Telephony: your own PBX
In most venues the phone lives apart from everything else: the number is at the carrier, the handset is on the counter, and who called is remembered only by whoever picked up. The Telephony section turns that into a working tool: the location gets its own PBX — a phone exchange that answers calls to your number, distributes them to the team by your rules, records conversations and hands them over for analysis.
Open: Admin panel → Telephony (admin.meni.ge/telephony). The section is switched on by the Telephony card in the app store — it is hidden by default.
What "your own PBX" means#
The PBX is dedicated to your location. It is a separate cloud machine with its own static IP address and its own connection address: nobody else's voice passes through it, the carrier accounts are yours, and we neither issue nor resell numbering. A single shared server for everyone is no longer offered to new locations at all.
| Item | How it works |
|---|---|
| Ordering the server | a button on the "Server" tab. The service is paid and charged to the account balance per day — the daily price is shown in the section header and next to each server size, and the confirmation before creation says so plainly. Daily charges appear as a collapsed block right in the server card (the last 7 days) and as a log for all locations in Subscription |
| Capacity | a ladder of presets labelled with the number of simultaneous conversations; a capacity change happens with no downtime, and you can wait until no calls are active |
| Load | CPU and network charts for 1 hour / 24 hours / 7 days right in the server card |
| Settings | save themselves; there is no "Save" button in the section. Every block has a "clock" icon — the history of its settings, with rollback |
| Where it physically runs | the cloud is chosen by the country of the location; Russian locations are served only in Yandex Cloud (cenaly.ru) |
At the top of the section there is the "Before the first call" checklist with three separate questions. Settings — four steps (create the location's SIP account → connect an inbound line → set up the incoming call path → issue at least one internal number); the "3 of 4" count is about them only, and every unfinished step is a link to the right tab. Test call — taken from the call log: the last answered inbound call; the *43 echo test checks sound but not the provider line, so it is shown as a hint next to it. Call recording — your choice (off / on with a notice to callers); it is not part of "all set": you do not have to record just to get a green tick. Once the settings are ready and the call went through, the checklist collapses into a single green line.
Where to start#
The order below is also the path of a call — from the outside in:
| # | Step | Where | In detail |
|---|---|---|---|
| 1 | Get a number: a provider's SIP trunk or a gateway with a SIM card | Phone providers → Lines | provider's number, GSM gateway |
| 2 | Issue extensions to the team | Operators | below |
| 3 | Connect phones and softphones | credentials from the extension window | setting up SIP phones |
| 4 | Set up inbound routing | Routing, Ring groups | below |
| 5 | Switch on outbound calls if the team calls out | Outbound | below |
| 6 | Switch on recording | Recording | below |
Step 1. The number: provider or gateway#
The inbound number is yours, under your own contract with a carrier. There are three sources, and they add up:
| Source | How it connects | What matters |
|---|---|---|
| A provider's SIP trunk | the "Phone providers" tab: name, SIP host, port, transport, mode (REGISTER — the PBX registers with the carrier, or IP auth — the carrier sends calls to our IP), username and password. The "Check connection" button runs a live probe | the number itself is added as a line on the "Lines" tab, and the line has a working "Accepts calls" switch. Full guide — connecting a provider's number |
| A gateway with a SIM card, a landline or ISDN | the gateway registers on our PBX as an extension and dials an internal number itself when a call arrives | the gateway does pass the caller's number on — it shows on the handset, in the log and in the "Calls" recording. Such a call has no line, so routing rules do not apply to it. Full guide — GSM gateway and landline |
| The built-in web-call line | nothing to connect: the guest speaks from the browser | one line, no number needed — calls from your website |
The number of trunks and lines is not capped — it is your machine and your carrier accounts; the overall limit of the PBX settings list is 500 entries.
