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.
Menu names differ slightly between MiVoice Business, MiVoice Office 250 and MiVoice Office 400, and between releases. The sequence below follows MiVoice Business; the concepts map directly onto the others.
1. Create the network element
System Administration Tool → Network Elements → Add.
- Name:
SipandVoip - Type: Other
- FQDN or IP Address:
sip.sipandvoip.com.cy - SIP Peer Transport: UDP (or TLS if you have requested it)
- Port:
5060(5061for TLS)
2. Add the SIP peer profile
SIP Peer Profile → Add, referencing the network element above.
- Registration: Enabled; enter your SIP username and password.
- Address type in From/To headers: Peer FQDN or IP.
- Calling line ID: use a DID you own — do not present an arbitrary number.
- Maximum simultaneous calls: set to your purchased channel count.
- Renegotiate SDP on hold: Yes. Session timer: Enabled.
- Allow using UPDATE for early media: Yes.
3. Codecs and DTMF
Under SIP Peer Profile → Codec Options, enable G.711 A-law as first preference and G.729 as a fallback. Set DTMF to RFC 2833, payload type 101. Disable T.38 unless you are actually faxing — if you are, tell us so we can enable it on our side.
4. Trunk attributes and routing
- Trunk Attributes: assign a trunk service number and a COS that permits the destinations you intend to call.
- Route: create a route pointing at the SIP peer, with digit modification that outputs full international format.
- Automatic Route Selection (ARS): add digit strings for local, national, mobile and international, each pointing at that route.
- DID Ranges / Incoming Digit Modification: map each inbound DID to the target extension, hunt group or auto-attendant.
5. NAT
If the Mitel system sits behind NAT, configure the external IP under System IP Properties and forward the SIP and RTP port ranges. As always, disable SIP ALG on the firewall.
6. Test
Check Maintenance Commands → SIP Peer Status for a registered peer, then place calls in both directions. Mitel's SIP logs, or a packet capture on the trunk interface, will show where a failing call stops.
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.