Phone not working: troubleshooting by symptom

Every SIP phone has three independent roads, and they break separately:

1. Registration — the phone tells the PBX “I am here”. These are signalling messages to port 5060, and they are exactly what the green dot in the admin panel shows.
2. Voice — a separate stream of packets on ports 10000–20000. The green dot knows nothing about it.
3. The way back to the phone — the road the PBX uses to deliver an incoming call. The venue’s router keeps it open — and quietly closes it, too.

So “not working” almost never means “the phone is broken” or “the PBX is broken”: usually one of the three roads is blocked in the venue network. This page helps you find out which one in five minutes and takes you to the detailed walkthrough. Setting up phones from scratch is covered in the overview “SIP Phone Setup”.

Check the admin panel first#

Everything below is in Admin panel → Telephony; nothing needs installing.

1. The dot next to the extension (“Staff & groups” tab). “Connected” — the phone is registered; “No connection” — it was there and dropped off; “Connection not checked yet” — it has never registered; “Turned off” — the operator is switched off in the admin panel itself. The status comes from the PBX about once a minute: after changing something on the phone, give it a minute.
2. The equipment tag in the extension row — the model and firmware the phone reported (for example Yealink SIP-T31G 124.86.0.130). A grey tag means the device is not connected right now. Someone else’s model in the tag means the wrong device is registering with this login.
3. The device card (click the tag). “Uptime” — a connected / not connected bar for a day, a week or a month: regular gaps are the signature of a router closing the way back. “Connections” — when and from which address and network the phone registered. “Check” — recorded echo tests and the “Call this phone” button.
4. The *43 echo test from the phone itself. Ten seconds of talking to yourself checks the voice in both directions: a beep and your own voice — all good; a beep but no voice — audio does not get from the phone; silence — audio does not get to the phone. All outcomes are explained in “Checking call quality”.
5. “Check connection” on the “Overview” tab walks the chain “server → line → route → extensions” and names the first broken link. This one is about incoming calls to your phone number.
6. The call log — every call shows who picked up, how long the conversation lasted, who hung up, the hangup cause and an “Audio” line: whether voice flowed both ways, packet loss, jitter. How to use it — “Call log”.
7. The SIP trace (“Server” tab) — a transcript of what the PBX and the phone said to each other, for when nothing else explains it. Instructions — “Wire diagnostics”.

What the admin panel does not show yet is the reason a phone failed to register: the PBX records successful registrations, while a rejection is seen by the phone itself (the message on its screen) and by the SIP trace. Messages and codes are decoded in “Not registering”.

Checklist: admin panel and phone#

Most faults are a mismatch between the two sides: one thing in the admin panel, another in the phone. Go through the table row by row — the left column in the admin panel, the right one on the phone and the network.

Symptom index#

No incoming calls, outgoing calls work#

Why it happens#

An outgoing call is started by the phone itself, from inside the network — the router lets it out. An incoming call arrives from outside: the PBX knocks on the address it remembered at registration, and the router lets it in only while it keeps open the “door” the phone opened with its last message. After a while of silence routers close such doors. Hence the signature: you call out — it works; someone calls you — silence, and the dot in the admin panel may still be green.

Causes, from the most common to the rarest:

1. The router closed the way back. Keep-alive is off on the phone, or its interval is longer than the router keeps the door open. Details — “Incoming calls only work for the first minutes”.
2. The call never reached the extension by its route. The number has “Accepts calls” switched off, a routing rule sends it to another group, the employee is outside their schedule in the group.
3. Do Not Disturb (DND) or unconditional forwarding is on in the phone.
4. More devices on the account than allowed. By default one account holds up to 5 devices, and each new one pushes out the oldest: the pushed-out device still calls out fine, but incoming calls no longer reach it.
5. The phone’s protection is too strict — it accepts calls “only from the server”, but the server address in the phone does not match the real one (see night calls).

What to do#

After each step, test an incoming call; go further only if it did not help.

1. “Overview” → “Check connection”. If it names the “line” or the “route” link, the phone is not the problem: open what it names.
2. Device card → “Check” → “Call this phone”. It rings — the road to the phone is there, look at the routing. It stays silent with a green dot — it is the router (step 4). Without a dedicated PBX, call this extension from the phone next to it.
3. Turn off DND and forwarding on the phone.
4. Enable keep-alive in the phone — on Yealink it is Account → Advanced → Keep Alive Type and Keep Alive Interval (factory interval 30 s) — and turn off SIP ALG on the router. Test an incoming call after 10 idle minutes, not straight away: right after setup the door is still open.
5. Open “Connections” in the device card: if several devices keep “replacing” one another on one account, disable the extra ones or give them separate extensions.

