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📞 Connecting a Provider Number

📞 Overview: connecting a provider number to the PBX

Two SIP trunk modes (registration and IP auth), what to ask the provider, what to enter in the admin panel, network and compatibility, verification, typical errors and the country list

Documentation

Connecting a provider number to the PBX

Your own PBX in the admin panel can receive calls on your landline or mobile number as long as that number comes from a SIP provider — a telecom operator or a virtual telephony service that offers a "SIP trunk" (sometimes called "external PBX connection" or "SIP account for your own PBX"). The number stays with the provider, and calls to it arrive in your PBX: to staff on desk phones and in the app, to the voice bot, or to a greeting with a menu; conversations are recorded and appear in "Calls".

This page covers what is the same for any provider: the two ways to connect, which data you need, what to enter where, what the network must allow and how to tell that everything works. The specifics of individual operators are in the country guides at the end of the page.

Open: Admin panel → Telephony → "Trunks & pairings" tab (admin.cenaly.ru/telephony?tab=trunks).


When you need this and what is required#

You connect a provider when the venue has (or should get) an external number guests call, and you want to receive those calls in the PBX: on several phones at once, with a queue, recording and call analysis. If calls from the website and the app are enough for you, an external number is not required.

You will need:

  1. A dedicated PBX for the location — the "Server" tab in the Telephony section. Provider trunks only work on a dedicated server: it has a static IP address you can give to the provider, and its own Asterisk configuration. The shared server is no longer offered to new locations.
  2. A contract with the provider and the SIP trunk service (or a SIP account that allows connecting your own PBX). It is usually a separate tariff item; some operators offer it only to legal entities or only together with their own cloud PBX (such a provider will not work).
  3. Trunk credentials from the provider's account — see "What to ask the provider".
Countries where the PBX is unavailable

A dedicated PBX is not created for locations in countries where receiving calls on your own switch requires a local telecom licence or is prohibited by the regulator: Turkey, Belarus, China, Egypt, the UAE, Qatar, Oman, Kuwait, and sanctioned countries. For locations in Russia the server is created in a Russian cloud (Yandex Cloud), as required by the law on storing call recordings. Guides for these countries are still in the list below: they will be useful once launching becomes possible.


Two trunk modes: REGISTER and IP auth#

The provider must somehow understand that a call to your number should be delivered to your PBX, and that a call from your PBX should be accepted and released into the network. There are two ways, and they are selected with one checkbox in the trunk form.

Registration (REGISTER) By IP (IP auth, no registration)
How it works The PBX "logs in" to the provider's server with a login and password, just like a phone. The provider sends calls to wherever the registration came from You tell the provider the static IP of your PBX. It sends calls straight to it and accepts calls from it without a login
What you need from the provider SIP host, login, password SIP host (where calls will come from) and the ability to specify your IP in the account or via support
Checkbox in the form ✅ "Register at provider (REGISTER)" ⬜ unchecked — "IP-auth (no registration)"
Who works this way Most virtual PBX providers and international number services (Zadarma, Telnyx, Verimor, sipgate…) Telecom operators and "grown-up" business trunks (Twilio Elastic SIP Trunking, MTS Exolve, Netgsm…)
Specifics The password is stored in the PBX; if you change it at the provider, the trunk drops Requires a static IP (a dedicated PBX has one); if the provider also requires TLS, it will not work

If the provider supports both modes, choose registration: it does not depend on allowlists and survives a server resize, during which the IP is kept but the provider may check both the address and the login.


What to ask the provider#

Before entering anything, collect from the provider (in the account, in the contract or from support) the answers to seven questions. Wording differs between accounts, so the usual names are listed alongside.

