# Yeastar S-Series SIP Trunk Configuration

**Category:** PBX Setup Guides
**URL:** https://www.simtex.com.au/support/kb/pbx-setup-guides/yeastar-s-series-sip-trunk-configuration
**Published:** 2026-03-30

Step-by-step guide to configuring a Simtex SIP trunk on Yeastar S-Series PBX systems (S20, S50, S100, S300) using TCP transport.

## Why this article matters

IT teams maintaining or refreshing Yeastar S-Series appliances (S20, S50, S100, S300) typically reach this guide when the existing analogue or ISDN service is being decommissioned as part of an NBN migration and a SIP trunk is needed to keep the on-premise PBX in service. The configuration uses a Register Trunk over TCP transport to siptcp.simtex.com.au, with G.711 codecs and standard inbound and outbound dial patterns for Australian numbering. The trunk service is Simtex SIP Trunks, priced from $4.99 per channel per month from geo-redundant Perth and Sydney POPs on a 99.99% uptime SLA. Because Simtex operates as the licensed carrier rather than reselling capacity, Yeastar S-Series deployments avoid the registration timeouts and one-way audio issues that often surface when transiting multiple wholesale layers. When comparing carriers for Yeastar S-Series, Australian businesses typically weigh Simtex against reseller offerings from Vonex, Maxotel, Buroserv, and Intelephony, and against incumbent SIP Connect services from Telstra Business and iiNet Business. Direct carrier ownership, transparent per-channel pricing, and Australian-based engineering support for trunk registration troubleshooting are the points of difference. Existing numbers can be ported in on a 10-business-day Category A timeline, and the trunk can later extend into Microsoft Teams via Simtex Teams Direct Route on the same account.

## Simtex SIP Trunks from $4.99/month

Business-grade SIP trunking with geo-redundant infrastructure, crystal-clear audio, and Australian-based support. Pay-as-you-go or unlimited plans available.

- [View SIP Trunk Plans](/products/sip-trunks)

## Overview

This guide walks you through configuring a Simtex SIP trunk on a **Yeastar S-Series PBX** (S20, S50, S100, S300) using a **VoIP Register Trunk** with TCP transport. The S-Series is Yeastar's established line of compact IP-PBX appliances with a straightforward web-based administration interface.

Yeastar S-Series PBX systems are reliable, cost-effective IP-PBX appliances ideal for small to mid-sized businesses — [yeastar.com](https://www.yeastar.com/s-series-voip-pbx/)

### What you'll need

- Your Simtex account number (e.g. `214XXXXXXX`)
- Your Simtex SIP password
- Your allocated DIDs (phone numbers)
- Yeastar S-Series PBX with web GUI access

> **Info:**
> **SIP Servers:**
> - **West Coast (AU):** `siptcp.simtex.com.au`
> - **East Coast (AU):** `siptcpeast.simtex.com.au`
> 
> Choose the server closest to your PBX. The S-Series supports DNS-NAPTR transport for automatic server discovery and failover.

> **Warning:**
> **Do not expose your PBX to the public internet.** In 99.9% of cases you do not need any ports forwarded on your router or firewall to connect to Simtex. If you are unsure, [speak to us](/contact) first.

> **Note:**
> **S-Series vs P-Series:** If you have a Yeastar P-Series (P550/P560/P570), use our [Yeastar P-Series guide](/support/kb/pbx-setup-guides/yeastar-p-series-sip-trunk-configuration) instead. The P-Series has a different menu layout and updated terminology.

## Step 1 — Create the SIP Trunk
1. Navigate to **Settings → PBX → Trunks**
2. Click **Add**
3. For **Select Country**, select **General** (Simtex is not in the pre-configured ITSP template list for the S-Series)
4. Set **Trunk Type** to **Register Trunk**

> **Note:**
> **Trunk types explained:** The S-Series calls these **VoIP Register Trunk**, **VoIP Peer Trunk**, and **VoIP Account Trunk**. Select **Register Trunk** for Simtex — the PBX authenticates with your account credentials and registers with our SIP servers.

## Step 2 — Trunk Settings

Configure the connection details for Simtex:

### Basic Settings

```
`Name:                Simtex-SIP
Trunk Status:        Enabled
Trunk Type:          Register Trunk
Protocol:            SIP
Transport:           TCP
Hostname/IP:         siptcp.simtex.com.au
Domain:              siptcp.simtex.com.au
Username:            214XXXXXXX
Password:            XXXXXXXX
Authentication Name: 214XXXXXXX
From User:           214XXXXXXX
Caller ID Number:    (your main DID, e.g. 61894883344)
Caller ID Name:      (your business name)`
```

> **Tip:**
> **DNS-NAPTR for automatic failover:** Instead of selecting **TCP** as the transport, you can select **DNS-NAPTR** and set the port to `0`. The PBX will query DNS SRV/NAPTR records to automatically discover the correct server, port, and transport — providing automatic failover across our geo-redundant server farms.

