# FreePBX 17 (PJSIP)

> Connect FreePBX 17 (Asterisk, chan_pjsip) to an Edesy SIP trunk for outbound calls to Indian numbers and inbound calls on your Edesy DID, with NAT, codec and dial-pattern settings.

Source: https://edesy.in/docs/sip-trunking/guides/freepbx-17-pjsip

To connect FreePBX 17 to Edesy, create a credentials termination trunk in the Edesy portal, then add a `chan_pjsip` trunk in FreePBX that registers to `tk-<id>.trunk.edesy.in` on port `5070`. Add an outbound route that sends numbers in E.164 and an inbound route for your DID. Use only the ulaw and alaw codecs.

## 1. Collect your connection details

From the trunk page in the Edesy portal (create a **SIP credentials** trunk if you have not yet):

| Setting | Where to find it | Example |
|---|---|---|
| SIP server | **Integration snippets → SIP server** | `tk-1a2b3c4d.trunk.edesy.in` |
| Port | Fixed | `5070` |
| Transport | Fixed | UDP (TCP also works) |
| Username | **Authentication → Username** | `user-1a2b3c4d` |
| Password | **Authentication → Reveal** | (from the portal) |
| Caller ID | **Phone Numbers** | `+91XXXXXXXXXX` |
| Edesy signalling IP | The **FreePBX 17** snippet on the trunk page, `Match (Permit)` line | (from the portal) |
| Max concurrent calls | Trunk page | `50` |

The trunk page's **FreePBX 17 (chan_pjsip trunk)** snippet has these values filled in for your trunk.

## 2. Prerequisites

- FreePBX 17 with the PJSIP driver (the default with Asterisk 21).
- Your FreePBX server can reach the Edesy signalling IP on UDP `5070`, and your RTP port range is open.
- If FreePBX is behind NAT, you know your public IP. See step 6.

## 3. Create the trunk

**Connectivity → Trunks → Add Trunk → Add SIP (chan_pjsip) Trunk**

**General tab**

| Field | Value |
|---|---|
| Trunk Name | `Edesy` |
| Outbound CallerID | Your Edesy number, such as `+91XXXXXXXXXX` |
| Maximum Channels | Same as the trunk's max concurrent calls, such as `50` |

**pjsip Settings → General**

| Field | Value |
|---|---|
| Username | `user-1a2b3c4d` |
| Secret | The revealed password |
| Authentication | Outbound |
| Registration | Send |
| SIP Server | `tk-1a2b3c4d.trunk.edesy.in` |
| SIP Server Port | `5070` |
| Context | `from-pstn` |
| Transport | `0.0.0.0-udp` |

**pjsip Settings → Advanced**

| Field | Value |
|---|---|
| Contact User | `user-1a2b3c4d` |
| From User | `user-1a2b3c4d` |
| From Domain | `tk-1a2b3c4d.trunk.edesy.in` |
| Client URI | `sip:user-1a2b3c4d@tk-1a2b3c4d.trunk.edesy.in:5070` |
| Server URI | `sip:tk-1a2b3c4d.trunk.edesy.in:5070` |
| AOR Contact | `sip:tk-1a2b3c4d.trunk.edesy.in:5070` |
| Match (Permit) | The Edesy signalling IP from your trunk snippet |
| Qualify Frequency | `60` |
| DTMF Mode | RFC 4733 |

`Match (Permit)` tells FreePBX to trust INVITEs from Edesy and send them into `from-pstn`. It is needed for inbound calls.

**pjsip Settings → Codecs:** enable **ulaw** and **alaw** only. Disable g722, g729, opus and the rest.

Click **Submit**, then **Apply Config**.

## 4. Outbound route

**Connectivity → Outbound Routes → Add Outbound Route**

| Field | Value |
|---|---|
| Route Name | `Edesy-Out` |
| Trunk Sequence for Matched Routes | `Edesy` |

Dial patterns: send numbers in E.164 (country code, no `+`). For India:

| prepend | prefix | match pattern | Meaning |
|---|---|---|---|
| `91` | | `XXXXXXXXXX` | 10-digit mobile: add `91` |
| | | `91XXXXXXXXXX` | Already has the country code: pass through |

Apply the config and dial a mobile number from an extension.

> **Caller ID:** The caller ID must be your Edesy number. If FreePBX sends the SIP username as the caller ID, Edesy rejects the call with `403 Caller ID Not Authorized`. Set **Outbound CallerID** on the trunk or route.

## 5. Inbound route (optional)

Inbound has two halves:

1. **On Edesy:** route your number to this trunk (**Phone Numbers → your number → Inbound routing → Termination Trunk**). Because the trunk registers, Edesy delivers calls to your FreePBX once it shows **Registered**.
2. **On FreePBX:** **Connectivity → Inbound Routes → Add Inbound Route**. Set **DID Number** to `91XXXXXXXXXX` (try with and without a leading `+`) and pick a destination: an IVR, ring group or extension.

## 6. NAT and firewall

An on-premises FreePBX is almost always behind NAT. In **Settings → Asterisk SIP Settings → chan_pjsip**:

- **External Address:** your site's public IP or dynamic-DNS name.
- **Local Networks:** your LAN subnets, such as `192.168.0.0/24`.

On your firewall or router:

- Allow outbound UDP to the Edesy signalling IP on port `5070`, and your RTP range.
- Allow inbound UDP from the Edesy signalling IP for SIP and RTP.
- Forward the RTP range (Asterisk default `10000-20000/udp`) to FreePBX if it is not the edge device.
- Turn off SIP ALG on the router. See [SIP ALG](https://edesy.in/docs/sip-trunking/troubleshooting/sip-alg).

A correct **External Address** is the usual fix for one-way or no audio.

## 7. Verify

- **Registration:** **Reports → Asterisk Info → Registrations** shows the Edesy trunk as Registered, and the trunk page in the Edesy portal shows **Registered**.
- **Outbound:** dial a mobile from an extension.
- **Edesy side on its own:** use the outbound test on the trunk page. See [Test Calls](https://edesy.in/docs/sip-trunking/inbound/test-call).

## Troubleshooting

| Symptom | Likely cause and fix |
|---|---|
| Not registered | Wrong password (reveal or rotate it again); port `5070` blocked; Server URI missing `:5070` |
| Outbound `401` / `403` | Authentication not set to Outbound; username or secret mismatch; From User and From Domain not set |
| `403 Caller ID Not Authorized` | Caller ID is not your Edesy number |
| Outbound `404` or not routed | Number not in E.164; fix the dial-pattern prepend (`91` for India) |
| One-way or no audio | NAT: set External Address and Local Networks; open RTP; ulaw/alaw only; disable SIP ALG |
| Inbound never rings | Number not routed to the trunk on Edesy; missing Match (Permit); DID format in the inbound route |

Have your trunk name and trunk ID ready when you contact support.

## FAQ

### Does this work with plain Asterisk or other Asterisk-based PBXs?

Yes. The same settings apply to any `chan_pjsip` setup: registration to the trunk host on port `5070`, outbound auth, G.711 codecs and an identify/permit for the Edesy IP.

### Can I use IP authentication instead of registration?

Yes, if your PBX has a static public IP. Create an IP allowlist trunk, set Registration to None, and route your number to the trunk with a trunk-level inbound SIP URI pointing at your PBX.

### Which codecs should I enable?

ulaw and alaw only. Other codecs can cause negotiation failures.

### How do I move from a PRI line to SIP?

Keep your FreePBX dialplan, add the Edesy trunk as above, and change the outbound route's trunk sequence. Buy Edesy numbers for inbound calls.