FreeVoIP runs on the open SIP standard, so almost any modern phone or app will work — a smart desk phone, a cordless handset, a free app on your mobile, or the analogue phone already on your desk. No proprietary hardware, no lock-in. Scroll down for setup samples covering IP phones, ATAs, softphones and PBX trunks.
Every option below needs just three things to connect — your user ID, password and our SIP proxy. Mix and match as many as you like.
Purpose-built SIP handsets from Yealink, Grandstream, Fanvil, Snom, Poly and Cisco. Plug into your network, enter your details and you have a full-featured office phone with lines, transfer, hold and speed dials.
Wireless SIP handsets and base stations — roam around the home or office while staying on your VoIP line. A single base can register several cordless handsets, each on its own free account.
Keep the analogue phone you already own. An ATA (Analogue Telephone Adapter) sits between your ordinary handset and your router, turning it into a VoIP line — ideal for reusing a much-loved deskphone or a DECT set.
Turn your Android or iPhone into a VoIP handset with a free softphone app such as Zoiper or Linphone. Take your number and low rates anywhere you have Wi-Fi or data — perfect for travel and remote work.
Make and take calls straight from your laptop, PC or Mac with a free softphone — MicroSIP, Zoiper or Linphone — using your headset or the built-in mic and speakers. On Windows we hand you MicroSIP pre-configured: download, unzip, run, done.
SIP conference units for the meeting room, and PBX gateways or door-entry and paging devices for larger sites. If it speaks SIP, it works on the FreeVoIP network.
The exact fields differ slightly by make, but the values are always the same — your user ID, your password and our SIP server sip.transcom.net. The examples below use the account 158461_myoffice; swap in your own.
Yealink, Grandstream, Fanvil, Snom, Poly and Cisco phones — and Grandstream/Cisco ATAs for an analogue phone — all take the same account details in their web interface. Log into the phone’s IP address in a browser and open the account/line page.
| Field (varies by brand) | Value |
|---|---|
| Enable / Register line | On |
| Display / Label name | My Office |
| User ID / SIP User / Auth name | 158461_myoffice |
| Register name / Authenticate ID | 158461_myoffice |
| Password | your account password |
| SIP server / Registrar / Host | sip.transcom.net |
| Outbound proxy | sip.transcom.net |
| Port | 5060 (UDP) · 5061 for TLS |
| Codecs (in order) | PCMU / G.711u, G.729, G.722 |
Menu paths: Yealink Account → Register · Grandstream Accounts → General Settings · Fanvil Line → SIP · Snom Identity · Grandstream ATA FXS Port. Your free account includes two SIP lines — one per handset — and the 25-line account covers a full office.
On Windows there’s nothing to configure. Every free account includes a ready-to-use MicroSIP softphone, already set up for your line. In My Account open My VoIP → Lines and click 📱 Windows phone beside the line you want. Unzip the download and run MicroSIP.exe — it registers by itself. It’s the portable edition: no installer, nothing written to your system, and you can run one folder per line if you use both.
If you’d rather set it up by hand, the tables below are the exact settings we ship — a known-good, tested configuration for our network. Menu → Add account takes the first table; Menu → Settings takes the second. Note the transport is TCP (not UDP) and media encryption is off.
| MicroSIP — Account | Value |
|---|---|
| Account name | FreeVoIP |
| SIP server / Domain | sip.transcom.net |
| Proxy | (leave empty) |
| Username | 158461_myoffice |
| Login / Auth ID | 158461_myoffice — same as Username |
| Password | your line’s password |
| Display name | your name |
| Transport | TCP |
| Public address | Auto |
| Media encryption | Disabled |
| NAT traversal | ICE only — STUN and TURN off |
| Hide Caller ID | Disabled — required for national rates |
| Allow IP rewrite / Publish presence | Off |
| MicroSIP — Settings | Value |
|---|---|
| Source port | 5061 |
| rport | Ticked |
| RTP ports | 10001 – 20000 |
| STUN server | (empty — disabled) |
| Audio codecs | PCMU (G.711u) |
| Register refresh · Keep-alive | 300 · 15 |
Free apps like Zoiper and Linphone take the same three details. Add a new SIP account and enter:
| Field | Value |
|---|---|
| Account name | FreeVoIP |
| Username / User ID | 158461_myoffice |
| Password | your account password |
| Domain / SIP server / Host | sip.transcom.net |
| Outbound proxy (if asked) | sip.transcom.net |
| Transport | UDP (or TLS) |
Zoiper: Settings → Accounts → Add → SIP, enter 158461_myoffice@sip.transcom.net and your password · Linphone: Assistant → Use a SIP account · MicroSIP: Menu → Add account.
Whatever device you set up, don’t hide your Caller ID if you want our national rates. Those rates — our cheapest — only apply when you present a local Caller ID from the country you’re dialling: a DID you rent from us, or a number you own and have validated. Hidden or unset, the call falls back to the standard rate. So leave Hide Caller ID disabled on the handset or softphone, and set the line’s outbound Caller ID in My Account (My VoIP → Lines). The US & Canada are the exception — no local Caller ID needed. Dial 1-555-555-0911 to hear your Caller ID read back, and see the national rates for the countries covered.
Point your own PBX at FreeVoIP with a registering SIP trunk. In FreePBX go to Connectivity → Trunks → Add (pjsip); the same values apply to a raw Asterisk pjsip.conf or a 3CX generic SIP trunk.
| Trunk field | Value |
|---|---|
| Trunk / provider name | freevoip |
| Username / Auth ID | 158461_myoffice |
| Secret / Password | your account password |
| SIP server / Host / Registrar | sip.transcom.net |
| Port | 5060 |
| Transport | udp |
| Registration | Send (register) |
| Inbound context | from-trunk |
| Allowed codecs | ulaw, g729, g722 |
Legacy Asterisk chan_sip register string: 158461_myoffice:password@sip.transcom.net. In 3CX pick “Generic SIP Trunk” and set the registrar/outbound proxy to sip.transcom.net.
Because FreeVoIP is built on open standards, you’re never tied to one manufacturer or a single supplier’s hardware. Buy a phone once and it works with us today and with any other SIP provider tomorrow — the choice, and the number, stay yours.
Need more than one phone? Your free account includes two SIP lines, so you can be registered in two places at once — a desk phone at the office and a softphone at home, say — and the second line is yours to add any time in My Account. For a busy office, the 25-line account gives you 25 concurrent lines on one bill. Add a DID or FreeFone number to any device, including a computer.
There are hundreds of SIP clients out there and many are completely free — any of them will work. Not sure which to pick, or which codec suits a slow connection? Just ask and we’ll recommend the best fit.
