How to Connect a VoIP Deskphone to Sparkforge
Open Sub-Account Settings > Phone Numbers > Advanced > VoIP Deskphone (SIP), generate a SIP user, assign to a staff member, and enter the credentials in the phone. Test calls confirm registration.
Connect a physical SIP deskphone directly to Sparkforge for inbound and outbound calls. CRM features — call recording, transcription, transfers, contact logging — keep working; you just place and receive calls on a handset instead of the web dialer.
Permissions
- Agency admins: Provision devices in any location.
- Sub-account admins: Provision devices inside their own location.
- Other users: Read-only. They see a tooltip pointing them to an admin.
Network and protocol requirements
- SIP over UDP, TCP, or TLS
- Outbound ports
5060and5061open for SIP signaling - UDP
10000-20000open for RTP audio - PoE switch or external power adaptor
Recommended hardware
Look for SIP and PoE on the spec sheet, at least two programmable line keys, and avoid carrier-locked or proprietary-provisioning devices.
Tested models
- Yealink T54W / T58W
- Poly VVX 450
- Grandstream GXP 2170
- Snom D785
- Cisco 7841
Connect a deskphone
Step 1: Open the wizard
Go to Sub-Account Settings > Phone Numbers > Advanced Settings > VoIP Deskphone (SIP) > Get Started.
Step 2: SIP server configuration
Confirm or edit the SIP domain. A suggested domain like <LocationName>.sip.ashburn.twilio.com appears. You can change it once before saving.
Step 3: SIP user
Pick an extension (number or text) and a strong password. Save them somewhere safe — you'll need both on the handset.
Step 4: Assign user
Each deskphone connects to one Sparkforge user. Pick the user from Assign to User, then open Settings > My Staff > [user] > Edit > Call & Voicemail Settings and enable Deskphone for the call paths you want — Forward Calls to, Ring all, and the IVR default channel as needed.
Step 5: Configure the phone
Enter the SIP Domain, Username, and Password into the phone's Registrar / Server / Proxy fields. Field names vary by manufacturer.
Step 6: Run test calls (optional)
Open VoIP Deskphone (SIP) > Test Calls > Simulate Calls.
- Outbound: Dial the displayed number. A successful call plays "This is a test call" three times.
- Inbound: Click Inbound Test Call. The deskphone should ring.
Manage devices
From Settings > Phone Numbers > Advanced Settings > VoIP Deskphone (SIP) > Manage Devices:
- Pencil icon resets the SIP user password.
- Trash icon deletes the device. Deletion immediately unregisters the handset.
Deskphone-to-deskphone calls
Dial a teammate's extension to call them directly — no external numbers, no IVR routing. Works between any two configured deskphones.
IVR call transfer between deskphones
During an active IVR call, initiate a transfer and dial the teammate's extension. The call routes to their handset. Only blind transfers are supported today; warm transfers aren't available.
Troubleshooting
| Symptom | Likely cause | Fix |
|---|---|---|
| 401 / 403 Unauthorized | Wrong username or password (case-sensitive) | Re-enter credentials |
| No audio | Firewall blocking RTP (UDP 10000-20000) | Open ports or disable SIP ALG |
| Phone rings but won't answer | NAT or SIP ALG interfering with RTP | Disable SIP ALG on the router |
| Can't save SIP domain | Domain name already taken | Edit the domain or accept the suggested increment |
| Inbound calls fail, outbound works | Keep Alive not configured | Set Keep Alive Type to Options under Account > Advanced |
Frequently Asked Questions
Which VoIP deskphones are compatible?
Are calls on deskphones recorded and transcribed?
Is there an extra charge for using a deskphone?
Can two deskphones share one Sparkforge user?
Why aren't inbound calls reaching my deskphone?
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