# Grandstream UCM SIP Trunk Configuration

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

Step-by-step guide to configuring a Simtex SIP trunk on Grandstream UCM series PBX systems (UCM6202, UCM6204, UCM6208, UCM6301, UCM6302, UCM6304, UCM6308).

## Why this article matters

Businesses running Grandstream UCM6200 or UCM6300 series appliances (UCM6202, UCM6204, UCM6208, UCM6301, UCM6302, UCM6304, UCM6308) typically reach this guide when evaluating SIP trunk providers for an in-house IP-PBX rather than a hosted phone system. The configuration uses a Register SIP Trunk over TCP transport to siptcp.simtex.com.au, terminating calls directly onto the Simtex carrier network without an intermediary reseller. The service referenced is Simtex SIP Trunks, priced from $4.99 per channel per month with channel counts scaling from 2 to 200+ on a single trunk. Delivery is from geo-redundant Perth and Sydney points of presence on a 99.99% uptime SLA, with documented interoperability against the Grandstream UCM firmware. Grandstream UCM is frequently selected as a lower-cost alternative to 3CX or FreePBX commercial editions, so the SIP carrier decision is often where the savings are realised or lost. Compared with reseller SIP providers such as Vonex, Maxotel, SIPTalk, and Intelephony, Simtex provides direct carrier provisioning, Australian-based engineering support, and a single porting pipeline for DIDs and 1300/1800 inbound numbers. The same trunk can be extended to Microsoft Teams via Simtex Teams Direct Route if the business later adds Teams calling alongside the Grandstream deployment.

## 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 **Grandstream UCM series** IP-PBX using a **Register SIP Trunk** with TCP transport. This guide covers both the **UCM6200 series** (UCM6202, UCM6204, UCM6208) and the newer **UCM6300 series** (UCM6301, UCM6302, UCM6304, UCM6308).

Grandstream UCM PBX systems are feature-rich, competitively priced IP-PBX appliances with a comprehensive web GUI — [grandstream.com](https://www.grandstream.com/products/ip-pbxs)

### What you'll need

- Your Simtex account number (e.g. `214XXXXXXX`)
- Your Simtex SIP password
- Your allocated DIDs (phone numbers)
- Grandstream UCM 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 UCM. Grandstream UCM supports TCP and TLS transport options on the Advanced Settings tab.

> **Warning:**
> **Do not expose your UCM 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.

> **Tip:**
> **UCM RemoteConnect users:** If you manage your UCM6300 remotely via Grandstream's UCM RemoteConnect / GDMS cloud portal, the SIP trunk configuration is identical. RemoteConnect provides remote access to the same web GUI — SIP trunks connect directly from the UCM to Simtex regardless of how you access the admin panel.

## Step 1 — Create the SIP Trunk
1. Log into your UCM web GUI
2. Navigate to **Extension/Trunk → VoIP Trunks**
3. Click **Create New SIP Trunk** (or **+ Add SIP Trunk**)
4. Set **Type** to **Register SIP Trunk**

> **Note:**
> **UCM6300 menu path:** On some UCM6300 firmware versions, the menu path may be **PBX → Basic/Call Routes → VoIP Trunks** instead. Both paths lead to the same configuration page.

> **Note:**
> **Trunk types explained:** Grandstream offers two trunk types. **Register SIP Trunk** is the correct choice for Simtex — the UCM authenticates with your account credentials and registers with our SIP servers. *Peer SIP Trunk* is for IP-based authentication (no credentials) and is used for interconnecting PBX systems or providers that whitelist your IP.

## Step 2 — Basic Settings

Configure the connection details on the **Basic Settings** tab:

```
`Type:                Register SIP Trunk
Provider Name:       Simtex-SIP
Host Name:           siptcp.simtex.com.au
Username:            214XXXXXXX
Password:            XXXXXXXX
Auth ID:             214XXXXXXX
Caller ID:           61894883344  (your main DID)
Caller ID Name:      (your business name)
Keep Trunk CID:      Checked
NAT:                 Checked  (if UCM is behind a router)
Need Registration:   Checked
Maximum Channels:    (your purchased channel count)`
```

> **Tip:**
> **Keep Trunk CID:** When checked, the trunk's own Caller ID is used for all outbound calls on this trunk. When unchecked, individual extensions can send their own caller ID. Check this initially to ensure your main DID is always presented, then uncheck later if you need per-extension caller ID.

