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 → “Staff & groups” tab (admin.meni.ge/telephony?tab=extensions).

Find your phone#

Type the model from the sticker on the bottom of the device or from the boot screen — you get the guide for exactly that model and its factory reset section.

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 → Staff & groups. 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 meant for softphones, but their scanners primarily expect a provisioning link from their own service: Zoiper is worth a try, current Linphone versions will not accept this code — details in “Softphones”. Desk phones do not understand the QR and are configured with the fields.

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.

In detail — where to turn off SIP ALG on popular routers, double NAT, Wi-Fi and VLANs, how much bandwidth a call needs: “Network, router and internet”.

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 → Staff & groups) 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.
• Audio check: dial *43 and the PBX plays your own voice back to you. Registration and audio break independently, and the green dot says nothing about audio: see “Checking call quality” below.

Checking call quality#

A registered phone is not yet a phone you can be heard on. Registration and audio travel over different channels and break independently: registration is port 5060, while the voice is a separate RTP stream on ports 10000–20000. The green dot in the admin panel stays lit even when no audio gets through at all.

That is why the PBX has a service number — *43, the echo test. Dial it from the phone you have just set up:

1. The PBX answers and, after a second, plays a short beep.
2. Then it plays your own voice back to you: speak into the handset and you hear yourself.
3. Speak for about ten seconds, not one — otherwise the test has no time to measure itself (why — below).
4. When you are done, just hang up (or press # — you will hear a second beep).

The test costs nothing, calls nobody and disturbs nobody: it never leaves your PBX. Run it as often as you like, including on a live venue at peak hour.

What the result means#

The beep comes before the echo on purpose: it tests one direction on its own, before the microphone gets involved. Hence four different outcomes — and they are four different faults:

Some delay in the echo is normal and equals the round trip to the PBX: under 300 ms it is barely noticeable. If your own voice comes back noticeably late (half a second or more), calls will be uncomfortable for guests — the link is to blame, not the phone.

Numbers instead of impressions#

Your ear honestly says “bad”, but it does not say “how bad” — and a conversation with the carrier or with whoever set up the router has to happen in numbers. So every *43 test lands in the call log (Telephony → Log) as an internal call, with the measurements attached:

• packet loss (loss) — under 1 % is unnoticeable, above 3 % sounds like gurgling and dropped syllables;
• jitter (jitter) — the spread in packet arrival times; up to 30 ms the phone smooths it out itself, past 50 ms the audio starts to break up;
• latency (rtt) — round trip; under 150 ms the conversation feels natural, past 300 ms people start talking over each other. Phones that do not send RTCP report no latency — that is “not measured”, not “zero”;
• packet counters (rx / tx) — how many arrived and left. A zero in rx is exactly “the PBX cannot hear you”.

The same log is a ready-made answer for the carrier: date, time, loss and jitter for a specific device.

The other way round: let the PBX call you#

In the admin panel every device has a card (click the equipment tag in the extension row), and in it — the “Check” tab with the “Call this phone” button. Press it and the PBX rings the phone of that extension showing “Line check” on its screen; you pick up, hear “speak and you will hear yourself”, a beep, and then the very same check. The result lands in the same log with the same numbers.

This does not replace *43, it is its other half: a hand-dialled *43 checks the road from the phone, a call from the PBX checks the road to the phone. That is the answer to the most common complaint “the dot is green but incoming calls do not ring”: the phone refreshes its registration from the inside, so it looks alive in the admin panel — while an incoming call no longer reaches it (a stale binding on the router, a short keep-alive, a firewall). The phone rings — the road is there; it stays silent — look at the router.

What to know beforehand:

• you still pick up the handset yourself: neither the button nor the admin panel knows whether you can hear anything;
• if the extension has several devices, one of them rings — to check a particular phone of a pair, dialling *43 on it is still the reliable way;
• an inherited carrier number has no button: such a phone is registered on someone else's station;
• the button only appears when your station can place such a call. No button — the manual *43 works as before.

When to run it#

• Right after connecting a phone — before the device is handed to an employee. That is the only moment when an audio fault is caught for free rather than by a guest complaint.
• After any network work — a new router or carrier, the phone moved to another socket or network, a firmware update.
• When someone complains about quality — start with *43 from that same device: in a minute it separates “bad line on our side” from “bad line on the caller's side”.

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.

How it is done on every brand — in one table (on the device, without the password, from the web interface, login after reset): “Factory reset: all models”.

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 same fields — step by step for each client: “Softphones”.

If something does not work#

Dial *43 first — the ten-second echo test tells you whether the problem is registration, audio or the network. Then go by symptom:

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#

How do I check the line without calling real people?#

Dial *43 from the phone and the PBX plays your own voice back to you (echo test). It tests the audio rather than the registration, costs nothing and disturbs nobody: the call never leaves your PBX. What each outcome means is covered in “Checking call quality”.

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
• Line code *code*number and the phone's dial plan — when a phone won't dial a provider's line code (“check the number”)
• Distinctive rings on SIP phones — a different ring for a guest call, a colleague and a transfer