What you need What the provider calls it Why
SIP server SIP host, Registrar, Proxy, "Server address", "Domain" (sip.example.com, <account>.sip.example.com) The SIP host field
Port Port (almost always 5060) The Port field
Transport Transport: UDP / TCP / TLS The Transport field. The PBX supports UDP and TCP; a "TLS only" provider will not work
Login and password SIP login / username / Auth ID, password The Login and Password fields (for registration)
Authentication mode "Registration" or "IP authentication / IP whitelist" The REGISTER checkbox
Codecs Codecs: G.711 a-law/u-law (alaw/ulaw), G.722, G.729 The PBX speaks G.711 and G.722; if the provider offers only G.729 there will be no audio
How the number arrives "Number in Request-URI / DID", format: +995322…, 0322… or the login So that the number in "Lines" matches what the provider sends

It is also useful to learn how many simultaneous calls (channels) the tariff includes, whether the provider allows substituting another number on outgoing calls (usually only the numbers of your contract), and whether it requires an IP allowlist for outgoing calls even with registration (it happens).


Step 1. Add the provider in the admin panel#

  1. Open Telephony → "Trunks & pairings" and press "Add provider". If your operator is in the catalog (Zadarma, Telnyx, Twilio, MANGO OFFICE, Telfin, Sipuni, MTS Exolve, Netgsm, Verimor, FCN…), pick its card — the server address and the mode are filled in for you. For any other, use the "SIP trunk (your own provider)" card.
  2. Fill in the form:
Field What to enter
Name How you want to see the trunk in lists: "Magticom — landline", "Zadarma +995"
SIP host The provider's server address without sip: and without the port: sip.zadarma.com, pbx.example.ge
Port From the provider's account, usually 5060
Transport UDP, unless the provider insists on TCP
Register at provider (REGISTER) On — for registration mode; off — for IP authentication
Login The SIP login from the provider's account (mandatory for registration)
Password The SIP password. An empty field on re-saving means "keep the password"
  1. Press "Save". The trunk appears in the list with a registration status line.
IP mode: what to give the provider

Open the "Server" tab and copy the "Static IP" value — that is the address of your PBX. In the provider's account (or via its support) specify this address as the delivery point for your numbers and, if asked, as the allowed address for outgoing calls; port 5060, transport UDP. The provider may have its own list of addresses it calls from: in the SIP host field enter the host named in its instructions — the PBX uses it to recognise incoming calls as "from this provider".


Step 2. Test the connection#

In the trunk row press "Test connection". The wizard sends a real SIP request (OPTIONS) to the provider's server from the cloud and shows the reply: "server responded" means the address and port are correct and the path to the provider is open. This checks the address, not the login: whether the password is right is visible in the registration line.

Registration line states:

  • "registered at provider" — the PBX has logged in to the provider's server; incoming calls should work.
  • "no registration at provider" — the provider rejected the login or password, or does not allow this address. Check the credentials in the provider's account; after a password change the PBX re-registers on its own, incoming calls pause for a few seconds.
  • "registration not needed (IP-auth)" — IP mode; check that the provider sees your address: make a test call to the number.
  • "the PBX has not reported registration yet" — the server was just created or is restarting; wait a minute.

Step 3. Add the number and routing#

  1. Open the "Lines" tab and add a line with the source "Provider trunk (DID)", choosing the trunk you just created. Enter the number as the provider issued it: the PBX compares the last seven digits of the number with what arrives in the call, so the format (+995 32 2… or 0322…) does not matter.
  2. Leave the "Accepts calls" checkbox on. If you uncheck it on all numbers of a trunk, the PBX withdraws the registration at the provider and stops recognising its address — there will be no incoming calls at all.
  3. In the "Routing" tab add a rule "call arrived on line … → where": to all operators, to a ring group, to the voice bot or to a greeting. Without a rule the call is handled by the default answer mode.
  4. Call your number from a mobile and make sure the call arrives and audio works both ways.
The provider sends its login instead of the number

Some providers (usually in registration mode) send the SIP login in the call instead of your number. Then a line with a number will not match what arrived — but the call is still accepted and goes into routing as "a call from this provider". If the trunk has one number, you can skip creating a line; if there are several numbers, ask the provider how to enable DID passing (for many it is an account setting).