> **Warning:**
> **Save before Advanced:** You must click **Save** on the Basic Settings first before the Advanced Settings tab becomes available for editing. After saving, click **Apply Changes** at the top of the page.

> **Note:**
> **East Coast?** Replace `siptcp.simtex.com.au` with `siptcpeast.simtex.com.au` in the Host Name field if your UCM is located on the East Coast of Australia (NSW, VIC, QLD).

## Step 3 — Advanced Settings

After saving the Basic Settings, click the **edit** icon on your trunk, then click the **Advanced Settings** tab:

### Transport and Proxy

```
`Transport:           TCP
From Domain:         siptcp.simtex.com.au
From User:           214XXXXXXX`
```

### Codec Preference

Set the codec preference list. Select and reorder:

```
`Priority 1:  PCMA  (G.711 A-Law / alaw)
Priority 2:  PCMU  (G.711 U-Law / ulaw)`
```

Remove any unnecessary codecs from the enabled list.

### DTMF and Signalling

```
`DTMF Mode:                  RFC2833
Enable Heartbeat Detection: Checked
Heartbeat Frequency:        60 seconds
Send PAI Header:            Checked`
```

> **Info:**
> **G.711 A-Law (PCMA)** is the standard codec for Australian PSTN interconnection. It provides toll-quality audio at 64kbps with zero transcoding overhead. Always place it first. Grandstream uses `PCMA` and `PCMU` as the codec names.

> **Warning:**
> **PPI and PAI are mutually exclusive.** You cannot enable both **Send PPI Header** and **Send PAI Header** at the same time on a Grandstream UCM. Only one of these SIP identity headers can be active per trunk. For Simtex, enable **Send PAI Header** (P-Asserted-Identity).

> **Tip:**
> **Heartbeat Detection** sends SIP OPTIONS packets at the configured interval to monitor trunk availability. This helps the UCM detect a trunk failure quickly and trigger failover to a backup trunk if configured.

Click **Save**, then **Apply Changes** at the top of the page. The trunk should register within a few seconds.

## Step 4 — Outbound Routes

Outbound routes (calling rules) tell your UCM which numbers are allowed to be dialled externally and which trunk to use.
1. Navigate to **Extension/Trunk → Outbound Routes**
2. Click **+ Add**
3. Enter a **Calling Rule Name** (e.g. “Simtex National”)
4. Under **Use Trunk**, select your **Simtex-SIP** trunk
5. Set the **Pattern** for the numbers this rule handles (see table below)
6. Set **Privilege Level** appropriately (ensure extensions have matching or higher privilege)
7. Click **Save**, then **Apply Changes**

> **Note:**
> **Route ordering matters.** The UCM processes outbound routes from top to bottom and uses the first matching pattern. Use the arrow buttons to reorder routes so more specific patterns (like emergency `000`) are above general catch-all patterns.

## Recommended Dial Patterns for Australian Numbering

| Description | Pattern | Strip | Prepend | Privilege | Example |
| --- | --- | --- | --- | --- | --- |
| Local (8-digit) | `_XXXXXXXX` | 0 |  | Local | 94883344 |
| National / Mobile | `_0XXXXXXXXX` | 0 |  | National | 0412345678 |
| Emergency | `000` | 0 |  | Internal | 000 |
| 1300 / 1800 Numbers | `_1[38]00XXXXXX` | 0 |  | National | 1300888519 |
| 13 Numbers (6-digit) | `_13XXXX` | 0 |  | National | 131313 |
| International (0011) | `_0011.` | 0 |  | International | 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. You can also restrict this by setting the **Privilege Level** to `International` and only granting that privilege to authorised extensions.

> **Note:**
> **Pattern prefix:** Grandstream dial patterns must be prefixed with an underscore `_` when using wildcards. `X` matches any digit 0–9, `N` matches 2–9, `Z` matches 1–9, and `.` matches one or more characters.

