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📶 GSM Gateways and Landlines

📶 AddPac AP-GS1001/1002/1004/GS2000

How to connect a GSM/LTE gateway with your own SIM card to your PBX: IP address, web interface login, extension SIP account, routing incoming SIM calls to a group or operator, testing and factory reset

Documentation

AddPac AP-GS1001/1002/1004/GS2000 — setup for our PBX

AddPac VoiceFinder AP-GS are GSM gateways from the Korean company AddPac Technology: the desktop AP-GS1001, AP-GS1002 and AP-GS1004 for 1, 2 and 4 SIM cards, and the rack-mount AP-GS2000 chassis for up to 12 SIMs. You insert an operator's SIM into the box, and calls to its number reach an IP PBX over SIP. The gateways run AddPac's own OS, APOS: they have the Smart Web Manager web interface and a Cisco-style command line (console or telnet).

Two important limitations. First, these models work only on 2G (GSM) networks: where the operator has switched 2G off, the gateway will not register on the network (see “By country”). Second, the vendor's documentation dates from 2010–2012, and the firmware is not publicly available. So this page is for venues that already have such a gateway installed. For a new purchase, look at gateways with LTE and VoLTE, for example Yeastar TG.

To our PBX the gateway is an extension. The gateway itself dials an incoming SIM call through to a ring group or an operator. What is the same for all gateways (credentials, where to send incoming calls, network, limitations) is in the overview “GSM gateways and landlines”. This page is the step-by-step guide for AddPac AP-GS, with screenshots from the manufacturer's manuals.

What you need#

  • Extension credentials. In the admin panel: Telephony → Operators → create an operator for the gateway (for example, “GSM gateway AddPac”) → “Reset password” → the “Extension credentials” window: SIP server, Extension number, Login, Password, Port 5060, Transport UDP. The password is shown once — copy it into the gateway right away. Details — “Extension credentials for the gateway”. The simplest option is one extension for the whole gateway; if you want to tell the SIMs apart in the PBX, create an extension for each SIM.
  • The number the gateway will send incoming calls to: a ring group number (Telephony → “Ring groups”, numbers from 700) or an operator's extension number.
  • A SIM card with a voice plan on a network where 2G still works. It is best to remove the PIN in a phone beforehand; if you keep it, enter it in the gateway (Advanced → GSM PINs).
  • An antenna. The AP-GS1001 and AP-GS1002 have one per channel. The AP-GS1004 and the chassis' AP-N1-GSM4 modules use one antenna for four channels (through a built-in combiner).
  • The power supply from the box (AP-GS1001 — 5 V/2 A, AP-GS1004 — 12 V/3.4 A), a network cable, a computer and a mobile phone to call the gateway's SIM. For the AP-GS1004 you may need a console cable RS-232C (RJ45 connector on the gateway) — see step 1.

Step 1. Find the gateway's IP address#

The gateway has two network ports with different factory settings:

Port Out of the box Used for
LAN1 static address 192.168.10.1, mask 255.255.255.0 first login from a computer over a cable
LAN0 / WAN DHCP: the address comes from the venue's router the working connection to the network and the PBX

AP-GS1001, AP-GS1002 and the chassis (AP-GS1200/1500/2000):

  1. Screw on the antenna, insert the SIM, connect power.
  2. Give the computer an address in the same network, for example 192.168.10.10 (mask 255.255.255.0), and connect it with a cable to the LAN1 port (the manual shows a crossover cable).
  3. Open http://192.168.10.1 in a browser and go on to step 2.
  4. Connect the LAN0/WAN port to the venue's router. Look up the address the gateway gets via DHCP in the router's DHCP lease list. In System → Network Setup you can also set a static address (Static IP), PPPoE, or keep DHCP; press Apply.

AP-GS1004 — it has one Ethernet port, and according to the manual the 192.168.10.1 method does not apply to it. First look for the gateway in the router's DHCP lease list. The documentation does not say whether the port uses DHCP out of the box; if it is not in the list, set the address from the console (below).

From the console (AP-GS1002, AP-GS1004, chassis; the AP-GS1001 has no console port). Terminal settings: 9600 baud, no parity, 1 stop bit, no flow control; log in with root / router. To set the address:

GSM> enable
GSM# configure terminal
GSM(config)# interface FastEthernet 0/0
GSM(config-if)# ip address <IP address> <mask>
GSM(config-if)# exit
GSM(config)# ip route 0.0.0.0 0.0.0.0 <router address>
GSM(config)# write

The show interface command shows the current address. AddPac has no network discovery utility and no “tell me the IP” voice menu.