Step 2. Extensions for the team#
The Staff & groups tab holds internal numbers. There are two kinds: our extension (an account on our PBX that a phone or softphone connects to; you choose the number yourself — 2–6 digits, your own venue numbering, with no cap on how many you create) and an inherited carrier number — behind it there is a number the person already has: an extension on someone else's PBX, a mobile or a landline. No password is issued for it, and the call reaches the person through the chosen channel.
| Feature | What it does |
|---|---|
| Assign to an employee | in the "Staff member" column click the name (or "Not assigned") and pick a person — the staff list opens on click instead of sitting in every row. The number shows up in the employee card, and dismissal revokes the extension automatically across all locations |
| "Hand out to the team…" | issues numbers in bulk filtered by job title; whoever already has one is skipped |
| Credentials | the password is shown once: copyable fields, "copy all settings" as text and a QR code for Zoiper and Linphone. Lost it — "Reset password" |
| Full number duplication | a number that lives at the carrier is set up so that the person's phone registers with us, while the call still reaches their old handset at the provider |
| Is the channel alive | for an inherited number the “Channels and SIP” column shows a connection dot next to the channel name: green — the trunk used to reach that person is registered with the provider; grey — there is no registration, or the PBX has not reported one yet; a “stop” square — you turned the trunk off. Hover the dot to read which trunk it is, what is going on with it and when that was observed. Our own extensions get no extra dot — their connection is shown by the dot at the left of the same row |
| Equipment and logs | the row shows what the employee connects with ("Yealink T31G", "Zoiper"), greyed out if the device is off. Next to the name there is a signal-bars icon — the PBX ping to that device (green — the link is fine, amber — a noticeable delay from 150 ms, red — from 300 ms, ✕ — the device stopped answering the checks). The figure ("180 ms") shows up next to the icon only when the delay is noticeable; the full numbers live in the hover tooltip and in the device card. Inside the card the ping sits in the state line ("online · address · network · last seen"), so the figure is always there and visible in every section of the card, while the hour, the 24 hours and the hourly bars live on the "Uptime" tab, in the "Connection to the PBX" block. A device that is switched off has no icon: the PBX does not check it. A click opens the device card: uptime over 24 hours / 7 / 30 days, connection history, calls, the line check and settings — the server, port, transport, username and extension of this phone (what you type into the device itself, copyable) plus the advanced SIP settings the PBX applies to it. No password there: it is shown once when issued |
Step 3. Phones and softphones#
The same five values (server, username, password, port 5060, UDP transport) go into any device: a desk phone, a waiter's DECT handset, a door intercom, an adapter for an old analogue phone, or a softphone on a computer or a smartphone. One extension can be registered on several devices at once.
The full guide is setting up SIP phones: where to get the credentials, what the venue network must allow (outbound UDP 5060 and 10000–20000, SIP ALG switched off on the router), when a phone has to be factory reset, and step-by-step instructions for 26 device families. It also covers the *43 echo test: you dial it, you hear your own voice, and the call log gets packet loss, jitter and latency for that exact handset.
Everyone adds their own personal accounts themselves: Settings → "Personal SIP accounts" (admin.meni.ge/settings?tab=sip-accounts) — these are a person's registrations on other systems (at a carrier, on an office PBX) or with their own extension. They are tied to that person's login, not to the company, only they see them, and they go only to the phone in the Cenaly Android apps on that person's devices — Cenaly Admin ("Phone" tab), Cenaly CRM ("More" → "Phone") and Cenaly SIP Phone. They never reach the location's PBX, desk phones or the web admin (see settings).
Step 4. Inbound routing#
The Incoming call path tab holds the rules of "who to ring when a call arrives". Rules are checked top to bottom and the first match wins; their order is the priority, and you reorder them with arrows.
| Rule axis | Options |
|---|---|
| Which number was called | chips grouped into "Lines (numbers)" and "Providers (trunks)"; several targets can be selected — any one of them matches |
| When | days of the week and a time window in the location's time zone |
| Who is calling | caller number prefix |
| Who answers | everyone who has an extension · selected employees as name chips · a ring group · an AI assistant by name (phone consultant, receptionist, any voice bot) · an external number |
| If nobody answered | a no-answer message, voicemail, an AI assistant, another group |
The ring plan grid is a table where a row is a participant and a column is 5 seconds: "for the first 10 seconds the host rings, from the 10th the manager joins, from the 20th the host goes quiet". Cells are painted by dragging the mouse. The last row is the AI: it does not ring, it answers, so it starts where the last human stopped ringing.
