, and missed calls in your history. New entries show talk time and flag unanswered or rejected outgoing calls. ☆ Check the call rate before dialing and your SIP balance in the extension. (https://customer.sipde.net/APICheckers) ☆ Match the phone and settings to a light, dark, or browser-controlled theme. ☆ Connect to your SIP setup through WebSocket addresses with ports, paths, queries, and IPv6. ☆ Choose automatic ICE or tune STUN/TURN yourself, check servers, and restore missing defaults without changing your entries. ☆ Send a custom HTTP notification when an incoming call is answered to connect with your CRM. (https://customer.sipde.net/how-to-CRM) Some calling features depend on your SIP provider or PBX configuration. Multiple calls and local conferencing require your SIP account to support multiple simultaneous calls. Headset button support depends on your device, operating system, and browser. WHAT’S NEW v0.2.7 - Check talk time and spot unanswered or rejected outgoing calls in new history entries. v0.2.6 - Take more than two calls with rooms L3 and beyond. Start a new call or answer an incoming one while another call is connecting. - Put a dialing call on Hold: the hold takes effect after connection, or press again before then to cancel. - Keep a conference in its room when adding or reconnecting participants. - Scroll the phone panel when the Side Panel has insufficient height. v0.2.5 - Connect through custom WebSocket addresses with ports, paths, queries, and IPv6. - Match the phone and settings to a light, dark, or browser-controlled theme. New installs follow the browser, and existing choices stay intact. - Leave a conference without dropping callers. A lone participant is held automatically, and you can keep the reconnect list for 0–3600 seconds (60 by default). - Hear steady conference speech, with ringing and call-progress tones kept quieter. - Call people back by their correct numbers, even when their display names appear in history or the conference list. - Restore missing default STUN servers without changing existing entries or their enabled states. - Keep account and system settings intact after updates and re-enabling. NOTES - When you enable Click-to-dial, Chrome asks for access to website data so the extension can handle phone links. The extension does not track your browsing history or collect webpage content.