Step 2. Log in to the web interface#

What Value
Address http://<gateway IP> (port 80)
Login root
Password router

The same root / router pair works for the console, telnet and FTP. The web interface language is set in System → Language (English or Korean). Each page is saved with its own Apply button.

Change the factory password right away. The gateway does not require it, and one factory password covers everything — anyone who reaches the gateway gets access to the SIM cards. The web interface description has no password-change item; the vendor describes it with commands (console or telnet — there is a Telnet icon on the web interface's tool bar):

GSM# configure terminal
GSM(config)# user change root <old password> <new password>
GSM(config)# write

The show user command displays passwords in clear text — do not leave its output on screen. Access to the web, telnet and FTP can be limited to a list of addresses: Advanced → Filtering (HTTP Filter, Telnet Filter, FTP Filter).

Step 3. Enter the PBX credentials#

In the vendor's manuals this section is called VoIP Setup; in the web interface menu it is in the Basic block.

1. Protocol. Basic → ProtocolSIP (the factory default).

2. Server. Basic → Server SIP:

  • Use SIP ServerYes;
  • Primary SIP Server — the server address from the credentials window, 5060 in the port field next to it;
  • Secondary SIP Server and Local Domain Name — empty;
  • SIP Signaling Port5060 (the gateway's own port; changing it requires a reboot);
  • Register Expiration60 (the factory default; 60–300 seconds works for our PBX);
  • Apply.

AddPac Smart Web Manager Server SIP page: Basic menu on the left, fields Use SIP Server, Primary SIP Server and Secondary SIP Server with port 5060, Local Domain name, SIP Signaling Port, Register Expiration 60, Session Re-Fresh, Session Expire Time, Apply button Screenshot: GSM Gateway WEB Setup Guide AP-GS1001/1002/1004 (2012), p. 21.

The manuals ask for an IP address in Primary SIP Server, but the CLI manual also allows a server name — provided the gateway has a DNS server set. If the gateway gets its address via DHCP, DNS usually comes from the router; with a static address, set it in System → Network Setup or with the command dnshost nameserver <DNS server IP>.

3. Account. Basic → SIP Registration — a table with one row per port. Fill in the port's row:

In the admin panel (“Extension credentials”) Field in the gateway (Basic → SIP Registration)
Login sip-…-<number> E.164 Number and User Name — the whole string in both fields
Password Password
SIP server not here — Primary SIP Server on the Server SIP page
Port 5060 the port field next to Primary SIP Server
Transport UDP no choice: the vendor's documentation describes UDP only
Extension number not entered — just remember which extension the gateway uses
DisplayName — any name; Reg — ✓ (without it the gateway does not register); HuntStop — ✓ (the vendor recommends it)

Press Apply, then refresh the page (F5) — the registration status appears in the Information panel on the right (screenshot in step 5).

AddPac SIP Registration page: SIP Registration Configuration table with columns Port, E.164 Number, User Name, Password, DisplayName, HuntStop, Reg for ports S0P2 and S0P3, and the Apply button Screenshot: GSM Gateway WEB Setup Guide AP-GS1001/1002/1004 (2012), p. 22.

Why the login also goes into E.164 Number. An AddPac gateway puts the number from the E.164 Number field into the registration request (the From/To addresses) and sends the User Name only during authentication — that is how the message exchange looks in the vendor's training guide. Our PBX identifies the extension by its login, so digits alone in E.164 Number are not enough: registration will fail. If the field does not accept letters and hyphens, or the table has no row for the port you need (in the vendor's AP-GS1002 example only the FXS ports S0P2 and S0P3 are listed), set up registration from the command line — below.