Below the list of rules is the default answer: what happens when no rule matched (a greeting, voicemail, a bot). The same block holds the "this call is being processed" notice with a language choice and your own text — in a number of countries such a warning is mandatory, so it is on by default.
If you delete a provider that rules point at, the panel asks first and tells you what will happen to each rule: "keeps working without this provider" (the condition still has other numbers or providers), "will be turned off" (without the provider the condition would become wider) or "will be turned off and left with no condition". Rules are not deleted — the ring plan grid and the fallback stay in place. A rule left with no condition at all is marked "no condition", and its switch will not turn it on: otherwise it would catch every call. Give it a new condition (a number, a provider, hours or a caller prefix) and turn it on, or delete the rule.
Ring groups (rows in "Staff & groups" marked "Group") are internal numbers (issued from 700) with their own distribution: a ring plan grid, a plain "all at once / one after another" hunt, or a queue with hold music, position announcements and a waiting cap. A member has a "Rings" toggle, personal working hours, a pause (the employee sets it from the handset: *71 — away, *72 — back) and their own hold music; the "Ring the person the guest already spoke to first" toggle tries the familiar employee for a few seconds. Three doors lead into a group: dialling its number from a phone, a routing rule, and a transfer by the AI receptionist. Next to them are conference rooms (the "Conferences" tab): extensions 600–649 with a PIN and video — dial the number from any registered device and you are in the shared conversation.
Step 5. Outbound calls#
Outbound calls are off by default, and switching them on is confirmed with plain wording about money: the call goes out over your trunk and the bill comes from your carrier. The "Outbound" tab decides which line a call dialled by an employee leaves from — three levels, in the order they are resolved:
- Smart rules "dialled-number pattern → line": digits plus
X(any digit),N(2–9),Z(1–9) and a trailing dot for "any number of digits". A rule can be narrowed to specific employees, strip or prepend digits before sending them to the carrier, and override the number shown to the callee; - the employee's personal line — "one each";
- the shared default line.
Anti-fraud is always on: a cap on simultaneous outbound calls per location and per employee, limits on dialled length, an unconditional block on premium and satellite destinations (no setting removes it) and a daily cap on talk minutes (no cap by default). At 80 % of the cap, at its exhaustion and on suspicious night traffic the PBX posts an alert with a push into the work chat — a hijacked account is recognised by minutes burnt overnight, not by the carrier's invoice at the end of the month.
Step 6. Call recording and "Calls"#
The Recording tab: switch it on, choose the scope (all calls / inbound only / operator calls only) and decide whether recordings older than N days are deleted. By default recordings are kept with no limit — you set the period yourself, and there is no upper bound.
A recorded conversation lands not only in the telephony player but also in the Calls section (admin.meni.ge/calls) — the hub for recordings: uploading and receiving files, a table with filters and saved views, speech recognition, AI analysis of the conversation, tags, metrics, rules and automations, search and linking to an order. Retention for the recording store and for the "Calls" section is configured separately. A detailed article about "Calls" is being prepared.
The AI receptionist, agent assist and phone consultant#
Telephony is the infrastructure three AI surfaces run on. They are configured in their own sections, while the PBX only decides "who answers the call". Detailed articles on the receptionist and the agent assist are being prepared.
| Surface | What it does | Where |
|---|---|---|
| AI receptionist | a robot answers inbound calls on behalf of the venue: persona and tone, schedule, questionnaires (reservation, message, request), rules for transferring to the team, permitted actions, anti-spam and a text simulator. It answers from the knowledge base, and the "you are speaking to an AI" disclosure cannot be switched off where it is mandatory | admin.meni.ge/secretary |
| Agent assist | during a conversation the employee gets answer cards from the knowledge base on top of a live transcript; no language model is called during the call — a pre-compiled pack does the work | admin.meni.ge/call-assist |
| Phone consultant | an AI assistant as an ordinary row in the extension list: you can give it an internal number, put it into the "if nobody answered" rule or into the last row of the ring plan grid | admin.meni.ge/telephony?tab=extensions |
Calls from your website#
Two free widgets run on the same PBX and arrive through the same inbound rules: calls from your website — a handset button, the guest speaks straight from the browser with no number to dial and no app to install; callback — the guest leaves a number, the PBX dials it and connects them to the venue, with retries on no-answer and a night queue with a morning round. You only pay for your own telephony — the widgets themselves are free.
