Getting started with your SIP Trunk Print

Set up your WebTic SIP Trunk

This guide walks you through putting your WebTic SIP trunk into service on your PBX or softphone in under 10 minutes.

Prerequisites

• An active WebTic account with a SIP trunk subscription
• A SIP-compatible PBX (3CX, FreePBX, Asterisk, Yealink, Grandstream…)
• Your network must allow the SIP ports: UDP/TCP 5060 and the RTP range 10000–20000

Retrieve your SIP credentials

From your client area, go to Services → Your SIP trunk → Details. There you will find:

• SIP server: sip.webtic.online
• SIP username (e.g. 41912XXXXXX)
• SIP password
• Preferred codec: G.711 (alaw/ulaw) or G.729

Configuration on 3CX

1. In the 3CX admin, go to SIP Trunks → Add
2. Choose the Generic SIP Trunk template
3. Enter the name, the proxy server (sip.webtic.online) and your credentials
4. Enable SIP registration (Registration: Trunk registers with provider)
5. Configure the inbound and outbound rules according to your DIDs

Configuration on FreePBX / Asterisk

In Connectivity → Trunks → PJSIP Settings, enter:

• SIP Server: sip.webtic.online
• SIP Username and SIP Password
• Context: from-trunk
• Codec: ulaw, alaw (enable in Settings → Asterisk SIP Settings)

Testing the line

Place a test call to +41 91 204 18 49 (our sales department) or to the internal echo test *43 if configured. A successful call confirms that the trunk is configured correctly.

Troubleshooting common issues

• 401 Unauthorized: check the SIP password in your client area
• No audio: open RTP ports 10000–20000 on your firewall
• 408 Request Timeout: check that UDP port 5060 is open