The voice menu ignores key presses (DTMF)#

Why it happens#

A key press during a call has to be delivered to the other side somehow. There are three ways: as a separate signalling packet next to the voice (RFC 2833, also known as RFC 4733), as a sound inside the voice (inband), and as a SIP INFO message. By default our PBX expects the first — RFC 2833 / 4733, and almost every phone sends exactly that out of the box. If the phone is set to another method, the PBX will not see the press, and the voice menu — yours or someone else’s — will not hear it.

A separate case is an analogue phone behind a gateway in pulse dialling mode (a rotary phone, or a P/T switch set to P): it cannot send star or hash at all.

What to do#

1. Test key presses with the echo test. Dial *43, wait for your voice and press #. A second beep and the test ends — the PBX received your press, the phone is fine. No beep, the echo goes on — the press does not reach the PBX.
2. Set the phone to RFC2833. On Yealink it is Account → Advanced → DTMF Type (RFC2833 is the factory value); on other brands the DTMF field sits in the same account settings. Replace “RFC2833 + SIP INFO” with plain RFC2833 — two methods at once are not needed.
3. On a dedicated PBX the method can be changed on the PBX side for a fussy device: at the extension — “Phone settings” → “How key presses are sent” (inband, info, auto). Change one value at a time and test each with the same # in *43.
4. # works in *43, but the bank menu still does not hear you — the press is lost between the PBX and the carrier. That is ours to fix: send us the call time and the number you dialled (see “What to send to support”).

Night calls from 100, 1000, admin#

Why it happens#

It is neither a guest nor the PBX. Scanner programs roam the internet around the clock: they go through addresses and send test calls to port 5060 as “100”, “1000”, “admin”, looking for unprotected phones. Such a call reaches the phone only if the phone is exposed to the internet directly: port 5060 is forwarded to it on the router, it sits in a DMZ, the router opened the port itself via UPnP, or the phone got a public address without a router. The PBX call log has no such call — it bypassed the PBX.

Our PBX needs neither forwarding nor DMZ: the phone reaches it on its own from inside the network.

What to do#

1. Remove port forwarding and DMZ to the phone on the router, turn off UPnP. If the provider’s cable goes straight into the phone, put a router in between. Test an incoming call from the extension next to it.
2. Let the phone accept calls only from its own server. Yealink: Features → General Information → Accept SIP Trust Server Only — according to the vendor’s documentation this setting exists precisely against “ghost” calls from numbers like 100 and 1000. Grandstream GRP: Account → SIP Settings → Security Settings → Accept Incoming SIP from Proxy Only. Grandstream HT gateways: Allow Incoming SIP Messages from SIP Proxy Only. After switching it on, always test a real incoming call: if the server address in the phone does not match the real one, genuine calls will be cut off too.
3. Calls continue with the ports closed — the phone is still visible from outside: check whether it sits on a network with public addresses (some providers connect offices this way) and move it behind a router.

The phone gets no IP address#

Why it happens#

The phone gets its address from the venue’s router or switch (DHCP). No address — the PBX never comes into play: the phone is simply not on the network. Causes: the cable or the port, power, a static address or VLAN left in the phone by a previous owner, a wall socket in a network segment that does not hand out addresses.

What to do#

1. Cable. Plug it into a port where a computer definitely works. Desk phones have two sockets: the network goes into LAN / Internet / WAN, not PC (that one is for a computer behind the phone).
2. Power. Without a PoE switch you need a power adapter. A PoE switch often powers only some of its ports — check its markings.
3. A second-hand phone. Open the network settings on the device: the address mode must be DHCP, not Static, and VLAN must be off. For example, the D-Link DPH-400S ships with a static 192.168.1.10 — see the D-Link page. The shortest route is a factory reset.
4. Phones in a separate VLAN need DHCP in that VLAN; that is a question for whoever set up the network. More in “Network, router and internet”.

DECT handset cannot find the base or crackles#

Why it happens#

DECT has two legs. Handset ↔ base is radio, and our PBX has nothing to do with it. Base ↔ PBX is ordinary SIP, just like a desk phone. A message like “Unregistered!” on a Yealink handset is about radio (the handset is not paired with the base), not about the PBX. Crackling and drop-outs in a far corner are the edge of radio coverage, not a bad internet connection.

