Before you start — you will need the details from your welcome email: SIP domain, username (usually your trunk or extension number), password, and the outbound proxy if one was issued. If you do not have them, contact support.
| Setting | Value |
|---|---|
| SIP domain / registrar | sip.sipandvoip.com.cy |
| Outbound proxy | sip.sipandvoip.com.cy |
| SIP port (UDP / TCP) | 5060 |
| SIP port (TLS) | 5061 |
| Transport | UDP, TCP or TLS |
| Codecs | G.711 A-law (alaw), G.729, OPUS |
| DTMF | RFC 2833 |
| Username / password | From your welcome email |
⚠ These are placeholder values — set the real ones in the editor (Global tab) and they update on every guide at once.
1. Add the SIP trunk
- In the 3CX Management Console, go to SIP Trunks → Add SIP Trunk.
- Country: Cyprus (or Generic if you are outside Cyprus).
- Provider: Generic SIP Trunk.
- Main Trunk Number: your primary DID in full international format, e.g.
+35700000000.
2. Registrar and authentication
| Registrar / Server | sip.sipandvoip.com.cy |
| Outbound proxy | sip.sipandvoip.com.cy |
| Type of authentication | Register / Account based |
| Authentication ID | Your SIP username |
| Authentication password | Your SIP password |
| Simultaneous calls | Set to the number of channels you have purchased |
3. Route inbound calls
On the trunk's DID tab, add each DID you own in full international format. Then under Inbound Rules, create a rule per DID and choose the destination — an extension, ring group, queue or digital receptionist — with separate office-hours and out-of-hours destinations if you need them.
4. Route outbound calls
- Go to Outbound Rules → Add.
- Rule name:
SipandVoip out. - Apply to: the extensions or groups allowed to dial out.
- Calls to numbers starting with:
+(and00if your users dial that way). - Route 1: your SipandVoip trunk. Strip digits / prepend as needed so the number leaves in E.164.
5. Caller ID
Set the outbound caller ID per extension under Extensions → extension → Options → Outbound Caller ID, using a DID that belongs to you. Presenting a number you do not own will be rejected.
6. Firewall check
Run the built-in Firewall Checker (Dashboard → Firewall) before going live. It verifies that your SIP and RTP ports are reachable and that your router is not mangling SIP. Disable SIP ALG on the router first — 3CX will flag it if it is on.
7. Test
Confirm the trunk shows green in SIP Trunks, then place a test call out and in. 3CX's Activity Log and SIP Logs (Settings → Logging, set to Verbose) give you the trace if anything fails.
If it does not register
- 401 / 403 on REGISTER — username or password is wrong, or the device is presenting a different username in the auth header than in the From header. Check both.
- No response at all — a firewall is dropping SIP. Allow outbound UDP 5060 (or TCP/TLS 5061) and the RTP range to our network, and make sure any SIP ALG on the router is disabled. SIP ALG causes more problems than it solves.
- Registers, but one-way or no audio — almost always NAT. Set the external address on your PBX, forward or open the RTP port range, and confirm the device is advertising its public IP in SDP.
- Calls drop after ~30 seconds — SIP session timers or ACK not reaching us. Enable session timers and re-check NAT traversal settings.
- Outbound rejected with 404 — number format. Send E.164 with the leading
+or full international format without a trunk prefix.
Still stuck? Send us the SIP trace and we will look at it against our side of the call.