Tabs of the Telephony section#
The tabs are grouped into four entries above them: Today (Overview), Work (Live board, Call log, Voicemail, Conferences), Setup (Lines, Phone providers, Incoming call path, Staff & groups, Outbound, Recording) and Diagnostics (Server). This is grouping only: tab addresses and sidebar pins stay the same. The "All features" button shows every tab at once, and the "Find a feature" field finds a setting by old words too ("trunk", "extension", "IVR"). The section opens on the entry for your role: the owner — "Today", an operator — "Work", a setup person — "Setup"; you can pin your own choice in the same place. Above the tabs is a status line: whether a change is saved, whether the PBX has applied it, and which settings wait for a PBX update.
| Tab | What is there |
|---|---|
| Overview | the "Before the first call" checklist, the "How an incoming call goes" block (a forecast from the saved settings with "Change" and "Check a number" buttons, plus the last real call), the location's SIP account summary, the "Check connection" button, the outage e-mail toggle, the PBX connection history |
| Live board | who is talking to whom right now: calls in progress, extensions, trunks; the "Console / Board" switch shows the call-centre figures large (waiting for an answer — in a queue or while the group's phones are still ringing; available; today's service level), agents by status and every call in progress as a list — who is calling, who they are talking to or whose phone is ringing, how long they have waited and how long the call has lasted; "Full screen (TV)" is for a wall screen |
| Call log | the history of every PBX call with statuses and the routing trace — see below |
| Voicemail | callers' messages by box: listen, assign, mark as handled, call back; a manager gets "Greeting and box settings" right from the message and links to the call in the log and to the "Call back" case |
| Lines | inbound numbers, the "Accepts calls" switch, the default hold music (the "Listen & choose" button plays the whole catalogue without changing anything) and the built-in web-call line |
| Phone providers | provider trunks as one row each (state, registration, check), editing credentials in place, the "Full number duplication" subsection, the "Connect a number or line" button (the wizard first asks: a number from a provider, another PBX, or a line through a gateway) and the catalogue of pairings with providers, external PBXs and gateways |
| Incoming call path | inbound rules top to bottom, the call path diagram with a number check, and the default answer "if no rule matched" — the same editor as in the "Phone consultant" card of the "Calls" section |
| Outbound | "number → line" rules, personal and shared lines, the master switch and the limits |
| Staff & groups | one list of people, groups and AI assistants ("Person / Group / AI" badge, "Ext."); an employee card opens on the "Overview" tab — how they take calls and a "Connect a phone" or "Check" button; below, as collapsed rows — phone equipment, desk phones, phone book, parking and intercom, voicemail boxes, links between locations |
| Conferences | rooms 600–649: name, PIN, video, participant cap |
| Recording | switch, recording scope, retention period |
| Server | the PBX machine and the bridge on a PC, capacity and load, advanced SIP settings, trunk diagnostics (a capture of the SIP exchange) |
"Advanced SIP settings" explain themselves. The protocol has three dozen compatibility parameters, and a phone that "will not connect" usually runs into exactly one of them. So every setting in that window carries a "More about this setting" disclosure with three parts: "When you need it" — whether you should touch it at all ("keep it on while the phone sits behind a router", "only at the request of the provider"); "What changes" — what happens if you turn it on or off, and what it costs you ("call recording and live prompts disappear: the PBX no longer hears that audio"); "How to check" — a concrete action and its expected outcome ("dial *43 from the phone — the echo test must play your voice back", "press Call this phone — it must ring", "open PBX connection history"). An empty field means the default value, written next to it; settings that can break calls outright (direct media, the protocol set, caller identification, encryption) are marked with an icon. If your telephony works, there is nothing here you need to touch.