Registering by user name from the command line (user-register mode)

The vendor's training guide (AddPac VoIP Gateway Training Guide, slides 52–55) describes two registration modes. The normal one (register e164) registers the port number, while user-register mode registers the port's user name (user-name); that is the one matching how our PBX expects the login. Connect via the console or telnet (root / router) and find the dial-peer voice <N> pots block for the GSM port (port 0/0, 0/1…) in show run. Then:

GSM# configure terminal
GSM(config)# dial-peer voice <N> pots
GSM(config-dialpeer-pots-N)# user-name sip-xxxxxxxx-NNN
GSM(config-dialpeer-pots-N)# user-password <extension password>
GSM(config-dialpeer-pots-N)# exit
GSM(config)# sip-ua
GSM(config-sip-ua)# user-register
GSM(config-sip-ua)# sip-server <server address>
GSM(config-sip-ua)# register e164
GSM(config-sip-ua)# exit
GSM(config)# write
  • APOS supports the user-name / user-password pair in a dial-peer from version 7.0.
  • If a global sip-username / sip-password is set under sip-ua, the dial-peer pair is ignored — the global setting takes priority (APOS Quick Operation Guide). Remove the global pair with no … in sip-ua mode.
  • Registration status — the show sip command.
Codecs, DTMF, NAT, protection against unwanted calls
  • CodecsBasic → DTMF/CODECVoice CODEC: Preference 1g711alaw (Europe, CIS, Asia), Preference 2g711ulaw (North America). Our PBX does not use G.729, G.723.1 or G.726; the gateway has no G.722 — G.711 is enough.
  • DTMF — same page, DTMF Relay mode: choose DTMF relay by RTP payload defined by RFC 2833. The factory default is Out-of-band signal — with it a voice menu or bot will not understand digits dialled during the call. The CLI manual also says that only RFC 2833 is fully supported for SIP.
  • NAT. The gateway has no separate keep-alive checkbox; the short Register Expiration (60 s) keeps the connection through the router open by itself. No port forwarding is needed; turn off SIP ALG on the router.
  • Advanced → Security. SIP Shutdown — the vendor writes that this item must allow SIP signalling for SIP to work. IP Filtering, War Dialing Filtering, Allow digit Length restrict calls from the IP side — they are not needed for incoming SIM calls.
  • Encryption. TLS and SRTP are not listed in the documentation — our PBX does not offer them for extensions either.

Step 4. Send incoming calls to the PBX#

Unconfigured, the gateway answers a call to the SIM with a second dial tone and waits for the caller to dial an extension (“two-stage dialling”). Guests do not know the extensions, so the call must go straight to the PBX. AddPac does this with Hot Line (a direct connection from a port to a number):

  1. Open Basic → Hot Line (in the manuals — GSM Setup → Hot Line).
  2. In the row of the GSM port (S0P0(G), S0P1(G)… — G means GSM, S means FXS) enter a number in our PBX in Hot Line Number: a ring group number (from 700) or an operator's extension number. According to the manual, the gateway forwards every incoming call on the GSM port to this number.
  3. Digit Input Timeout applies to FXS ports only — leave it alone for GSM. Press Apply.

AddPac Hot Line page: Hot Line Configuration table with ports S0P0(G), S0P1(G), S0P2(S), S0P3(S), Hot Line Number and Digit Input Timeout fields and the Apply button Screenshot: GSM Gateway WEB Setup Guide AP-GS1001/1002/1004 (2012), p. 17.

  1. For the dialled number to go to our server, Basic → VoIP Dial Plan must have a prefix served by the SIP server in the Prefix Table: add the exact number from Hot Line (for example, 701) or T (any number), PlanIndexN.A. (no number conversion), Apply. AddPac patterns: T — any digits, . — one digit, [ ] — a range; for example, 7.. means three-digit numbers starting with 7.

Caller number. The gateway detects the caller's number (Advanced → Gain & CIDCaller-ID), but the documentation does not say where it puts it in the SIP call. Be prepared for the call to appear in the operator panel as coming from the gateway's extension. The routing rules for provider numbers (schedule, voice menu) do not apply to calls from a gateway — a ring group covers the same need (see “Where to send incoming calls”).

Outgoing calls. Calling out from the PBX through this gateway's SIM cards is not possible yet: the PBX's outgoing rule only leads to a SIP provider line. The AddPac itself can make outgoing calls (GSM Extension, GSM Dial Plan, LCR), but leave those pages alone for our PBX. For outgoing calls from your own number use a SIP provider; details — “Outgoing calls through a gateway — not yet”.

