GSM gateway, landline or ISDN in your PBX

Your own PBX in the admin panel receives calls not only on a SIP provider's number (“Connecting a provider number”) but also on an ordinary SIM card, an analog landline or ISDN, as long as a gateway sits between them and the network — a small box that turns the operator's signal into SIP. To the PBX such a gateway is just another extension, and a call to the venue's SIM card lands in the operator panel like any other.

This page covers what is the same for every gateway: what kinds exist, which credentials the admin panel issues, what to enter, where to send incoming calls, what a gateway cannot do with us yet, how to check it and how to reset it. Guides for specific families (Yeastar, Dinstar, GoIP, Grandstream, OpenVox, Synway, Patton, Sangoma, beroNet and others) are below.

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

When you need a gateway#

• The venue has a number on a SIM card (the mobile number guests and couriers call) and you want it in the PBX: recorded, distributed among staff, handed to the AI consultant. That is a GSM/LTE gateway.
• The premises have a landline from an operator with a regular phone socket, and all regulars know its number. That is an FXO gateway.
• The operator's line is digital, ISDN (BRI or PRI/E1) — common in European offices and hotels. That is an ISDN gateway.
• The other way round, you want old analog phones in the kitchen and warehouse as PBX extensions — that is an FXS gateway or ATA, covered in “Setting up SIP phones”.

If the number can be obtained from a SIP provider directly, no gateway is needed: a SIP trunk is cheaper, needs no hardware and supports outgoing calls. A gateway is for numbers tied to a SIM card or a physical line that you do not want to, or cannot, move to a SIP provider (see “By country”).

Four kinds of gateways#

Many families are modular: one chassis takes GSM, FXO, FXS and E1 modules (OpenVox, Synway, Patton, Sangoma, beroNet). Their SIP side is configured the same way; only the port tabs differ.

How our PBX sees a gateway#

A gateway registers on our PBX as an extension — exactly like a desk phone. It gets an extension account, connects to the PBX server over SIP and keeps the registration alive. The PBX never connects to the gateway and does not know its address, so the gateway can sit behind the venue's ordinary router without a public IP or port forwarding.

Three things follow from this:

1. One port — one extension. On a four-SIM gateway each SIM can get its own extension, so the PBX shows which number was called. Many gateways also support a shared account for all ports — then there is one extension.
2. The gateway itself sends an incoming call from the SIM or line to a number inside the PBX — an operator's extension or a ring group number (see “Where to send incoming calls”).
3. Outgoing calls from the PBX through the gateway are not supported yet — details below.

In the extension store the gateway cards (Yeastar, Grandstream, GoIP, Dinstar, AudioCodes, Cisco ATA, “other SIP gateway”) have a connection wizard: it shows the server address, port and transport and reminds you to issue an extension. The gateway itself is configured in its web interface.

Extension credentials for the gateway#

The credentials are the same as for a phone: Telephony → Operators → “Reset password” on the operator you need (or when creating a new operator) — the “Extension credentials” window opens.

Create a separate operator with a clear name for the gateway (“GSM gateway, SIM Beeline”) rather than giving it a live employee's account: the extension password is shown once, and the gateway needs it forever.

What to enter in any gateway#

The SIP side of every gateway is configured alike; tab names differ — see the family page.

1. SIP server, port, transport — from the credentials window. If the gateway asks for a separate “Outbound Proxy”, leave it empty; “Domain / Realm” — the same server address.
2. Login and password of the extension — in the port account fields or the shared account (on GSM gateways this is usually “SIP Trunk / Account” in registration mode, not “Peer”).
3. Registration — enable it (“Register: Yes”), interval 60–300 s.
4. Codecs — enable G.711 A-law and µ-law, optionally G.722; our PBX does not use G.729, AMR or Opus, a gateway with only those will not negotiate a call.
5. DTMF — RFC 2833 (RFC 4733); not “Inband” and not “SIP INFO”.
6. NAT — enable “NAT Traversal / Keep-alive” if the gateway is behind a router (almost always); STUN is not needed.

A green “Registered” on the gateway's status page and a green dot next to the extension in the “Operators” tab mean the link is up.

Where to send incoming calls#

The gateway receives a call to the SIM card or line itself and must dial a number inside our PBX. In the gateway this is the “Mobile → IP”, “FXO → SIP”, “PSTN → VoIP”, “Inbound route” or “Hotline” route: it says which SIP account to call through and which number to dial.

What you can specify:

• a ring group number (Telephony → “Ring groups”, numbers from 700) — the call is distributed among the group's staff by its rules, with a fallback to voicemail or the AI consultant; this is the most convenient option;
• an operator's extension number — the call reaches one person;
• the number of the AI consultant or bot, if it is enabled in the “Telephony” section;
• temporarily *43 — the echo test, to check audio (see “How to check”).

The routing rules configured for SIP provider numbers (schedule, voice menu, queues) do not apply to calls from a gateway: the gateway has no “line” in the PBX, it calls as an extension. A ring group with its own time-based variants and fallback covers the same need.

The caller's number reaches the panel. The gateway inserts it into its SIP call (an option called “Caller ID passthrough” or similar, enabled by default on most families), and from there the call behaves like any other inbound one: the number shows on the employee's phone, lands in the call log and in the recording in the “Calls” section. Where exactly that option lives in your box — see the family page.

Outgoing calls through the gateway — not yet#

Calling out from the PBX through the gateway's SIM card or landline is not possible right now: the outgoing rule in the PBX can only send a call to a SIP provider's line, and to the PBX a gateway is an extension, not a line. If you call the gateway's extension, the call reaches the gateway itself but does not “exit” to the operator's network to the number you want.