Call log: every call, not only the recorded ones#
The Call log tab (?tab=log) shows the history of every call on your own PBX, including those with no recording: answered, missed, busy, abandoned, answered by a robot. A row shows the time, who called, which line they reached, whose phone rang (extension chips), who picked up, the status in words and the duration.
A click opens the "how the call went" modal: the routing rule that matched, every dialling attempt with the fate of each target ("picked up after 5 s", "rang 29 s, no answer", "busy", "rejected"), the fallback and the final phrase. A transfer to a colleague is an event of the same call, not a new one: the trace gets a "transfer" step and the conversation is split into segments ("guest ↔ ext. 101 Anna — 0:16", "guest ↔ ext. 102 Boris — 0:20"). Audio and AI analysis are on the "Recordings" tab of the Calls section: the "Open recording" button in the call card takes you straight there.
Wire diagnostics (SIP trace)#
When "the call does not go through" while every setting looks right, arguing is pointless — you need to see what our PBX and the carrier actually said to each other. That is what the "Wire diagnostics (SIP trace)" block on the "Server" tab is for.
SIP is the signalling that sets a call up: REGISTER (a phone or a trunk announcing itself), INVITE (someone is calling), response codes (200 OK, 401, 403, 488…) and the SDP block inside, which names the codecs and the addresses for audio. The trace is a transcript of that exchange as the server sees it.
How to capture one:
- Telephony → "Server" → the "Wire diagnostics" block. "Record for 15 minutes" opens a recording window (an "open until …" badge appears next to it; the cap is 2 hours).
- While the window is open, place that very call — the trace only shows what happened while the window was open.
- "Refresh" pulls in the latest tail of the exchange; the lines appear in the dark box below the buttons. "The PBX has not published the exchange yet" means "give it a minute" — the machine publishes the tail on its own cycle, it is not an error.
- "Stop now" closes the window early; it closes by itself in any case — the exchange contains caller numbers, that is personal data, so "switch it on and forget" is deliberately impossible here.
Worth knowing about the contents: the values of the authorization headers (Authorization, WWW-Authenticate) are stripped from the lines — there is no password in the trace; what you see is the tail of the exchange, and if the beginning did not fit, it says so right above the text. On a dedicated PBX the exchange is, in addition, recorded continuously and kept on the machine for 24 hours, so a complaint like "an hour ago the call had no audio" can still be looked into after the fact — there is no interval picker in the panel yet, so support retrieves the slice for the hour in question.
How to read the typical pictures:
| What the exchange shows | What it means |
|---|---|
REGISTER → 401/407, then a repeated REGISTER → 200 OK |
a normal two-step registration (digest challenge) — this is how it should look |
REGISTER → 403 Forbidden (or 404) |
the carrier refused: wrong username, password or domain — and with IP authorization, the call came from the wrong address |
no response at all, 408 Request Timeout |
packets are not arriving: wrong port or transport, a filter on your network or the carrier's side, or with IP auth our address is not whitelisted |
INVITE → 488 Not Acceptable Here (or 606) |
the codecs did not match: our PBX speaks G.711 A-law, µ-law and G.722, and does not support G.729, Opus or AMR |
INVITE → 200 OK, but there is no audio or only one side is heard |
signalling went through, the problem is the audio stream (RTP): SIP ALG on the router, UDP 10000–20000 closed, or NAT — the SDP block (c=, m=audio) shows which address each side announced |
INVITE → 480/486/603 |
the far end is unavailable, busy or declined — an answer from the other side, not a fault |
INVITE → 503 Service Unavailable |
the carrier refused: channels, balance or the destination on their side |
BYE right after the answer |
one of the sides is tearing the session down — usually session timers or keep-alive on an intermediate node |
a pause between INVITE and the first response |
you can see at which step the PBX is waiting for the carrier — the very "silence before an outbound call" |
What to send to support. There is no file export in the panel: select the text in the trace box and attach it to your message (or a screenshot), naming the time of the call and the number — that is enough for a diagnosis. If you prefer, switch on "Support access" in settings and an agent will look at the exchange themselves; how to write to support — here.