Step 5. Check registration and a call#

  1. SIM on the network. Miscellaneous (Monitoring in the 2012 manual) → GSM Status: GSM network registration status, signal strength, SIM number. No registration — check the PIN, the antenna and whether your operator still runs 2G.
  2. Registration in the gateway. After Apply on the SIP Registration page, press F5: the Information panel on the right should show Status : Registered. More detail — Miscellaneous → System Status (SIP client state) and Alarm Status (SIP register fail counter — the count of failed attempts).
  3. Registration in the admin panel. Telephony → “Operators” tab — the gateway's extension shows a green dot.
  4. A call. Call the SIM number from a mobile phone — the operators of the group (or the operator) from Hot Line should ring.
  5. Audio — the *43 echo test. Registration says nothing about audio. Temporarily enter *43 in Hot Line Number (and the same prefix in the Prefix Table), press Apply and call the SIM from a mobile: the PBX plays a short beep and then returns your voice with a delay — talk for about ten seconds. You hear the beep and yourself — audio works both ways; the beep but not yourself — audio does not get from the gateway to the PBX; not even the beep — audio does not get from the PBX to the gateway. The test result is visible in the call log. If the gateway does not save the asterisk in the number field, check audio with a live conversation with an operator of the group. After the check, put the group or operator number back.

Example AddPac SIP Registration page with a filled-in row for port S0P2 (E.164 Number, User Name, Password, DisplayName, Reg and HuntStop checkboxes) and the Information panel on the right with the line Status : Registered Screenshot: GSM Gateway WEB Setup Guide AP-GS1001/1002/1004 (2012), p. 23.

Everything about checking is in the overview, section “How to check”.

Factory reset#

A reset is needed if the gateway came with someone else's settings, the password is unknown, or the gateway behaves unpredictably. The documentation has no factory-reset button: the RST button on the AP-GS3000 chassis only restarts the gateway, and the manuals for the desktop models describe no reset button.

From the web interface. The Smart Web Manager tool bar (Tool Bar, the icons at the top right) holds System Update, Configuration Backup, Initialization, Restart and Telnet. Save the configuration first (Configuration Backup), then choose Initialization.

AddPac Smart Web Manager main page with the vendor's callouts: Main Menu (System, Basic, Advanced), Workspace, the Information panel on the right and the Tool Bar at the top right — System Update, Configuration Backup, Initialization, Restart, Telnet Screenshot: Smart Web Manager for GSM Gateway Series (2010), p. 3.

From the boot loader over the console (AP-GS1002, AP-GS1004, chassis). Restart the gateway (reboot or power) and, while it starts, alternately press Ctrl-C and Ctrl-X until the BOOT# prompt appears. Then:

BOOT# erase
Do you want to ERASE configuration ? [y|n] y
Erasing configuration....done

If the root password is forgotten. Boot loader version 2.0 and later asks for the root password itself. Without it, log in to the boot loader as guest / guest, then configpassword router router. The vendor warns that this erases the configuration. Older boot loaders let you in without a password and show it (BOOT# show password). The AP-GS1001 has no console — the documentation does not say how to reset it with a lost password; contact your supplier or AddPac support.

What is erased: the whole gateway configuration — the PBX account, Hot Line, dial plan, SIM PIN codes, network settings. What to do next: LAN1 is 192.168.10.1 again, WAN uses DHCP, login root / router; repeat steps 1–4 and change the password. Resetting the gateway does not change the extension's login and password in the admin panel; if you did not keep the password, issue a new one with “Reset password”. General notes on resetting gateways — in the overview.

If the gateway does not register#

  1. Only digits in E.164 Number. Our PBX looks up the extension by login, and the gateway registers exactly the E.164 Number. Enter the whole login sip-…-<number> there (and in User Name too); if the field does not accept it — use user-register mode from the command line (step 3).
  2. Registration is not enabled. Use SIP Server = No on the Server SIP page, the Reg box is not ticked in the port's row, H.323 is selected in Basic → Protocol, or the page was not saved with Apply.
  3. The server address cannot be resolved. If Primary SIP Server holds a name, the gateway needs a DNS server (step 3). You can check the connection to the server on the Ping page (Miscellaneous / Monitoring).
  4. Registered, but calls do not arrive. An empty Hot Line Number (the caller hears a second dial tone), no Prefix Table entry for the dialled number, or a non-existent group number. Check that the group number can be dialled from any PBX phone.
  5. The SIM is not on the network. GSM Status shows no registration: the PIN is not removed, the antenna is not connected, or 2G is switched off in the country — then this gateway will not work with any SIM.
  6. Registration drops or audio is one-way. Turn off SIP ALG on the router and do not raise Register Expiration above 300 seconds.