> **Note:**
> **East Coast?** Replace `siptcp.simtex.com.au` with `siptcpeast.simtex.com.au` in both the Hostname/IP and Domain fields if your PBX is located on the East Coast of Australia (NSW, VIC, QLD).

## Step 3 — Advanced Settings

Click the **Advanced** tab and configure the following:

### DID Number

Enter your DID numbers. Click **+** to add each DID in E.164 format without the `+` prefix (e.g. `61894883344`). Tick **DNIS Name** to add a friendly label for each DID.

### VoIP Settings

```
`Qualify:             Enabled
DTMF Mode:           RFC4733 (RFC2833)
T.38 Support:        Disabled (unless you need fax)
Enable RTP Keep-alive: Enabled
Maximum Channels:    (your purchased channel count)`
```

### Inbound Parameters

```
`Get Caller ID From:  P-Asserted-Identity
Get DID From:        Follow System`
```

### Outbound Parameters

```
`P Asserted Identity: Default
Remote Party ID:     Default`
```

> **Warning:**
> **Realm field:** The S-Series has a **Realm** field that defaults to `YSAsterisk`. **Clear this field** (leave it blank) to avoid registration failures. An incorrect Realm value will cause your trunk to fail authentication with a 401/407 error.

> **Tip:**
> **Channel security:** Set **Maximum Channels** to the number of concurrent calls you actually need. This acts as a hard limit — if your system is compromised, attackers can't exceed this call count.

## Step 4 — Codec Configuration

Click the **Codec** tab within the trunk settings.

Move the following codecs from the **Available** list to the **Selected** list. Use the priority buttons to set this order:

```
`Priority 1:  G711 a-law  (alaw)
Priority 2:  G711 u-law  (ulaw)`
```

Remove any unnecessary codecs from the Selected list — a cleaner codec list speeds up call negotiation.

Click **Save and Apply** to save the trunk configuration.

> **Info:**
> **G.711 A-Law** is the standard codec for Australian PSTN interconnection. It provides toll-quality audio at 64kbps with zero transcoding overhead. Always place it first.

## Step 5 — Outbound Routes

Outbound routes tell your PBX which numbers are allowed to be dialled externally and which trunk to use.
1. Navigate to **Settings → PBX → Call Control → Outbound Routes**
2. Click **Add**
3. Enter a **Name** (e.g. “Simtex Outbound”)
4. Under **Member Trunks**, select your **Simtex-SIP** trunk
5. Under **Member Extensions**, select the extensions that should be allowed to make external calls
6. Configure **Dial Patterns** for Australian numbering (see table below)
7. Click **Save and Apply**

## Recommended Dial Patterns for Australian Numbering

| Description | Dial Pattern | Strip | Prepend | Example Dialled |
| --- | --- | --- | --- | --- |
| Local (8-digit with area code) | `XXXXXXXX` | 0 |  | 94883344 |
| National / Mobile (leading 0) | `0XXXXXXXXX` | 0 |  | 0412345678 |
| Emergency | `000` | 0 |  | 000 |
| 1300 / 1800 Numbers | `1[38]00XXXXXX` | 0 |  | 1300888519 |
| 13 Numbers (6-digit) | `13XXXX` | 0 |  | 131313 |
| International (0011) | `0011.` | 0 |  | 001161894883344 |

> **Warning:**
> **International dialling:** If you do not need international calls, simply leave out the `0011` dial pattern. This is the simplest way to prevent unauthorised international dialling and potential toll fraud.

> **Note:**
> **Wildcard syntax:** Yeastar uses Asterisk dial pattern notation. `X` matches any digit 0–9, `Z` matches 1–9, `N` matches 2–9, and `.` matches one or more characters.

## Step 6 — Inbound Routes

Inbound routes direct incoming calls on your DIDs to the correct destination.
1. Navigate to **Settings → PBX → Call Control → Inbound Routes**
2. Click **Add**
3. Enter a **Name** (e.g. “Main Line”)
4. Under **Member Trunks**, select your **Simtex-SIP** trunk
5. Under **DID Pattern**, enter the DID number in E.164 format (e.g. `61894883344`) — or leave blank for a catch-all route
6. Set the **Destination** to the target extension, ring group, IVR, or queue
7. Click **Save and Apply**

### Create a catch-all route first

Leave the **DID Pattern** blank to create a catch-all route. This handles any inbound DID that doesn't have a specific route — point it at your receptionist or main ring group.

### Then add DID-specific routes

Create additional inbound routes for individual DIDs or DID ranges that need specific routing.

