Web Phone

Alias Phone is a web based application that allows users to make and receive voice calls as well as text messages across different platforms, such as:

  1. Chrome on Windows 11
  2. Chrome on Android
  3. Safari on iOS


Home Assistant installations for

Home Assistant settings can be found at

On app.contacttrace.com.au:

  • A Guest or a Member alias
  • A Session ID by selecting on Citizen Timeline -> Agent Session -> Create Session
  • Go to Dynamic Alias -> Anonymous Interaction -> Alias Phone and enter your mobile number, e.g: 6149933111 in the Destination Number field.

If the account has been successfully created, a pop-up window with the password will be displayed. Please note down the password or do a screenshot.

Set Up

The web address is https://phone.quuvoo4ohcequuox.0.88.io and if this is your first time, you may get a prompt that the browser wants to use the camera and the microphone.

IMPORTANT: this permission must be allowed by selecting the Allow button.


Then select the Account menu


Enter the following details:

  1. Full Name: your name, e.g: Robin Hood
  2. SIP Username, e.g: 123456
  3. SIP Password: e.g: 123456789012
  4. Extension Number: MUST be the same as the SIP Username, e.g: 123456

The rest of the fields can be left as default. Scroll down to the bottom of the page and click on the Save button. The Reload Required pop-up window will appear, simply click on the OK button to continue.


If the registration is successful, you should see the dial pad. This will allow you to make and receive voice calls.


To be able to send and receive text messages, please click on the Add User button to add the SIP Username of the user you want to send the text to. That user MUST also add your SIP Username on their device.

In the example below:

  1. Full Name: Iphone
  2. Extension including Message Exchange MUST be selected
  3. Extension Number: the SIP Username of the remote user


Once the Add button has been clicked, the iPhone user will be shown.



To show an avatar, add a contact number or an email address, click on the settings button and select Configure Extension -> Appearance.

Upload an image, enter your details and click on the Save button.


If successfully saved. your image will be visible by you and other users who add your SIP Username.


Voice Calls


1 - Mute Microphone
2 - Call on Hold
3 - Send DTMF
4 - End Call
5 - Record Call
6 - Conference Call
7 - Call Forward
8 - Speakerphone/Headset selection
9 - Call Connection Statistics
10 - Call History

Standard Voice Calls

The Home Assistant app needs to be running (can be in the background).

To make a call, click on the call button which will display the dial pad. Enter the SIP Username of the user you want to ring and press on the ring button.

If there is a history of a voice call or a text message exchange, you can also click on the name of the user directly and click on the call button.



After dialling, there will be a Please hold, your call is being connected to most appropriate device voice prompt.

There is an option to leave a voice message to the callee without connected to the callee's device. To do this, while the call is being connected, select the send_dtmf_button button, press 1 and leave the message.

If you do not have an internet access, you can still make a call by ringing +61 499 331 111 and follow the voice prompt. There are two options, to join a Meeting or dial a SIP number.

if the caller chooses to join a meeting by calling +61 499 331 111, then an existing meeting must already be available, and the caller will need to enter the meeting number to join.


When there is an incoming call to the Alias Phone, there will be a normal incoming call to your device from one of our numbers, e.g: 0499 331 111. Once you answer this call, the call will automatically be disconnected.


This call is just a notification to let you know that there is an incoming call to the Alias Phone. Simply open the notification and select the Home Assistant Incoming Call from notification.

On iOS, long press the notification to display the Webphone, Meeting and Voicemail.


On Android, simply select the down arrow key to display the Webphone, Meeting and Voicemail.



If Webphone is selected, it means you want to take the call and it will open the Alias Phone which will enable you to accept the call. If you decide not to accept the call, simply select the Reject Call button which will prompt the caller to leave a voice message.



If meeting is selected, the caller will be transferred to the Meeting page of the callee automatically.


If voicemail is selected, the caller will be prompted to leave a voice message which is emailed to user@quuvoo4ohcequuox.0.88.io.

The voice messages are also accessible by dialling *98 from the Alias Phone and enter the first 4 digits of the SIP password.

Call Transfer

To transfer call, select the call_transfer button (which will automatically put the current call on hold) and enter the number you want to transfer to.

