Manually provision a device in Business Phone

Deskphone Provisioning

Yealink Deskphone

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning your Yealink

1. Get the IP Address:On Handset click “Ok” in middle of direction pad > Status > System Status. Enter this in to where you need to place your URL in your browser. Username: adminPassword: adminOnce logged in go to Account > Basic and make sure the following are set correctYou may notice that Account 1 is selected at the top. If you would like multiple lines on a supported desktop softphone, you must have a third party softphone item for each line, and program each individually for Account 1, Account 2, etc. Line Active: EnabledLabel: Extension NumberDisplay Name: Extension NumberRegister Name: SIP UsernameUsername: SIP UsernamePassword: SIP PasswordEnable Outbound Proxy Server: EnabledOutbound Proxy Server: proxy-west.virtualpbx.net / Outbound Proxy Server Port: 0SIP Server: realm name / SIP Server Port: 0Click “Confirm” to save

   

   Then click on Account > Codecs

   

2. Drag the PCMU codec from the left column to the right column

3. Confirm changes

Yealink Base & Wireless Handsets

Depending on your Yealink model and firmware version, your Yealink interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning your Yealink

1. Get the IP Address:On Handset click “Ok” in middle of direction pad > Status > System Status. Enter this in to where you need to place your URL in your browser. Username: adminPassword: admin

   

   Once logged in go to Account > Basic and make sure the following are set correctly

2. You may notice that Account 1 is selected at the top. If you would like multiple handsets to register individual lines with VirtualPBX, you must have a third party softphone item (Instructed above) for each line, and program each individually for Account 1, Account 2, etc.

3. Line Active: Enabled

4. Label: Extension Number

5. Display Name: Extension Number

6. Register Name: SIP Username

7. Username: SIP Username

8. Password: SIP Password

9. Enable Outbound Proxy Server: Enabled

10. Outbound Proxy Server: proxy-west.virtualpbx.net / Outbound Proxy Server Port: 7000

11. SIP Server 1 - Server Host - Account Realm - SIP Server Port: 5060

12. Click “Confirm” to save

13. Account > Codecs

    

14. Enable PCMU codec only.

15. Click “Confirm” to save.

    

    1. Navigate to Account > Handset Name. 1. Name each handset what you would like to see displayed on handset. Name, extension number or something else. 1. Click “Confirm” to save.

16. 1. Navigate to Account > Handset Name.

17. 1. Name each handset what you would like to see displayed on handset. Name, extension number or something else.

18. 1. Click “Confirm” to save.

    

19. Navigate to Account > Number Assignment. This is where you determine which handset can answer which lines when taking and making calls. Right now ext 815 will ring only handset 1. You can have it ring others or different ones based on the checkboxes selected above. Click “Confirm” when the changes are made.

    

20. Navigate to Phone > Preference and make sure the following are set correctly:

21. Time Server: pool.ntp.org

22. Time Zone: Select local time zone

23. Daylight Savings Time: Automatic

24. Click “Confirm” to save

Aastra

Depending on your Aastra model and firmware version, your Aastra interface may look different than the screenshots included below. The steps, however, are the same.

Start by creating a Soft Phone

1. Click on Users

2. Click on the Devices tab

3. Choose Soft Phone

Change your device name and record the SIP Username, password, and realm to use in the Aastra interface. You can always return to reference your credentials by clicking on devices (under users), selecting the device, clicking edit, advanced, and selecting “SIP”.

Provisioning your Aastra

1. Change your device name and record the SIP Username, password, and realm to use in the Aastra interface. You can always return to reference your credentials by clicking on devices (under users), selecting the device, clicking edit, advanced, and selecting “SIP”.

2. Click “Create Device”.

3. Next, find the phone’s IP Address by pressing the Options button, scrolling down to and selecting Phone Status, then IP and MAC Addresses. If it says 0.0.0.0, the device is not properly set up on the network and you should talk to your system administrator.