Step 4. Outgoing calls and "Show as"#

Staff can call out through the same trunk. Open the "Outgoing" tab:

  • switch outgoing calls on and assign each employee a default line (the number they call from) or add a pattern rule ("mobile numbers of the country → through this trunk, show number …");
  • a rule can strip or prepend a prefix before sending to the provider if it expects the number in a particular format (for example 8… instead of +7…);
  • the "Show as" field is the number the called party sees. Providers usually allow only the numbers of your contract; a foreign number is silently replaced with theirs or the call is rejected.

The PBX presents itself to the provider with the provider's own domain (From: <number>@<SIP host>) and the trunk login in Contact — as most operators require. If the provider insists on passing the number in the P-Asserted-Identity header, write to support: that option is enabled on the server side.


Network and compatibility#

The dedicated PBX sits in the cloud with a static address, so nothing is required from the venue's network — traffic flows between the provider and the cloud. Requirements for the provider:

Parameter What the PBX supports
Signalling SIP over UDP or TCP, any provider port (usually 5060). TLS is not supported for trunks — a "TLS/SRTP only" provider will not work
Codecs G.711 a-law, G.711 u-law, G.722. The provider must offer at least one of them; "G.729 only" means no audio
DTMF RFC 2833/4733 (tone dialling in separate packets)
Media RTP to the PBX address, UDP ports 10000–20000; the PBX does not pass media directly between the parties
Incoming Recognised by the provider's SIP host; the number is matched by its last 7 digits
Outgoing From domain — the provider's host; Contact — the trunk login; caller ID — the line number or "Show as"
Registration Retried every 30 s on error, every 5 min after a 403 rejection; provider availability check (OPTIONS) every 60 s

If calls do not go through#

Symptom Usual cause What to do
"no registration at provider" Wrong login/password; the provider requires a different host for registration than for calls; the account is blocked for non-payment Check the credentials in the account; at some providers the registration host differs from the "domain" — take the one from the "connect an external PBX" instructions
Registered, but no incoming calls The number in the provider's account is not routed to this SIP account; the line is disabled; the provider sends the login instead of the number Check "number routing" at the provider; the "Accepts calls" checkbox; see the login section above
The call connects, but silence in one or both directions The provider sends media only to an allowlisted address; codec mismatch (the provider offers only G.729) Add the PBX IP to the provider's allowlist; enable G.711 in the provider's account settings
Outgoing calls are rejected (403) The provider does not accept "Show as" with a foreign number; expects the number in another format; blocks outgoing calls from an unregistered address Leave the caller ID empty (the line number will be sent); set a prefix in the outgoing rule; add the IP to the allowlist
Everything drops a minute after connecting The provider does not see our media address behind NAT (typical for home-PBX-style providers) Try TCP; write to the provider — they have a "NAT mode"/"symmetric RTP"
The provider requires TLS 5061 or SRTP Not supported for trunks Choose another provider or ask this one for a plain UDP/TCP account (most have one)

The SIP trace in the trunk menu — "Trunk diagnostics" — helps dig deeper: while the window is open, the PBX records the exchange with the provider and shows at which step the refusal happens.


Providers by country#

Below are guides for specific countries: who issues numbers with a SIP trunk there, in which mode, which parameters to enter and what to enable in the provider's account. The list is growing; countries without a link are in progress.