What to do#

1. The handset is not paired with the base — pair it: Yealink W — “Pairing the handsets with the base”; Grandstream DP — hold the Radio/Page key on the base for about 8 seconds, then Subscribe on the handset (Step 1 of the DP page).
2. The handset is paired but does not ring — no account is assigned to it: Yealink — Account → Number Assignment, Gigaset — Settings → SIP accounts → Assignments, Grandstream — Account Assignment / Handset Line Settings.
3. The handset will not pair although the base is in registration mode — the base is full: Yealink W70B takes up to 10 handsets, W60B up to 8, Grandstream DP750/DP752 up to 5.
4. It crackles and drops away from the base — dial *43 and, without hanging up, walk around the dining room, kitchen, storeroom and stairs: where your own voice starts breaking up is the edge of coverage. Mount the base higher and away from metal and thick walls; if that is not enough, add a repeater (on Yealink: OK → Settings → System Settings → Repeater Mode) or a second base. The rated indoor range is modest: up to 50 m for the Grandstream DP750/DP752.
5. The handset was bought in another country — European DECT (1880–1900 MHz) and American DECT 6.0 (1920–1930 MHz) use different frequencies, and such a handset will not see your base. On Gigaset the regional radio band is chosen at the first login to the base — make sure it is yours.

What to send to support#

If this page and the detailed symptom page did not help, send us:

• the date and time of the problem call to the minute, the calling number and the extension;
• the model and firmware — copy the string from the equipment tag in the admin panel;
• what happened on *43: beep, your own voice, choppy — and the test result from the log or the device card;
• a screenshot of the call card from the log and, if you captured one, the SIP trace text (passwords are stripped from it automatically);
• how the phone is connected: cable or Wi-Fi, the router model, whether SIP ALG is off;
• what you have already tried from this page.

How to reach us — “Support”. If it is easier, turn on “Support access” in settings and our staff will take a look themselves.

Frequently asked questions#

The dot is green — does that mean the phone definitely works?#

No. The green dot is registration only: the phone told the PBX it is there. Voice travels a separate road and breaks separately, and the way back to the phone may close a minute after registration. To check that it “works”, use *43 (voice) and “Call this phone” or a call from the extension next to it (incoming).

Why does the admin panel not show why the phone fails to register?#

The PBX keeps a history of successful registrations, while a rejection is an answer the phone received, and it shows first of all on the phone’s screen. On a dedicated PBX the SIP trace shows the reason: start a capture and reboot the phone. What each message means — “Not registering”.

Rebooting the phone helps — is that normal?#

A reboot re-registers the phone and re-opens the “doors” on the router — that is why it cures the symptoms of the way back. If it helps for an hour or two and then everything repeats, the cause is keep-alive or the router: see “Incoming calls only work for the first minutes”.

Can I test the line without bothering guests or spending money?#

Yes: *43 is an internal number of your PBX, it never calls outside and costs nothing. Dial it as often as you like, even at peak hour.

Related sections#

• SIP Phone Setup — credentials, venue network, checking call quality, factory reset
• Not registering: on-screen messages and error codes
• No audio, one-way audio, echo, “robot” voice
• Calls drop: after 30 seconds, after 15–30 minutes
• Network, router and internet for telephony
• Telephony: call log and SIP trace
• Brand guides, the “If … does not register” section: Yealink T, Grandstream GRP, Fanvil, Yealink W (DECT)

Sources#

• RFC 3261 — SIP: registration, response codes — https://www.rfc-editor.org/rfc/rfc3261
• RFC 4733 (replaced RFC 2833) — sending key presses as separate packets — https://www.rfc-editor.org/rfc/rfc4733
• Yealink SIP-T2/T3/T4/T5/CP920 Administrator Guide V86.60 — Accept SIP Trust Server Only (sip.trust_ctrl, protection against calls from numbers 100, 1000), DTMF Type, Keep Alive Type / Interval (factory 30 s)
• Grandstream GRP261x/262x/263x Administration Guide — Accept Incoming SIP from Proxy Only — https://documentation.grandstream.com/knowledge-base/grp261x-grp262x-grp263x-series-administration-guide/
• Grandstream HT80x Administration Guide — Allow Incoming SIP Messages from SIP Proxy Only — https://documentation.grandstream.com/knowledge-base/ht80x-administration-guide/
• PBX code: sip-edge/sipsync/linecheck.go (the *43 echo test, # ends the test with a second beep), shared/sip/advanced-options.json (DTMF RFC 4733 by default, up to 5 devices per account with the oldest pushed out, device check every 60 s), sip-edge/sipsync/render.go
• Admin panel screens: admin/components/integrations/SipStatusDot.tsx, SipDeviceModal.tsx, SipDeviceLineCheck.tsx