4. Open a browser window and go to http://[IP Address] using the IP Address found in the last step.

5. Next, you will be asked to enter a username and password. If the default username and password below do not work, please contact Aastra to reset the phone to factory defaults:User: adminPassword: 22222

   

6. Select Line 1 in the Aastra interface and enter the following information:

• Screen Name: Your Name or Extension

• Phone Number: use the SIP Username from Dash

• Authentication Name: use the SIP Username from Dash

• Password: use the SIP Password from Dash (click on the field to reveal the password in plain text)

• Proxy Server: use the Realm from Dash

• Registrar Server: use the Realm from Dash

• RTP Encryption: Please set to SRTP Disabled

1. Click on Save Settings.

2. Your phone should now be successfully registered.

SNOM

Depending on your SNOM model and firmware version, your SNOM interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

1. Once selected, a window will appear. Here you will need to name the device. You’ll see that the SIP Username, password and realm will be provided. Save once you have named the device. You can get back to this screen by clicking on the devices wrench (edit) icon to the right of the new device then click on advanced and select “SIP”. You will need this information in the next step.

Provisioning your SNOM

To access the web interface of the SNOM device, go to the physical phone and press the Settings key. Scroll down to Maintenance, and then System Info. Type the IP address found here into your web browser. You will be prompted to login. The default username is “admin” and the default password is also “admin”. Please note that the computer and phone must be on the same network.

1. Once logged in, go to Identity 1 > Login and input the credentials from step 4 in your Dash account. When you are finished, click Save. Account = SIP UsernamePassword = Password (click on the password field in Dash to reveal the password)Registrar = Realm

   

2. Go to Identity 1 > RTP Tab and turn RTP Encryption Off. When you are finished, click Save.

Bitrix 24 CRM Configuration

Depending on your Bitrix model and firmware version, your Bitrix interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. You’ll be using the SIP user name SIP Password and Realm information, along with http://proxy-west.virtualpbx.net to enter into the Cloud PBX.

5. Make sure to click on Create Device, and then Save changes on the user.

Provisioning your Bitrix

1. In Bitrix 24 navigate to Telephony and click on Cloud PBX.

2. Click on Connect Cloud PBX in the upper right and fill in the following fields.

3. Once this is done click Save.

4. You can check the status by clicking on Connection Settings.

5. Go to the main Telephony screen and Click on Configure Telephony. You’ll need to configure the Telephony Settings followed by the Telephony user. See the screenshots below.

6. Then, set up your Telephony Settings.

7. Under Telephony Users select the agent you wish to place into this setup. Click on the hamburger menu to configure. Set the extension number if it is not already set. Choose the sip connector and sip phone then save.

8. Going Back to Cloud PBX, you can add additional devices and adjust incoming call rules.

Grandstream IP

Depending on your Grandstream model and firmware version, your Grandstream interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your Device a name for your identification.

5. Record your SIP Username, SIP Password, and Realm – you will need these shortly. To display your SIP Password simply click on the text box to reveal it.

Provisioning your Grandstream

Login to your Grandstream phone’s interface using username and password “admin” – you may need to refer to your phone’s manual if you have forgotten how to access this interface.

1. Click on Accounts

2. Click on “General Settings” under account 1 on the left

3. Account Active: Set to Yes

4. Account Name - Name this device for your identification

5. SIP Server: (enter the Realm you were given in the Dash interface)

6. Outbound Proxy - proxy-west.virtualpbx.net

7. Backup Outbound Proxy - proxy-east.virtualpbx.net

8. SIP UserID: (enter the SIP Username you were given in the Dash interface)

9. SIP Authentication ID: (enter the SIP Username you were given in the Dash interface)

10. SIP Authentication Password: (enter the SIP Password you were given in the Dash interface)

11. Click on Save

Panasonic TGP 550/550/600

Depending on your Panasonic model and firmware version, your Panasonic interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning Your Panasonic TGP 550/550/600

