SIP Phone Setup

The PBX in your admin panel can deliver calls not only to the browser and the mobile app but also to regular office IP phones: a desk phone at the front desk, a DECT handset in a waiter's pocket, a door station at the service entrance or an adapter for an old analog phone. All of them connect the same way: you create an extension for an employee in the admin panel, get five values and enter them into the phone.

This page covers what is the same for every brand: where to get the credentials, what they mean, what the venue network must allow and when a phone must be reset to factory defaults. Step-by-step guides with the screens of specific phones are in the Guides by brand table below.

Open: Admin panel → Telephony → “Operators” tab (admin.cenaly.ru/telephony?tab=extensions).

Credentials#

Credentials are issued per extension, not per phone: every employee (or workplace — “Bar”, “Reception”) has their own internal number and their own account. One extension can be registered on several devices at once — for example, on a desk phone and in the app.

Where to get them:

1. Telephony → Operators. When you create an operator, the “Extension credentials” window opens right away.
2. If the window was closed, the password cannot be viewed — we store only its hash. Click “Reset password” next to the operator: a new password appears, the old one stops working.
3. The same data is available in the employee card (Staff → employee → extension).

The “Copy all settings” button copies the whole window as one text — handy to send to the employee in a messenger. The “Show QR” button is only for the Zoiper and Linphone softphones; desk phones do not understand the QR and are configured with the fields.

The most common setup mistake is entering the extension number (“101”) into the login field instead of the long login sip-…-101. The phone answers “Register Failed” / “Forbidden”.

What to enter into any phone#

If your device has no page of its own in the table below, these values are enough — they are the same for every brand:

There is no “one-click” automatic configuration (auto-provisioning from our server): the phone is configured manually with the fields above or — for a fleet of phones — with your own provisioning server.

Venue network#

The phone only needs outgoing internet access to the PBX address — no port forwarding on the router is required:

• UDP 5060 — signalling (registration, calls);
• UDP 10000–20000 — voice (RTP). The shared contour uses 10000–10200, a dedicated machine uses the whole range;
• UDP 123 — time (NTP), UDP 53 — DNS.

Three things that break telephony most often:

1. SIP ALG on the router (“SIP Helper”, “SIP Passthrough”). The router “helps” SIP by rewriting packets — the phone registers, but the call drops after 30 seconds or only one side is heard. Disable SIP ALG in the router settings.
2. Wi-Fi with client isolation. A venue's guest network usually forbids devices from seeing each other — the phone will not find the computer you are configuring it from. Connect the phone to the staff network, preferably by cable.
3. Power. Most office phones are powered via PoE (802.3af) from the switch; if the switch has no PoE, a power adapter is needed, and it is often sold separately.

How to tell the phone is connected#

• On the phone the “No Service” / “Unregistered” message disappears, the line shows a registration icon or a name.
• In the admin panel (Telephony → Operators) the operator gets a green dot, next to it — the time of the last activity and the string the phone identifies itself with (for example Yealink SIP-T31G 124.86.0.130). The status comes from the PBX and refreshes no more than once a minute.
• Test call: dial a colleague's extension from the phone — or call this extension from another one. If the call goes through but one side cannot be heard — that is not registration but RTP: see “SIP ALG” above.

Factory reset#

A reset is needed more often than it seems, and every brand page describes it separately (from the device and from the web UI). When there is no way around it:

• The phone was bought second-hand or came from a previous provider. Someone else's accounts remain inside, and above all — the auto-provisioning server address of the previous provider: the phone will dutifully download the old configuration and overwrite yours at every reboot. The symptom is “I set it up, it worked, next morning it does not register again”.
• The web UI administrator password is forgotten. It cannot be recovered, only reset.
• The phone behaves strangely after a firmware update or experiments with settings — a reset is cheaper than an investigation.

What gets erased: accounts, network settings (static IP, VLAN), language, directories and call history. The firmware stays. After the reset: language → time zone → network (if not DHCP) → account.

Guides by brand#

One page — one family of phones with a common web UI. Each page covers: how to find the IP, the factory password, the account screen with a screenshot from the manufacturer's documentation, verification, factory reset and the typical errors of that brand.

Softphones (Zoiper, Linphone, MicroSIP, the smartphone app) are configured with the QR code or the same fields — they do not need a separate page.

If it does not register#

In order, from the most common to the rarest — regardless of the brand:

1. The login was not entered in full. It must be sip-…-<number>, not the number itself. Check both fields — User Name and Auth Name.
2. The password is outdated. Someone clicked “Reset password” — the old one no longer works. Do not guess, issue a new one.
3. Wrong server address. A venue with a dedicated machine has its own address, not the shared sip.<domain>. Copy it from the window.
4. TCP transport with “silence”. If the phone does not even get a rejection — switch to UDP.
5. Network. The router blocks UDP 5060, SIP ALG is on, the phone is on the guest Wi-Fi.
6. Foreign provisioning. Settings “change by themselves” — see “Factory reset”.
7. Wrong firmware. Devices with Microsoft Teams / Skype for Business / Zoom firmware (Yealink, Poly, AudioCodes and Cisco have such variants) do not register on a regular SIP PBX until the SIP firmware is loaded. The model looks the same from the outside — check the boot screen.

If this did not help — the brand page lists the messages of that specific device and what they mean.

FAQ#

Which phone should I buy?#

For the front desk and the office — any desk SIP phone from the table; mass-market and inexpensive are Yealink T31/T33, Grandstream GRP2601/2602, Fanvil X3/X4. For waiters and the floor — a DECT base with handsets (Yealink W70B, Grandstream DP752, Gigaset N670): one base covers the floor, the handsets stay in pockets. For a hotel — hotel models without extra buttons (Fanvil H series, Snom HD1xx). For an old analog phone — an ATA adapter (Grandstream HT801). The main thing: the model must support open SIP; “Teams-only” and “Zoom-only” devices will not work.

Can one extension be put on two phones?#

Yes: one account registers on several devices at the same time (for example, a desk phone and the app), the call reaches all of them. If you need calls to ring in turn or simultaneously for different people — that is a ring group in “Telephony → Ring groups”, not a shared account.

An employee got a new extension number — does the phone need reconfiguring?#

Yes. The login contains the number, so a new number means a new account and a new password. Rewrite the login (in both fields) and the password in the phone.

Do I need a static IP or port forwarding?#

No. The phone establishes the connection to the PBX from inside your network; no incoming connections from outside are required. Port forwarding on the router is unnecessary and harmful.

Is there automatic configuration by MAC address (provisioning)?#

No. The platform does not run a provisioning server: the phone is configured with the fields from the “Extension credentials” window. For a fleet of dozens of devices use your own provisioning server — the brand pages have a collapsed “Fleet provisioning” block with parameter names.

Is encryption (TLS/SRTP) supported?#

Not at the moment: registration goes over UDP/TCP to port 5060, voice is plain RTP. Do not enable TLS/SRTP in the phone “just in case” — it will not register.

Related sections#

• Staff & shifts — who gets extensions
• Call from website and Callback — guest calls land on the same extensions
• Needs attention — missed calls in the shared to-do list