In practice this means the gateway today is a way to receive calls on the SIM card's or line's number. For outgoing calls from your own number use a SIP provider's number (“Connecting a provider number”). Outgoing calls through a gateway are planned; as soon as they arrive, this page will describe how to enable them.

Venue network, NAT and the SIM card#

• Ports. The gateway opens outgoing connections to the PBX server itself: 5060/UDP for signalling and 10000–20000/UDP for audio. No inbound port forwarding is needed on the router. If the router has SIP ALG — turn it off: it breaks registration and causes one-way audio on gateways just as on phones.
• One network segment. The gateway and the computer used to configure it must be on the same network; exposing the gateway's web interface to the internet is unnecessary and unsafe — change the factory password right after setup.
• The SIM card for the gateway. The operator may require the SIM to be registered to a legal entity, and in a number of countries using a SIM in a gateway is restricted by law or by the operator's terms — see “By country”. Remove the PIN from the SIM in a phone before inserting it into the gateway, or enter it in the port settings.
• The operator's network. Where 2G and 3G are already switched off, you need a gateway with VoLTE (4G voice); a plain GSM gateway will not register on such a network. The list of switch-offs by country is on the same page.
• The landline. FXO gateways have country-specific settings: line impedance, busy tone for hang-up detection, CallerID format. Without them a call may not end after the caller hangs up.

How to check#

1. Registration. In the gateway — account status “Registered” (green); in the admin panel — a green dot next to the gateway's extension in the “Operators” tab.
2. A call. Call the SIM card or the landline number connected to the gateway from a mobile: the call must reach the group or operator the inbound route points to.
3. Audio. Registration says nothing about audio — it travels as a separate stream. Temporarily point the gateway's inbound route to the service number *43 (echo test: a tone, then your own speech returned with a delay) and call the SIM card. If you hear yourself, audio works both ways; the test results with loss and delay appear as a step in the call log. Then point the route back to the group.

Factory reset#

A reset is needed when the web interface password is unknown, the gateway came from a previous owner with foreign settings, or a firmware update went wrong. Every family has two ways: by button (RESET on the case, hold for 5–10 seconds with the power on) and from the web interface (System → Reset / Factory Default). A reset wipes SIP accounts, routes and network settings, and on GSM gateways also the stored SIM PIN codes; afterwards the gateway takes an address by DHCP again (or returns to its factory static address — see the family page) and asks for the factory password. Exact combinations, addresses and passwords are on the family page.

Guides by family#

Below are the pages for specific product lines: how to find this box's IP, the factory password, exactly where the SIP account fields are, how to create the inbound route, how to reset. The list keeps growing; families without a link are in progress.

Your gateway's brand is not on the list? Any SIP gateway that can register on an external SIP server will do: configure it following “What to enter” and “Where to send incoming calls”.

By country#

What a gateway is good for depends heavily on the country: in some places a GSM gateway with your own SIM is routine, in others using a SIM in a gateway requires registration or is banned; in some, analog lines are already switched off and a landline number is issued only as VoIP; in others, 2G and 3G are gone and you need a VoLTE gateway. A summary table for 98 countries — legality of a GSM gateway for your own calls, SIM requirements, the fate of the landline and ISDN, 2G/3G switch-offs, certification and local resellers — lives on a separate page: “By country: where GSM gateways are legal, where landlines and ISDN still exist”.

If the gateway does not register#

1. The login was not entered in full — the gateway has separate “User Name” and “Auth Name / Auth ID” fields: paste the whole sip-…-<number> login into both.
2. The password is stale — the password is shown once; if it was reset, issue a new one and enter it again.
3. TLS or TCP transport where UDP was expected — set UDP and port 5060.
4. SIP ALG on the router — turn it off and restart the gateway.
5. Wrong account mode — on GSM gateways the “Peer / IP trunk” mode without registration does not work with us; use “Register / Account”.
6. The gateway is registered but the call does not arrive — the inbound route is not set or points to a non-existent number; check that the group's or operator's number can be dialled from any PBX phone.
7. The call arrives but never ends (FXO) — hang-up detection is not set for the country: busy tone or polarity reversal.
8. The GSM gateway does not see the network — the SIM has a PIN, the antenna is not attached, or 2G/3G is switched off in the country and a VoLTE gateway is needed.

FAQ#

Can I receive calls on the SIM card and call out from it through the PBX at the same time?#

Receiving — yes, that is the main scenario. Calling out through the SIM card from the PBX is not possible yet (see “Outgoing calls through the gateway”); outgoing calls go through a SIP provider's number.

How many extensions does a multi-SIM gateway need?#

As many as the SIM cards you want to tell apart in the PBX: one extension per port. If you do not need to tell them apart, many gateways register with a single shared account.

Does the gateway need a public IP address or port forwarding?#

No. The gateway connects to the PBX server itself and keeps the registration alive; ordinary internet access from the venue's network is enough.

A call arrived but the panel shows the gateway's number, not the caller's?#

The gateway does pass the caller's number on — that works. If you see the gateway's extension number instead, caller number passthrough is switched off (or misconfigured) in the port settings: enable “Caller ID passthrough” — depending on the family it is called “From Mobile Number”, “Caller ID Source” or “CID from Mobile” — and restart the gateway. The family page says where the option lives in your box.

Will a gateway that is not in the family list work?#

Yes, if it can register on an external SIP server and supports G.711. Configure it following the general sections of this page; the extension store has an “Other SIP gateway” card for this.