1. First, you’ll need your IP AddressPress Menu (center joystick)Enter #501Write your IP address down

2. Next, enable Web InterfacePress Menu (center joystick)Enter #534Set Embedded Web to “On”Press Enter

3. Enter IP Address into Web browserUsername: adminPassword: adminpass

VoIP > SIP Settings > Line 1

1. Phone Number: Enter what you want displayed (does not display on TGP50 Handsets)

2. Line ID: Enter SIP Username

3. Registrar Server Address: proxy-west.virtualpbx.net

4. Registrar Server Port: 7000

5. Proxy Server Address: proxy-west.virtualpbx.net

6. Proxy Server Port: 7000SIP

7. Service Domain – Enter realm

8. SIP Source Port – 7000 (increased by 10 for each line)

9. Authentication ID – Enter SIP Username

10. Authentication Password – Enter SIP Password

11. Enable SSAF (SIP Source Address Filter): Yes

12. Click “Save” on Bottom

Telephone > Call Control > Line 1

1. Voice Mail Access Number – *97Click “Save”

Telephone

After programing multiple lines as necessary, you can use the “Telephone” tab to determine which lines ring/use which handsets. The top part “Grouping Handset/Handset selection for receiving calls” tells the Panasonic base when a call comes in on Line X, ring handset X. In the example below, line 1 will ring all registered handsets (1-6). You can program line 1 to only ring handset 1, or handset 3 and 6 or any combination.

The bottom part “Handset and Line No. selection for making calls” allows you to select when a handset makes an outbound call, which line it can use. In the above example, Handset 1 can make an outbound call on any line but by default it goes through line 1. You can change it so handset 2 only rings from line 2 or handset 4 rings from line 2 and 5 but defaults to 5.

There are two main types of behavior you can program.

• All handsets ring simultaneous (which is the default selection)

• Have each line ring handsets individually. You will want to adjust the Telephone tab so each handset is assigned to it’s own individual line (IE: Line 1 / Handset 1 - Line 2 / Handset 2).

Panasonic KX-UT113/123

Depending on your Panasonic model and firmware version, your Panasonic interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning Your Panasonic KX-UT113/123

1. Find your IP Address:Go to Settings > Information Display > IP address, on your computerEnable Web Interface by going to Setting > #534 > Set Embedded Web to “On” on your computer > EnterEnter IP address into Web browserUsername: adminPassword: adminpass

   To Enable VoIP - Click on SIP Settings > Line 1

   

2. Phone Number – Enter what you want to be displayed on the phone

3. SIP URI – Enter SIP Username

4. Registrar Server Address: proxy-west.virtualpbx.net

5. Registrar Server Port - 5060

6. Proxy Server Address – proxy-west.virtualpbx.net

7. Proxy Server Port - 5060

8. SIP Service Domain – Enter realm

9. SIP Source Port – 5060

10. Authentication ID – Enter SIP Username

11. Authentication Password – Enter SIP Password

12. Enable SSAF (SIP Source Address Filter) – Yes

13. Click Save on bottom

    

14. Phone Number – Enter what you want to be displayed on the phone

15. SIP URI – Enter SIP Username

16. Registrar Server Address: proxy-west.virtualpbx.net

17. Registrar Server Port - 5060

18. Proxy Server Address – proxy-west.virtualpbx.net

19. Proxy Server Port - 5060

20. SIP Service Domain – Enter realm

21. SIP Source Port – 5060

22. Authentication ID – Enter SIP Username

23. Authentication Password – Enter SIP Password

24. Enable SSAF (SIP Source Address Filter) – Yes

25. Click Save on bottom

26. To Enable Call ControlTelephone > Call Control > Line 1Voice Mail Access Number – *97Click Save on bottom

    

    To Adjust Your Time Settings - System > Time Adjust Settings

    1. NTP Server Address – pool.ntp.org

    2. Time Zone – Select based off your time zone (pacific -8, mountain -7, central -6, eastern -5) 3. Click save on bottom

27. NTP Server Address – pool.ntp.org

28. Time Zone – Select based off your time zone (pacific -8, mountain -7, central -6, eastern -5)

29. Click ‘Save’ on bottom

Sangoma S305

Depending on your Sangoma model and firmware version, your Sangoma interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning Your Sangoma

1. Next, open a browser and login to the phone’s IP address: Admin / 4221200 (this may vary depending on your provider. Please verify with your system administrator)”.

2. Click on “Go to the “Account” Tab – “Basic””.

3. Finally, verify that the settings match the following:

   

4. Account Active – Yes

5. Primary Sip Server – Account Realm

6. Outbound Proxy – proxy-west.virtualpbx.net

7. SIP Transport: UDP

8. SIP User ID – SIP Username

9. Authenticate ID – SIP Username

10. Authenticate Password – SIP Password

11. DNS Mode – SRV

12. Click on “Save Settings”.

13. Congratulations! Your Sangoma S305phone should now be successfully registered.

Polycom

Depending on your Polycom model and firmware version, your Polycom interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your Device a name for your identification.