⚠️ Diagnostics are available on a dedicated PBX only: the Asterisk logger command is global, so on a shared server it would capture other tenants' exchanges. On the shared tier switching it on is refused.
API & MCP: CRM, your own programs and AI assistants#
Your PBX can be managed from outside the panel too. The "API & MCP" button in the Telephony header (shown to the account owner once the PBX is connected) issues keys for outside programs: a CRM, your own reporting script, or an AI assistant such as Claude or Cursor.
Issuing a key. "Create key" → a name ("CRM", "Claude") → a set of rights: "CRM" (call log, recordings, click-to-call, extensions), "AI assistant" (the same plus forwarding, do-not-disturb and who is free right now), "Read only" or "Everything" — or pick rights yourself. Three rights are granted only as a separate checkbox and are not part of any set, not even "Everything": "Change call rules", "Export the configuration" and "Load the configuration from tables". Then choose locations (all or selected) and validity (until revoked, 30, 90 or 365 days). The cpx_… key is shown once: copy it right away, we only store its fingerprint. Rights can be changed at any time; "Revoke" disables the key from the next request.
What the API can do. The call log with filters (missed, by extension, by number, by period) and a call card with its routing; a recording link and the transcript with its summary from Calls; click-to-call from a chosen extension's phone (it rings first, then the number is dialed); extensions and groups, personal forwarding and do-not-disturb; who is free and who is talking; numbers and inbound/outbound rules with their change history. With "Change call rules" a key creates, changes, deletes and reorders rules one at a time (only the fields you pass change — neighbouring rules stay untouched), turns outside calls on or off and rolls rules back to an earlier version. Distribution of a call to employees (schedule grid, queue, "everyone") is set up in the panel — an API rule sends the call to a group by its number; allowed countries and anti-fraud limits are not changed by keys. With the configuration rights — export and load of the same tables as the Excel/CSV file: a preview of what will be added / changed / removed first, then apply. Passwords and logins are never returned by the API — passwords of new extensions are shown only in the panel.
Connecting an AI assistant. The same key works for the MCP server https://api.cenaly.com/pbx-api/mcp (streamable HTTP). The "How to connect" tab has ready-made configs for Claude Code, Claude Desktop and Cursor — paste the key and the assistant can answer "who called today and didn't get through", "call this guest back from Anna's phone" or "forward my calls to my mobile". The assistant only sees the tools the key allows.
ChatGPT and claude.ai — no key. Cloud assistants connect the same MCP server with OAuth: add https://api.meni.ge/pbx-api/mcp as a custom connector, sign in to Cenaly and tick what the app may do — the window shows which app is asking. The connection appears in the key list with an OAuth badge and is revoked there. Details, examples and limits — in Your PBX API and MCP.
For developers. REST lives at https://api.cenaly.com/pbx-api/v1, key in the Authorization: Bearer <key> header; the OpenAPI 3.1 description and the reference page are linked in the window (/pbx-api/v1/openapi.json, /pbx-api/v1/docs). Limits per key: 120 requests per minute and 20,000 per day; click-to-call — 10 per minute and 300 per day. Call events (each recorded call with transcript and analysis) arrive by webhook — the "Webhook" source in Calls. Every change made with a key appears in the admin action log under the key's name.
Limitations#
- Outbound calls through a gateway are not there yet. A gateway with a SIM card or a landline brings calls INTO the PBX; you cannot dial out through it from the panel — an outbound rule can only point to a line or a provider trunk. And routing rules do not apply to calls from a gateway: such a call has no line, so the gateway's own settings must point straight at a group or an employee number.
- There is no encryption for provider trunks: a trunk works over UDP or TCP, and TLS and SRTP are not available for it. For phones, encryption is enabled by an advanced setting of the dedicated machine (which issues the certificate itself) — that part has not yet been verified on a live fleet of handsets.