Country Providers Guide
🇬🇪 Georgia: SIP providers and number connection Telecom 1, CityNet Open
🇷🇺 Russia: SIP providers and number connection MANGO OFFICE, МТТ, МТС Exolve, MCN Telecom, Novofon, UIS, Sipuni, МегаФон, МГТС Open
🇹🇷 Turkey: SIP providers and number connection Verimor Telekom, Netgsm, Telsam Telekom Open
🇦🇱 Albania: SIP providers and number connection Neofone Open
🇰🇿 Kazakhstan: SIP providers and number connection Kcell, Freedom Telecom, Казахтелеком, Tele2 / Altel Open
🇵🇱 Poland: SIP providers and number connection SuperVoIP, ACTIO, Spikon, FCN, PLFON, ipfon, HaloNet Open
🇦🇲 Armenia: SIP providers and number connection Ucom, Viva Armenia, OVIO Open
🇺🇸 United States: SIP providers and number connection Sangoma SIPStation, DIDForSale, SIP.US, SIPTRUNK.com, Skyetel, Voxtelesys, Callcentric, Sangoma Carrier Services, Commio Open
🇬🇧 United Kingdom: SIP providers and number connection sipgate trunking, Andrews & Arnold (AAISP), Voipfone, VoiceHost, Orbtalk, Yay.com, Magrathea, Gradwell Open
🇩🇪 Germany: SIP providers and number connection easybell, sipgate, Placetel, dus.net, fonial, peoplefone, reventix Open
🇫🇷 France: SIP providers and number connection OVHcloud Telecom, ippi, boxIP, Keyyo, Axialys, Dstny France Open
🇪🇸 Spain: SIP providers and number connection netelip, GoTrunk Open
🇮🇹 Italy: SIP providers and number connection MessageNet, OlimonTel, VoipVoice Open
🇳🇱 Netherlands: SIP providers and number connection Voys, VoIP Telecom, CheapConnect, Business-ISP, MaxiTEL Telecom, SpeakUp Open
🇨🇦 Canada: SIP providers and number connection ThinkTel, IXICA Communications, VoIP.ms Open
🇦🇺 Australia: SIP providers and number connection Crazytel, SIPcity, MaxoTel, Breeze Connect, Simtex, Aatrox Communications, Australian Phone Company Open
🇦🇿 Azerbaijan: SIP providers and number connection Azerizone Open
🇺🇿 Uzbekistan: SIP providers and number connection Sarkor Telecom, UZTELECOM, East Telecom Open
🇰🇬 Kyrgyzstan: SIP providers and number connection Beeline, MEGA / MegaCom, O!, ЭлКат, Мега-Лайн Open
🇲🇩 Moldova: SIP providers and number connection Moldtelecom, Moldcell, IPHOST, StarNet Open
🇺🇦 Ukraine: SIP providers and number connection Stream Telecom, Ringostat, Інтертелеком Open
🇧🇾 Belarus: SIP providers and number connection Анитэкс, ПрофТелеком, Деловая сеть Open
🇷🇸 Serbia: SIP providers and number connection SoftNET, Creomate, Orion telekom Open
🇭🇷 Croatia: SIP providers and number connection Telemach Hrvatska, TOP VoIP, Signum telekomunikacije, Tehnoline Telekom Open
🇸🇮 Slovenia: SIP providers and number connection T-2, Telekom Slovenije Open
🇧🇦 Bosnia and Herzegovina: SIP providers and number connection SoftNET, HT Eronet, Telemach BH, *supernova, Logosoft Open
🇲🇰 North Macedonia: SIP providers and number connection НЕОТЕЛ, Gonet, СофтНЕТ ЕУ Open
🇲🇪 Montenegro: SIP providers and number connection IPMONT, Crnogorski Telekom, One Crna Gora Open
🇽🇰 Kosovo: SIP providers and number connection IllyVoIP Open
🇷🇴 Romania: SIP providers and number connection ClickPhone, Voxbee, Optivoice, DOTRO Telecom, Iptelis Networks Open
🇧🇬 Bulgaria: SIP providers and number connection BGTel, PlanetPhone, IP Telecom България Open
🇬🇷 Greece: SIP providers and number connection modulus, Yuboto Telephony, Inter Telecom, VoIP.gr Open
🇨🇿 Czechia: SIP providers and number connection Odorik.cz, FAYN, AXFONE, ha-loo, 802.cz Open
🇸🇰 Slovakia: SIP providers and number connection Slovak Telekom, SWAN, VIPTel, Slovanet Open
🇭🇺 Hungary: SIP providers and number connection MCN Telecom Magyarország / KOMPaaS, Ephone Open
🇱🇹 Lithuania: SIP providers and number connection peoplefone Lietuva Open
🇱🇻 Latvia: SIP providers and number connection CSC TELECOM, Tet, LMT, Bite Latvija Open
🇪🇪 Estonia: SIP providers and number connection Telia Eesti — ViPÜ, Telia Eesti — Ärilahendus VoIP, Apifonica Open
🇦🇹 Austria: SIP providers and number connection easybell Österreich, yuutel, Magenta Telekom, peoplefone, Innosoft Open
🇨🇭 Switzerland: SIP providers and number connection sipcall, peoplefone, iWay, VTX Services Open
🇧🇪 Belgium: SIP providers and number connection edpnet, 3StarsNet, Voys België, Mixvoip, Proximus Open
🇱🇺 Luxembourg: SIP providers and number connection Mixvoip, Luxembourg Online, Visual Online Open
🇮🇪 Ireland: SIP providers and number connection Speechpath, VoIPLine Telecom, in2tel, Byphone, Telxi, NTES, Goldfish Telecoms Open
🇵🇹 Portugal: SIP providers and number connection MEO / Altice Portugal, Vodafone Portugal, G9 Telecom Open
🇸🇪 Sweden: SIP providers and number connection InfraCom / Weblink, Cellip, 46elks, Telavox Open
🇩🇰 Denmark: SIP providers and number connection Redspot, Uni-tel через 1TEL Open
🇳🇴 Norway: SIP providers and number connection Telenor Norge, Telavox Open
🇫🇮 Finland: SIP providers and number connection JNT, Line Carrier, Elisa, DNA Open
🇮🇸 Iceland: SIP providers and number connection Síminn Netsími, Telxi Open
🇲🇹 Malta: SIP providers and number connection Vanilla Telecoms, GO p.l.c., Telxi Open
🇨🇾 Cyprus: SIP providers and number connection Cyta, PrimeTel, Cablenet, Telxi, KCM Telecom Open
🇳🇿 New Zealand: SIP providers and number connection 2talk, Hero Internet, Voyager Open
🇮🇱 Israel: SIP providers and number connection IsraelNumber, Partner, Cloudonix, Telxi Open