5. Record your SIP Username, SIP Password, and Realm – you will need these shortly. To display your SIP Password simply click on the text box to reveal it.

Provisioning your Polycom

Next, you’ll need to find the private IP address of your device, and then input that in your web browser. You can usually find the IP address in your phone’s settings, look for a “Status”, or potentially a “Network” setting. Private IP addresses typically look like 192.168.1. X (X can be any number between 2 and 255) or 10.0.0. X

Once you’ve found it, type it in your web browser URL as if you’re entering the address for a website. It should come up with the Manufacturer interface for your device.

Default Login for Polycom devices are:

• Username: Polycom (Case sensitive)

• Password: 456

Now that you’re in, complete the following steps:

Go to Settings tab at the top of the page

Then Lines and select Line 1

Under Identification:

1. SIP Username: Enter in The Address and Authentication User ID

2. SIP Password: enter in the Authentication Password

   3. Realm: Enter in the Address field

3. Click Save

   Setting Date/Time: Click on Preferences > Date & Time

   

4. Adjust the time/date format to suit your preference

5. Then select the correct time zone for your location

6. Click Save

Vtech VCS752/VSP805/VSP810/VSP815G/VPS945G/VSO875G

Please note that these are older model phones. These instructions are offered as a courtesy for end-of-life devices to allow them to make and receive calls.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your Device a name for your identification.

5. Record your SIP Username, SIP Password, and Realm – you will need these shortly. To display your SIP Password simply click on the text box to reveal it.

6. Find the IP Address: On your VTech deskset or base station (for DECT), press Menu, go to Status, then Network. Note the IP Address.

7. Access WebUI: Open a web browser on a computer connected to the same network and type in the VTech phone's IP address.

8. Log In: Log in with the administrator account. (Default credentials are typically admin for the username and 1592 for the password).

9. Configure the SIP Account:Navigate to the Network or Account settings tab. Select an unused Account / Line. Set Account Active to Yes. Enter your Dash Username into the Auth Name, User ID, and Label fields. Enter your Dash Password into the Password field. Enter your Dash Realm into the Server Address or SIP Server field. Set the SIP Transport (usually UDP).

10. Save and Reboot: Click Save or Apply at the bottom of the page. The phone will register to the Dash platform

Avaya J129/J139/J159/J179

Please note that these are older model phones. These instructions are offered as a courtesy for end-of-life devices to allow them to make and receive calls.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   To remove any previous enterprise configurations:

4. Press the Main Menu or Admin key on the phone.

5. Enter the default administrator password (e.g., 27238) and press Enter.

6. Scroll to and select Reset to defaults (or Clear).

7. Press Reset to confirm and reboot the phone.

   Configure SIP Settings. You can program the phone via its physical keypad:

8. During the initial boot session, press No to skip Auto-Provisioning.

9. Go to the Administration menu and enter the admin passcode (27238). Select IP Configuration > Servers.

10. Enter your Dash SIP details into the appropriate fields:SIP Server/Proxy: Your Dash domain. Registrar Server: Your Dash domain. Domain: Your Dash realm. Port: 5060 (or your specific Kazoo proxy port).

11. Navigate to your Accounts/Extension settings and input your Kazoo Extension Username and Password.

4. Save and Reboot

Once all settings are input, save your changes and reboot the phone. It should now successfully register on your Dash platform.

ATA & Misc Provisioning

ATA

An ATA adaptor is a converter box that allows you to use an analog phone/fax machine via your internet connection. On one side you plug in the analog device, on the other is a network cable that will go to your router. Utilize this option for an analog fax machine.

Device Name: This is the name of the device that will show up in your Dashboard. It does not affect any part of provisioning or your system functionality.

SIP Username: This is a system-generated username that will be used to input into your ATA device’s web interface. Please avoid changing this item in the Dashboard to avoid complications with provisioning and registration.

SIP username is often entered into fields such as: User ID / Account name / Authentication name or ID / Authorization name / SIP User ID

SIP Password: This is a system-generated username that will be used to input into your ATA device’s web interface. Please avoid changing this item in the Dashboard to avoid complications with provisioning and registration.

