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IP Phone Overview & Compatibility

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An IP phone (also called a SIP phone or VoIP phone) is a desk phone that makes and receives calls over your internet connection rather than a traditional phone line. Instead of plugging into a phone jack, it connects to your office network via an Ethernet cable (or Wi-Fi on select models). The phone communicates using the SIP (Session Initiation Protocol), the same protocol that powers SIPSTACK Nova PBX.

Once connected to your network, an IP phone registers with Nova PBX and behaves just like a traditional desk phone — with the addition of features like call forwarding, voicemail, call recording, and extension dialing built directly into the platform.

SIPSTACK Nova PBX uses auto-provisioning to configure your phones automatically. Rather than manually entering SIP credentials and server addresses on each device, you register the phone in Switchboard and it fetches its configuration directly from the SIPSTACK provisioning server.

Every organization has one provisioning URL and one provisioning PIN that all of its phones share:

https://<slug>:<PIN>@cfg.sipstack.com/<slug>/

<slug> is your tenant slug (also the username) and <PIN> is your provisioning PIN — 8 digits for organizations set up since September 2026. Both are in Switchboard → Nova PBX → System → PBX Settings → Provisioning Access; the PIN is shown once when generated. Lost it? Rotate it in Provisioning Access — phones keep working on the old PIN for 24 hours. There are no per-device passwords: the server tells phones apart by the MAC address you register.

Basic workflow:

  1. Generate your provisioning PIN once, in PBX Settings → Provisioning Access, and put the URL above in DHCP option 160 on the network your phones use (how, per DHCP server). Phones cannot be registered until the PIN has been generated.
  2. Navigate to Nova PBX → Devices → Devices, click Register device and enter the phone’s MAC address (found on the label on the bottom of the phone), brand and model.
  3. Assign an extension to line 1 in the key editor and click Apply Changes — this publishes the phone’s configuration to the provisioning server.
  4. Plug the phone in. It picks up the URL from DHCP option 160 (or you enter it once on the phone), fetches its configuration, and registers with Nova PBX.

For models SIPSTACK does not generate a configuration for (marked Manual setup in each guide), you enter the extension’s SIP credentials on the device by hand.

What each brand does with configuration changes

Section titled “What each brand does with configuration changes”
BrandAfter you click Apply Changes
Yealink, PolyReceive a “reprovision” signal when you click Reprovision on the device page and re-download their configuration
Cisco, Snom, Htek, Poly OBi (Beta)Are sent the same “reprovision” signal; it is not yet confirmed on every model, so reboot the device if it has not picked up the change after a minute
GrandstreamPoll the provisioning server on a schedule — wait for the next poll or reboot the phone
Fanvil, Algo, CyberDataReboot the device to pick up changes

SIPSTACK can keep phones on a firmware version you choose where its firmware catalog has an image for the model — see Firmware pinning. Without a pin, SIPSTACK leaves firmware alone: keep devices on the vendor’s recommended release yourself.

On phones that SIPSTACK provisions with a dial plan, 911, 988 and 112 are sent the moment you finish dialling them. You don’t need to press # or wait.

Because of this, a provisioned desk phone can’t dial a 4-digit extension that starts with 911, 988 or 112 (for example 9115 or 1120). The phone sends the emergency call after the third digit. Choose extension numbers that don’t start with these digits.

Some makes and models carry a Beta label in the Switchboard model picker and in these guides. The label means exactly this:

Models labelled Manual setup are in the model picker, but SIPSTACK does not generate a configuration for them yet; you enter the extension’s SIP credentials on the device by hand.

BrandDevice typesStatusGuide
Yealink (recommended)Desk, conference, DECTAuto-provisioned; newer models Beta; T58W, W90 manual setupYealink Setup Guide
GrandstreamDesk, Wi-Fi, ATA, DECTAuto-provisioned; newer models Beta; GRP260x, GRP2650/2670, GXV, most WP, GAC, GDS manual setupGrandstream Setup Guide
Poly (VVX, Edge E, CCX, Trio)Desk, conferenceAuto-provisionedPolycom VVX Setup Guide
Poly RoveDECTBetaDECT Cordless Phones
Poly OBiATABetaPoly OBi Setup Guide
Cisco (SPA500, MPP 6800/7800/8800, SPA112/122, ATA191/192)Desk, conference, ATABetaCisco Setup Guide
FanvilDesk, intercom, pagingBetaFanvil Setup Guide
SnomDesk, paging, DECTBeta; M400 manual setupSnom Setup Guide
HtekDeskBetaHtek Setup Guide
AlgoPaging, intercomBetaAlgo Setup Guide
CyberDataPaging, intercomBetaCyberData Setup Guide
2NIntercomManual setup2N Setup Guide

Each guide lists every model in the catalog with its status.

Cisco firmware: Cisco 6800, 7800 and 8800 phones work only on Cisco’s Multiplatform (MPP / 3PCC) firmware. On Cisco’s enterprise firmware for Unified Communications Manager they do not read SIPSTACK’s configuration.

Most IP phones support two power methods:

  • Power over Ethernet (PoE): The phone draws power directly from a PoE-capable network switch (IEEE 802.3af). This is the recommended approach — no separate power adapter is needed, and it simplifies cabling.
  • Power adapter: If your switch does not support PoE, use the manufacturer-supplied power adapter (typically 5V or 12V DC, included in the box).

A PoE switch is strongly recommended for office deployments. It also allows the phone to restart cleanly when the switch port is power-cycled, which is useful for remote troubleshooting.

IP phones require a stable internet or LAN connection. For best call quality:

  • Disable SIP ALG on your router or firewall. SIP ALG (Application Layer Gateway) is a feature on many consumer and business routers that attempts to inspect SIP traffic and frequently corrupts it. Check your router’s advanced settings and disable it.
  • Enable QoS (Quality of Service) if your network carries both voice and data traffic. Prioritize RTP and SIP traffic (UDP ports 5060 and 16384–32768) to reduce jitter and packet loss.
  • Use a wired Ethernet connection where possible. Wi-Fi is available on select models (for example the Yealink T53W, T54W and T57W) but a wired connection is more reliable for voice.
  • Minimum bandwidth: 100 kbps up/down per concurrent call using G.711 codec.

For detailed network configuration guidance, see the Network & Firewall knowledge base section.

If you need help selecting a device, registering it in Switchboard, or troubleshooting a connection issue, visit the setup guides for your specific brand:

For additional support, contact SIPSTACK at https://www.sipstack.com/contact.