General causes (SIP ALG on the router, UDP 5060 blocked) — in the overview, section “If the gateway does not register”; on network and SIM — “Venue network, NAT and SIM card”.

Models in the family#

The web interface, SIP account and Hot Line are the same on all models; they differ in the number of SIMs, network ports and whether there is a console. All models are 2G GSM only (bands are chosen in Advanced → GSM Band, Auto by default).

Model SIMs / channels Network and ports Setup differences
AP-GS1001 1 / 1 2G GSM; 2× Ethernet (LAN0 = WAN, LAN1); Model A — no analog port, B — 1 FXS, C — 1 FXO no console: web and telnet only; first login via LAN1
AP-GS1002 2 / 2 2G GSM + 2 FXS; 2× Ethernet; 2 antennas RS-232C console; the vendor uses it to show Asterisk integration
AP-GS1004 4 / 4 2G GSM, 1 antenna for 4 channels; Model A — no analog ports, B — 4 FXS, C — 4 FXO one Ethernet port, no 192.168.10.1 access — address from the router's DHCP or from the console (RJ45)
AP-GS1004I 4 / 4 like AP-GS1004, but 4 separate antennas like AP-GS1004
AP-GS1200 up to 4 chassis, 1 slot: AP-N1-GSM4 module (or analog modules) Multi-Port web guide; console
AP-GS1500 up to 8 chassis, AP-N1-GSM4 modules like AP-GS1200
AP-GS2000 up to 12 19″ 1U, 3 slots: up to 3× AP-N1-GSM4 or 8 GSM + 8 FXS/FXO; optional E1/T1 like AP-GS1200; SIP Registration has a Group column for E1

For a café or salon with one or two mobile numbers, an AP-GS1001 or AP-GS1002 is enough. The larger chassis (AP-GS2500…AP-GS5000, Multi-SIM for hundreds of SIM cards) are equipment for traffic carriers; a venue does not need them.

The AP-3G100x 3G series and the AP-4GV1002B 4G model
  • AP-3G1001 / AP-3G1002 / AP-3G1002H / AP-3G1004 — the same APOS, but a 3G network (UMTS/HSPA) with fallback to GSM. The 3G frequency pair is chosen when ordering: Band I (2100) + VIII (900) or Band II (1900) + V (850). They do not work where 3G is switched off too. Our set of vendor documents has no manuals for this series — check the details with your supplier.
  • AP-4GV1002B — not a gateway for a SIM number: LTE is only its internet link, and voice goes over SIP to two analog phones (FXS). It does not bring the SIM card's number into the PBX; its analog ports are connected as ordinary extensions — see “SIP phone setup”.
  • VoLTE (voice over 4G) is not supported by any AddPac model.

FAQ#

Can I call out from the PBX through this gateway using the SIM's number?#

Not yet — with us the gateway handles incoming calls only. The AddPac itself can make outgoing calls (GSM Extension, GSM Dial Plan, LCR), but our PBX does not yet send calls to a gateway extension as if it were a line. For outgoing calls from your own number, use a SIP provider.

Will the gateway work in my country?#

Only if the operator still runs a 2G network: the AP-GS1001/1002/1004/GS2000 use no other networks. In many countries 2G is already switched off fully or partly — check the “By country” page before going on site.

Why does the caller hear a second dial tone instead of an operator?#

The GSM port has no Hot Line Number — without it the gateway works in two-stage dialling mode and waits for the caller to dial an extension. Set up Hot Line (step 4).

Where do I get new firmware?#

The firmware is not publicly available: the downloads section of the AddPac site has no working files for GSM gateways, and updates go over FTP/TFTP through vendor support. Everything our PBX needs (SIP, G.711, RFC 2833) is already described in the 2010–2012 manuals.

What about SMS and the SIM balance?#

The gateway handles them itself: incoming and outgoing SMS — the SMS section of the web interface, balance — Advanced → GSM USSD. Leave cell selection (BTS) on Auto: holding or changing cells manually is a tool of bulk traffic termination, and the operator may block the SIM for it. On SIM cards in gateways in general — “Venue network, NAT and SIM card”.

Sources#

AddPac Technology documents (www.addpac.com), accessed 10.09.2026:

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