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Creating a SIP Register Trunk on Yeastar P-Series Cloud Edition

SIP trunk register trunk VoIP provider PSTN

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Introduction

A SIP trunk connects your PBX to a VoIP carrier, enabling inbound and outbound calls to real phone numbers. The most common type is a Register Trunk, where the PBX registers to the carrier's SIP server. Your reseller can supply SIP trunks alongside your Cloud Edition service.

What you need from your VoIP carrier

  • SIP server hostname or IP (e.g. sip.carrier.co.uk)
  • SIP port (usually 5060)
  • Username and password
  • Your inbound phone number(s) / DDI(s)

Menu path

Extension and Trunk > Trunk, click Add.

Step-by-Step Guide

Step 1 — Basic section

Field What to enter
Name Descriptive name, e.g. Main SIP Trunk
Trunk Status Enabled
Select ITSP Template Choose your carrier from the list, or select General if not listed

Step 2 — Detailed configuration

Change Trunk Type to Register Trunk, then fill in:

Field What to enter
Transport UDP (most carriers)
Hostname/IP Your carrier's SIP server (e.g. sip.carrier.co.uk)
Port 5060 (default)
Domain Usually the same as Hostname/IP
Username Your SIP account username
Password Your SIP account password
Authentication Name Usually the same as Username

Step 3 — Advanced SIP settings (important for caller ID)

In the Advanced tab, set these three fields to Default to ensure correct caller ID presentation and smooth compatibility with UK carriers:

Setting Value
Remote-Party-ID Default
P-Asserted-Identity Default
P-Preferred-Identity Default

Setting these incorrectly can cause caller ID to display wrong on outbound calls or cause rejection by the carrier.

Step 4 — DIDs/DDIs tab

Add your inbound phone number(s):
1. Click the DIDs tab.
2. Click Add.
3. Enter your DDI number (e.g. 01234567890).
4. Add additional DDIs if you have multiple numbers.
5. Click OK.

Step 5 — Save and verify

Click Save and Apply.

Go back to Extension and Trunk > Trunk. Within 30–60 seconds the trunk status should show Registered (green).

Checking your channel capacity

The number of concurrent calls your trunk supports is set by your carrier. Contact your reseller if you need to check or increase your channel allocation.

Troubleshooting

Problem Likely cause and fix
Trunk not registering Check username, password, and hostname — copy-paste from carrier documentation
One-way audio Go to trunk Codec tab, ensure G.711 a-law (alaw) is listed first
Wrong caller ID displayed Check Remote-Party-ID, P-Asserted-Identity, P-Preferred-Identity are set to Default
Calls failing with "403 Forbidden" Check Authentication Name matches exactly what the carrier expects

Summary

The SIP trunk is registered and ready. Remember to set Remote-Party-ID, P-Asserted-Identity, and P-Preferred-Identity to Default in the Advanced tab for correct caller ID behaviour.

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