SIP Password is often entered into fields such as: Password / account password / authentication password / SIP password

Realm: This is a system-generated string unique to your VirtualPBX account which will be input into your ATA device’s web interface

Realm is often entered into fields such as: Domain / Server / Proxy or Proxy Server / Register server / SIP domain or Server

It is possible that you may need to setup the Outbound Proxy server as well

Outbound Proxy should be set to proxy-west.virtualpbx.net

Mouse over Advanced and you will see your Advanced options. In general these can be adjusted as you see fit or simply left as is.

1: Click on Advanced in the top right, and select “Audio”

2: Move the following options from the left column to the right column in descending order: PCMU / PCMA / OPUS

Click on Create Device when done.

Cisco SPA 112

Depending on your Cisco model and firmware version, your Cisco interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning your Cisco SPA

1. Get the IP Address for your Cisco SPA 112Hook up analog phone to Phone 1Pick up and press ****Enter 110# and record IP address

2. Enter IP Address into web browser and loginusername: adminpassword: admin

3. In Quick Setup (Initial Screen) enter the following informationProxy – SIP RealmDisplay Name – Name of PhoneUser ID- SIP usernamePassword – SIP Password

   

4. Navigate to the Administration tab, select Voice > Line 1, and ensure the outbound proxy and DNS SRV information is correct.

   

5. Click Submit to save. No other changes are needed by default.

Algo Paging System

Depending on your Algo model and firmware version, your Algo interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning your Algo

• Go to the Basic Settings Tab

• Click on the SIP tab, and input the following information copied previously

  

• Domain : Realm

• Page Extension: SIP Username

• Authentication ID: SIP Username (Again)

• Authentication Password: SIP Password

X-Lite & Bria

Depending on your model and firmware version, your interface may look different than the screenshots included below. The steps, however, are the same.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning your X-Lite/Bria

1. Open X-Lite/Bria

   

2. Go to Account Settings and complete the following:Uncheck IM/PresenceUser ID = SIP UsernameDomain = realm listed for softphonePassword = SIP PasswordDisplay Name = SIP usernameAuthorization Name = SIP Username

   

3. Review Voicemail settings. Set the number to dial for checking voicemail to *97

   

4. Review Topology SettingsRange of ports used for signaling – 10000 to 25000Range of ports used for RTP (both audio and video) – 10000 to 25000

   

5. Review Presence settings. In most cases no changes will be necessary.

   

6. If you use the X-Lite version, review Transport settings. In most cases no changes will be necessary. If you use the Bria version, your Transport settings should look like this:

   

7. Review the Advanced settings. In most cases no changes will be necessary.

Grandstream GDS3712 Intercom

The GDS3712 requires a manual wiring setup with an ethernet cable as it does not have an RJ45 plug. You can find the wiring instructions here: https://documentation.grandstream.com/knowledge-base/gds3710-gds3712-user-manual/

We recommend that you start by making sure that your Grandstream device has the most recent version of the firmware. Please note that updating the firmware may change the screen images that we present here. The purpose of the fields should be the same regardless of version.

1. Click on Users on the left

2. Click on the Devices box for the user

3. On ‘New Devices’, click on Softphone

   

4. Give your device name & record the SIP Username/Password for future use

5. Click Create Device.

Provisioning your Grandstream

Find the IP Address of the GDS3702 from your router and enter it into the web browser for the Login Page.

Username: Admin Password: Password is on the sticker on the front face of the intercom. First time logging in, it will ask you to change the password. Please note the new password. Once Logged in, go to Account > Account 1 (on the left side)

SIP Server = Realm Outbound Proxy = proxy-west.virtualpbx.net DNS Mode = NAPTR/SRV SIP User ID = SIP Username Authentication ID = SIP Username Password = SIP Password Display Name = Name of device Click “Save” at bottom to save account information.

To configure the button on the intercom to ring do the following: Next go to Door System Settings > Basic Settings DoorBell Call Out Account = Account 1 Doorbell Mode = Call Doorbell Number Number 1 Called When Door Bell Pressed = Extension number you want dialed when the intercom button is pressed. Primarily should be a user extension or ring group extension Click “Save”

When answering the call from the GDS3712, please make sure to answer it as a video call if you want to see video.. If you answer as an audio call, it will only stay audio even if you try to enable video afterwards.