In the example below, the number entered is 299947 and you can either do a Blind Transfer or an Attended Transfer.

Blind Transfer

A Blind Transfer will forward the current call without having to wait for the receiving party to answer. In the example above, if the Blind Transfer is selected, number 190830 will be connected to 299947 regardless whether 299947 answers the call or not.

Attended Transfer

An Attended Transfer requires the forwarded call to be answered by the receiving party first. Once the call is answered, the person who initiates the call transfer can then connect number 190830 to number 299947 or cancel the transfer.

Conference Call

To initiate a Conference Call, select the call_transfer button (which will automatically put the current call on hold) and enter the number you want to invite to join the conference call and press the Call button.

Text Messages

To be able to send or receive messages. each user must be added in each other's device by selecting the Extension including Message Exchange.

The two ticks indicate that the message has been successfully delivered to a user called iPhone.

XMPP client

https://phone.quuvoo4ohcequuox.0.88.io:7443/inverse is a web based XMPP client that allows Alias Phone users to exchange text messages.

Messages sent externally, e.g: via SMS from a mobile network can also be received. The SMS needs to be sent to +61499331111 and the format is

Sent at 1.50pm


the first line containing the Alias Phone ID of the receiver and the second line is the message.

Below is how the message appears on the receiver's browser


xmpp:agent@phone.quuvoo4ohcequuox.0.88.io user must first be added before SMS can be received.

One Click Activation

Some Web Phones come with ONE CLICK activation feature (e.g. android, ios, windows, macos, linux), so you can supply all the parameters required in the web address, without needing to enter them by hand.

Any Server Version

mandatory description example
server= the sip server sip:phone.oztrlia.org
identity= the sip account sip:614099331111@oztralia.org
authentication= authentication number 614099331111
password= the sip password 75647563
optional description example
destination= destination phone number 61138813
display= name to display to callee robinhood

Fixed Server Version

mandatory description example
username= sip username 61499331111
password= the sip password 75647563
optional description example
destination= destination phone number 61138813

the domain is fixed e.g. vahfoom3iquahfah.infinitedisk.com
the display=, identity=, authentication= are created from username=

Open Source Web Phone

All Personal Console comes with a built in Web Phone based on WebRTC technology that can be used in most browsers on all platforms (Android, iOS, Linux, macOS, Windows), WITHOUT installing of any applications.

All features work on Chrome (and associated browsers like Edge and Opera) while most features work on Safari and Firefox.

1.1 Asterisk

Currently the built in Web Phone is based on Browser Phone frontend and Asterisk backend. Browser Phone can handle video and text as well as audio:

  1. Audio Calling
  2. Video Calling
  3. Blind Call Transfer
  4. Attended Call Transfer
  5. 3rd Party Conference Call
  6. Call Detail Records
  7. Audio & Video Call Recording
  8. Screen Share during Video Call
  9. Scratchpad Share during Video Call
  10. Video/Audio File Share during Video Call
  11. Selective Forwarding Unit (e.g. talker notification, Caller ID)
  12. Text Messaging
  13. Contacts Roster
  14. Contacts vCard
  15. Contacts Picture
  16. Message Typing Indication
  17. Message Delivery & Read Notification
  18. Offline Message History

1.2. FreeSwitch

For those who prefer FreeSWITCH instead of Asterisk, another Web Phone version based on SaraPhone frontend and FusionPBX backend is being tested. Saraphone has the more traditional voice only pure SIP features:

  1. AutoAnswer
  2. Attended Transfer
  3. Blind Transfer
  4. Busy Lamp Field (BLF)
  5. Call Error Cause Display
  6. Caller Name and Number Display
  7. Do Not Disturb (DND)
  8. Hold
  9. Hot Desking
  10. Mute
  11. Message Waiting Indicator (MWI)
  12. Notifications
  13. Network Disconnect Reload
  14. Phone Book
  15. Redial

2. Current Limitations

Limitations currently being investigated:

2.1. Closing Window

You must close the WebRTC phone browser window or tab AFTER the call is finished. A proposed interim action of closing the browser window automatically on pressing the Hang Up button is being investigated.

2.2. Outbound Ring Tone

When making outbound calls there is NO ringing tone in some cases, although the destination is still being called and will be put through to you properly when they answer. This is being investigated.