Cloud Voice Go: manual configuration

Overview: whilst we provide pre-configured Technicolor routers and Grandstream ATAs, you may want to try using your existing hardware with Cloud Voice Go. This article outlines the generic settings, and known (tested) configurations for some hardware. If you have successfully used these generic settings on make/model not listed, please let us know so that we can add it to this list.

What is Cloud Voice Go? An economical Pay-As-You-Go replacement for a single phone line, fully hosted in the cloud, to make and receive calls; it includes a free voicemail-to-email service and calls can also be forward to other numbers including mobiles (charged per minute). Call rates are low, and this service can be used for both small businesses and residential consumers who do not make a lot of calls; for high outbound call volumes we recommend the full Cloud Voice which includes unlimited calls. Both Cloud Voice Go and Cloud Voice support porting existing number(s) from existing providers, and/or provision of new UK numbers with your chosen area dialing code.

Generic settings you'll need for every Cloud Voice Go account (we provide these details)

• SIP server: voip.sogeavoice.com

  Proxy: not needed, but can use voip.sogeavoice.com

  STUN server: stun.voiceflex.com

  Registration timeout: 300 seconds

  Unique Username: phone number, such as 01234567890

  Unique Password: Trunk password, such as aBC?1234z

• Additional settings (usually no need to adjust):

  • DTMF: RFC-2833

  • Registration timeout: 300 seconds

Cisco ATA191 ► PDF manual 

• Display name: phone number
• User ID: phone number
• STUN Enable: Yes
• STUN Test Enable: Yes
• Send Resp To Src Port: Yes
• NAT Mapping Enable: Yes
• NAT Keep Alive Enable: Yes

Polycom VVX310 ► PDF manual 

• Identificaiton
  • Address: phone number
  • Label: phone number 
  • Enable SRTP: Yes | Offer SRTP: No
  • Server Auto Discovery: Disable
• Authentication
  • Use Login Credential: Disable
  • Domain: voip.sogeavoice.com  
• Outbound proxy
  • Address: voip.sogeavoice.com  
  • Port: 5060
  • Transport: DNSnaptr
• Server 1
  • Special Interop: Standard
  • Address: phone number 
  • Port: 5060
  • Transport: DNSnaptr
  • Expires: 3600
  • Register: Yes
  • Line Sieze Timeout: 30

Gigaset N300IP ► PDF manual 

• Navigate to Settings [tab] > Telephony > Connections > [Edit one of the spare lines, such as "IP1"]
  • Click Advanced Settings [button]
  • Connection name or number: Cloud Voice Go
  • Personal Provider Data
    • Authentication name:  phone number 
    • Username:  phone number 
    • Display name:  phone number 
  • General Data of your Service Provider
    • Domain: voip.sogeavoice.com 
    • Registration server: voip.sogeavoice.com 
  • Network Data of your Service Provider
    • STUN enabled: Yes
    • STUN server address: stun.voiceflex.com
  • Click Set [button]
  • Mark the line as active (tick "Active" on line "IP1"), then click Set [button]
  • Navigate to Telephony > Number Assignment > [Ensure this new service is marked as enabled for both outgoing and incoming calls, then click Set]

Grandstream HT801 ► PDF manual | ► Firmware

• Navigate to ADVANCED SETTINGS [tab]
  • STUN server is:  stun.voiceflex.com:3478
  • Firmware Server Path: firmware.grandstream.com
  • Allow DHCP Option 66 or 160 to override server: No
  • Apply [button]
• FXS PORT [tab]
  • Primary SIP Server: voip.sogeavoice.com 
  • NAT Traversal: STUN
  • SIP User ID: phone number 
  • Authenticate ID: phone number 

  • Authenticate Password: password

  • Name: phone number 

  • Enable SIP OPTIONS/NOTIFY Keep Alive: OPTIONS

  • Use Random SIP Port: Yes

  • Use Random RTP Port: Yes

  • Apply [button]

  • Reboot [button]

Technicolor DGA0122 | identical config for DGA4134

Connect analogue phone to green port 1

• Navigate to Telephony [tile]
  • Enabled: ON
  • SIP network: [Edit; ✎ pencil icon]
    • Primary Registrar: voip.sogeavoice.com 
    • Primary Proxy: voip.sogeavoice.com 
    • Apply [button; ✔ check icon]
  • Phone Numbers [tab] > sip_profile_0
    • Username: phone number 
    • URI: phone number 
    • Password: password 
    • DisplayName: phone number 
    • sip_profile_0: ON
    • Save [button]