International providers with numbers in many countries (Telnyx, Twilio, DIDWW, Zadarma, sipgate, VoIP.ms…) are suitable when the local operator does not offer a trunk or you need a number from another country; they verify documents for the number more strictly (an address and business registration in the number's country).


FAQ#

It depends on the country and the provider. International number services sell to individuals but require proof of address for landline numbers; telecom operators almost always work with businesses under a contract. The country guides state this for every provider.

Can I connect several providers at once?#

Yes. Each provider is a separate trunk, each with its own number lines; routing and outgoing rules tell them apart. This is how venues combine a landline from the local operator with a mobile number from a virtual PBX.

What happens to calls when the server capacity is changed?#

The static IP moves to the new server and the registration at the provider is restored automatically; calls are possible the whole time except for a few seconds of re-registration. Providers with IP authentication need no changes — the address is the same.

The provider asks for a "SIP URI for forwarding" — what is that?#

That is the third way, without a trunk: the provider forwards calls from the number to the SIP address of your extension. It can be used when the provider has no trunk at all, but outgoing calls from the number will not work this way. In the admin panel there is the "Forwarding" line source for this — the wizard shows the address to enter at the provider.

Where are conversations recorded?#

All calls through the trunk are recorded the same way as calls to the app: recording, transcript and analysis are in the "Calls" section. The recording notice for the other party is enabled in the location settings.


  • SIP phone setup — how to connect desk phones and DECT to extensions.
  • The "Telephony" section in the admin panel: lines, routing, ring groups, outgoing calls.