- Some features only exist on the dedicated machine: ring groups and the queue, conference rooms, call recording, two-sided ringing for a duplicated number, and the SIP exchange capture. A PBX raised on your own PC through the bridge, and the remnants of the shared server, do not execute them; outbound calls and routing rules are likewise assembled by whoever holds the trunks.
- Codecs — G.711 A-law, µ-law and G.722; G.729, Opus and AMR are not supported. There is no phone auto-provisioning by MAC address — devices are configured with the fields from the extension window, or by your own provisioning server.
- Countries where the PBX is unavailable (a local licence is required or sanctions apply): Türkiye, Belarus, China, Egypt, the UAE, Qatar, Oman, Kuwait, Iran, North Korea, Cuba, Syria. In Russia a dedicated PBX is allowed and runs only in Yandex Cloud — on
cenaly.ru, where the shared server is closed. - The synthesiser voice depends on the cloud: on
cenaly.ruthe notice language list only offers what the local synthesiser can speak (six languages). Queue positions are announced in Russian and English — in other languages the queue works but does not read out the position. - Conferences v1: a room is entered from the inside only (an external call cannot be transferred into it), video shows the active speaker and requires the VP8 codec; there is no roster, no moderation and no conference recording.
- We do not sell numbers. You buy the trunk and the number from a carrier yourself — a number cannot be moved "inside" the platform, it stays with the provider.
Troubleshooting#
| Symptom | Check |
|---|---|
| The phone does not register | the username is entered in full (sip-…-<number>, not the number alone), the password has not been reset, the server address is taken from the window, transport is UDP. Full list — setting up SIP phones |
| The call connects but there is no audio, or only one side is heard | dial *43 from that handset: a beep without your own voice means SIP ALG on the router, complete silence means UDP 10000–20000 is closed |
| Inbound calls do not arrive | "Overview" → "Check connection": it walks the chain server → line → route → extensions and names the first broken link. Check the "Accepts calls" switch on the number and the trunk registration |
| An outbound call is rejected as "forbidden" | whether the outbound master switch is on, whether the daily minute cap is exhausted, whether the destination is a premium one |
| A call is rejected for no clear reason, the trunk keeps flapping, there is a long silence before it connects | open wire diagnostics on the "Server" tab and place the call again: the exchange shows which side answers and with which code |
| Calls stopped coming through and nobody noticed | the "E-mail me when calls stop coming through" toggle on "Overview" is on by default: the letter arrives if the fault lasted 10 minutes |
The general symptom-by-symptom guide is troubleshooting; if none of it helped, write to us — how to get help.
FAQ#
Do I need an external number to start?#
No. Without a number the PBX already takes calls from your website and from the app, and the team calls each other by internal numbers. A number is needed when guests have to reach you from an ordinary phone.
Can calls be taken on an employee's mobile?#
Yes, in two ways. An inherited number — the internal number lives with us while the call goes out to their mobile through the chosen channel (they get no account of ours). Full number duplication — the employee's phone registers with us, and then it takes part in groups, the ring plan grid and the rules on equal terms with everyone else.
How much does telephony cost?#
The PBX itself is a paid service: the dedicated server is charged to the account balance per day (a full day of work costs the package price, a partial day — in proportion to the time worked; days follow UTC), and the confirmation before creation says so. What was charged for which day is shown in the server card ("Daily charges") and in Subscription. Conversations are paid to your carrier at their tariff — we are not an intermediary in that chain. The "Calls from your website" and "Callback" widgets are free.
Related articles#
- Connecting a provider's number — a SIP trunk: registration or IP authorisation
- GSM gateway, landline or ISDN — a SIM card and an analogue line in your PBX
- Setting up SIP phones — extension credentials, the network, the
*43echo test, per-brand instructions - Hardware: what connects — phones, DECT bases and gateways in the overall picture of the location's hardware
- Calls from your website and Callback — guest calls without a number
- Employees and shifts — who gets an extension
- Settings — the "Personal SIP accounts" tab and other account settings
- Needs attention — missed calls and telephony faults in the shared to-do list