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.
Set up your account
Open account settings
On desktop: Softphone → Account Settings → + → SIP. On mobile: Settings → Accounts → + → SIP.
Enter your credentials
Account name: e.g. “SipandVoip”. User ID: your SIP username. Domain: your SIP domain. Password: your SIP password. Display name: your name. Authorization name: your SIP username again.
Configure the proxy
On the Account tab, tick Register with domain and receive calls and set Send outbound via to Proxy, entering the outbound proxy address.
Set transport and codecs
On the Transport tab choose UDP (or TLS on port 5061 if requested). On the Audio Codecs tab enable G.711 aLaw, OPUS and G.729, in that order for Wi-Fi.
Enable it
Tick Enabled and click OK. Bria shows “Available” and a green indicator once registration succeeds.
Check NAT settings
If audio is one-way, open Preferences → Advanced and confirm Firewall traversal is set to Auto-detect rather than disabled.
Number rewriting
Bria's Dialing Rules can add a + automatically or convert 00 prefixes into E.164 before a call leaves the client, which saves retraining users who are used to the old system.
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.