## Testing Your Trunk

Once configured, verify everything works:
1. **Check trunk status** — navigate to **Settings → PBX → Trunks** and confirm the trunk shows a green “Registered” status
2. **Make an outbound call** — dial an external number from an extension and confirm two-way audio
3. **Receive an inbound call** — call one of your DIDs from a mobile and confirm it routes correctly
4. **Verify caller ID** — check your outbound caller ID displays correctly on the receiving end
5. **Check voicemail** — leave a voicemail to ensure DTMF tones are working correctly through the trunk

> **Note:**
> **Trunk not registering?** Check these common issues:
> - Verify your account number (`214XXXXXXX`) and password — copy/paste to avoid typos
> - Ensure the **Realm** field is blank (not the default `YSAsterisk`)
> - Confirm the Transport is set to **TCP** on the trunk
> - Check your firewall allows outbound TCP connections
> - Try the alternate server (`siptcpeast` / `siptcp`) in case of regional issues
> - Verify DNS resolution: the hostname must resolve to a valid SIP server

## Security Hardening

After configuring your trunk, take these essential steps to secure your Yeastar PBX:

### Disable SIP Guest Calls
1. Navigate to **Settings → PBX → General → SIP**
2. Scroll to the Advanced SIP settings
3. Set **Allow Guest** to **No**

### Additional security measures

- **Strong extension passwords** — use the auto-generated passwords; do not simplify them
- **IP auto-defense** — Yeastar includes built-in brute-force protection; ensure it is enabled under **Security → Security Rules**
- **Firmware updates** — keep your PBX firmware up to date via **Maintenance → Upgrade**
- **Channel limits** — keep Maximum Channels on the trunk set to your actual requirement
- **Disable unused services** — turn off SSH, FTP, or other remote access protocols you don't use

> **Warning:**
> **Always disable SIP Guest calls on production systems.** Leaving this enabled allows unauthenticated callers to place calls through your PBX — one of the most common causes of toll fraud.

## Extension-Level Caller ID

By default, outbound calls present the Caller ID Number set on the trunk. To override per extension:
1. Navigate to **Settings → PBX → Trunks**
2. Edit your **Simtex-SIP** trunk
3. Under the **Adapt Caller ID** tab, configure rules to modify the caller ID for specific extensions or patterns

Alternatively, configure the trunk's **Caller ID Number** field with your main DID, and use outbound route overrides for specific routing needs.

The DID must be allocated to your Simtex account — you cannot present arbitrary numbers.

> **Info:**
> **Caller ID validation:** Simtex validates outbound caller IDs against your account. Only DIDs allocated to your trunk will be transmitted — any other number is replaced with your main trunk number automatically.

## Common Questions

**Q: Should I use Register Trunk or Peer Trunk for Simtex?**

A: **Register Trunk.** Simtex uses credential-based authentication. Your PBX registers with our SIP servers using your account number and password. Peer Trunk is only used when a provider authenticates by IP address.

**Q: What is the Realm field and should I change it?**

A: The Realm field defaults to `YSAsterisk` on the S-Series. You should **clear this field** (leave it blank) when connecting to Simtex. An incorrect Realm value causes SIP authentication failures (401/407 errors).

**Q: What is DNS-NAPTR transport?**

A: DNS-NAPTR is a transport option that allows the PBX to query DNS records to automatically discover the correct SIP server, port, and transport protocol. Select DNS-NAPTR and set port to `0` for automatic failover across Simtex's geo-redundant server farms.

**Q: What's the difference between the S-Series and P-Series?**

A: The S-Series uses a different web interface layout. The trunk menu is at **Settings → PBX → Trunks** (vs **Extension and Trunk → Trunk** on P-Series). The S-Series calls them “VoIP Register Trunk” while the P-Series uses “SIP Register Trunk”. The S-Series also has a **From User** field and a **Realm** field that the P-Series handles differently. Core trunk functionality is the same.

**Q: What format should I use for DID numbers?**

A: Enter DIDs in E.164 format without the `+` prefix. For example, `61894883344` not `+61894883344` or `894883344`.

**Q: I'm getting one-way audio. How do I fix it?**

A: One-way audio is almost always a NAT issue. Check: (1) NAT settings under **Settings → PBX → General → SIP → NAT** — set the correct External IP or DDNS hostname. (2) Enable **RTP Keep-alive** on the trunk's Advanced tab. (3) Ensure your router's SIP ALG is **disabled**. (4) Check that RTP ports (10000-12000) are not being blocked by your firewall.

## Need Help?

If you run into any issues configuring your Yeastar S-Series trunk, our support team can verify your trunk registration status from our side and assist with troubleshooting.

**[Contact Simtex Support](/contact)**

## Related

- Article: https://www.simtex.com.au/support/kb/pbx-setup-guides/yeastar-s-series-sip-trunk-configuration
- Category: https://www.simtex.com.au/support/kb/pbx-setup-guides
- Knowledge Base: https://www.simtex.com.au/support/kb
- Full AI reference: https://www.simtex.com.au/llms-full.txt