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How do I set up sipgate Webphone?

Set up sipgate Webphone and make calls directly in the browser – without additional software.

With the webphone you can make calls directly in the browser — completely without additional software. For good voice quality, we recommend a headset. Alternatively, you can sipgate app use.

Supported browsers

  • Brave from v1.8.95

  • Chrome from v75

  • Edge from v79 (Chromium, Windows 10)

  • Safari: yes

  • Internet Explorer 11: not supported

Start webphone

  1. Go in the user interface to My telephony.

  2. Click at bottom left on Webphone.

  3. Your selected phone connects automatically.

Note: The button only appears if you have set up a VoIP phone in the account. Admins do not see the button when they switch to the view of another user.

Phone numbers The webphone shows the phone number for incoming and outgoing calls. If multiple phones are set up, you can select the desired number here.

Test function

  • Call the 10005 for free to check voice quality.

  • If you have problems: check audio input and output in the system settings of your computer.

Important functions

  • Calls: Enter or paste phone numbers (spaces, +, -, () are allowed). Unknown numbers appear as Unknown, until they are saved in the contacts.

  • Call forwarding (transfer): *

    • Keyboard shortcut: ∗3 + phone number + # → connection is established.

    • Menu: Transfer click, select phone number/contact → confirm.

    • Cancel: enter ∗8 → wait for announcement.

  • Notifications: Under Settings → push notifications enable. With multiple monitors, notifications appear only on monitor 1.

  • Audio settings: Adjust the volume in the operating system or on the headset. In Chrome, Brave and Opera, you can choose a separate device for the ringtone.

Restrictions

  • No three-way conferences (use a VoIP app or a VoIP phone instead).

  • Click2Dial is prepared, but not yet available.

Security and technology

  • Voice codecs: G.711 (60–100 kbit/s), G.722 (160–200 kbit/s).

  • Encryption: signaling via TLS, voice via SRTP.

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