## Step 5 — Inbound Routes

Inbound routes direct incoming calls on your DIDs to the correct destination.
1. Navigate to **Extension/Trunk → Inbound Routes**
2. Click **+ Add**
3. Under **Trunks**, select your **Simtex-SIP** trunk
4. Under **Pattern** (DID Pattern), enter the DID number in E.164 format without `+` (e.g. `61894883344`) — or leave blank for a catch-all route
5. Set the **Default Destination** to the target extension, ring group, IVR, queue, or other destination
6. Click **Save**, then **Apply Changes**

### Create a catch-all route first

Leave the **Pattern** field 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 that need specific routing.

## Step 6 — Per-Extension Caller ID (DOD)

To assign different outbound caller IDs to different extensions using Direct Outward Dialling (DOD):
1. Navigate to **Extension/Trunk → VoIP Trunks**
2. Click your **Simtex-SIP** trunk to edit it
3. Click **+ Add DOD**
4. Enter the **DOD Number** (one of your DIDs in E.164 format)
5. Enter a **DOD Name** (friendly label)
6. Select the **Available Extensions** that should use this DOD number
7. Click **Save**, then **Apply Changes**

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.

## Testing Your Trunk

Once configured, verify everything works:
1. **Check trunk status** — navigate to **System Status → Dashboard** and check the **Trunks** widget. A blue dot indicates the trunk is registered
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 **Transport** is set to **TCP** on the Advanced Settings tab
> - Confirm the **From Domain** matches the Host Name (`siptcp.simtex.com.au`)
> - Check that **Need Registration** is ticked on the Basic Settings tab
> - Check your firewall allows outbound TCP connections
> - Try the alternate server (`siptcpeast` / `siptcp`) in case of regional issues

## Security Hardening

After configuring your trunk, take these essential steps to secure your Grandstream UCM:

### Change Default Credentials

If you haven't already, change the default admin password immediately. The factory default is `admin` / `admin`.

### Firewall and Access Control

- **Restrict web GUI access** — use the UCM's built-in firewall rules to limit admin access to trusted IPs
- **Disable unused protocols** — turn off SSH, Telnet, and HTTP (use HTTPS only) if not needed
- **SIP/RTP ports** — ensure only the necessary SIP (5060) and RTP (10000-20000) ports are accessible

### Additional security measures

- **Strong extension passwords** — use complex, auto-generated passwords for all extensions
- **Firmware updates** — keep your UCM firmware current via **Maintenance → Upgrade**
- **Channel limits** — keep Maximum Channels on the trunk set to your actual requirement
- **Failover trunk** — configure a failover trunk in outbound routes for resilience

> **Warning:**
> **Change default passwords immediately.** Grandstream UCM ships with the well-known default credentials `admin` / `admin`. Leaving these unchanged is one of the most common causes of PBX compromise.

## Common Questions

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

A: **Register SIP Trunk.** Simtex uses credential-based authentication. Your UCM registers with our SIP servers using your account number and password. Peer SIP Trunk is only for IP-based authentication where the provider whitelists your public IP.

**Q: Does this guide apply to both UCM6200 and UCM6300 series?**

A: Yes. The core SIP trunk configuration is identical across both series. The UCM6300 may show slightly different menu paths on some firmware versions (**PBX → Basic/Call Routes → VoIP Trunks** instead of **Extension/Trunk → VoIP Trunks**), but the fields and options are the same.

**Q: Does UCM RemoteConnect affect SIP trunk configuration?**

A: No. UCM RemoteConnect provides remote access to your UCM's web GUI via Grandstream's cloud relay. The SIP trunk connects directly from the UCM's network to Simtex — RemoteConnect is only involved in remote extension connectivity, not trunk-side calls. The configuration interface is identical whether accessed locally or via RemoteConnect.

**Q: What is the difference between Send PPI Header and Send PAI Header?**

A: PPI (P-Preferred-Identity) and PAI (P-Asserted-Identity) are SIP headers for caller ID transmission. They **cannot both be enabled** on the same trunk. For Simtex, enable **Send PAI Header** for correct caller ID passthrough.

**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 on Grandstream UCM is almost always a NAT misconfiguration. Check: (1) **NAT** is ticked on the trunk's Basic Settings. (2) Go to **PBX Settings → SIP Settings → NAT** and set the correct **External Host** (your public IP or DDNS) and **Local Network** (your LAN subnet). (3) Ensure your router's SIP ALG is **disabled**.

## Need Help?

If you run into any issues configuring your Grandstream UCM 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/grandstream-ucm-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