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
Add a SIP account
Open GS Wave and go to Settings → Account Settings → + → SIP Account.
Fill in the account details
Account Name: anything you like, e.g. “Office”. SIP Server: your SIP domain. SIP User ID, Authentication ID and Password: as supplied in your welcome email. Display Name: your name as you want it shown internally.
Set the outbound proxy
Open Advanced Settings for the account and enter the outbound proxy. Leave SIP Transport on UDP unless TLS was requested for your account.
Tune codecs
Under Codec Settings, put OPUS or G.729 first for mobile data and PCMA (G.711 A-law) first on Wi-Fi. Set DTMF to RFC 2833.
Register
Return to the account list. A green dot next to the account means it is registered. Grey or red means it is not — check the SIP server address and password first.
Allow background operation
In your phone’s system settings, exempt GS Wave from battery optimisation and allow notifications. Aggressive battery savers on Android are the most common reason a softphone misses calls.
Wi-Fi and mobile data
GS Wave can be restricted to Wi-Fi only under Settings → Advanced Settings → Network. If your staff take calls on mobile data, leave both enabled and prefer a low-bandwidth codec.
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.