---
title: Use TelecomsXchange with Amazon Connect
url: https://help.telecomsxchange.com/integrations/amazon-connect/
audience: Integrations
summary: 'Amazon Connect can''t use an outside carrier as its trunk: its inbound and
  outbound PSTN calls go through numbers claimed or ported into Connect, and the Chime
  SDK Voice Connector is AWS''s own SIP trunk, not a bridge to another carrier. To
  use TelecomsXchange numbers, deliver them to a SIP server you run (Asterisk, FreeSWITCH
  or an SBC) and forward each call to a number claimed in Connect, through a TelecomsXchange
  route over UDP.'
updated: '2026-10-23'
verified_commit: 83b4ebc96d0081a25213776d5a8871bd04941125
---

# Use TelecomsXchange with Amazon Connect

> Amazon Connect can't use an outside carrier as its trunk: its inbound and outbound PSTN calls go through numbers claimed or ported into Connect, and the Chime SDK Voice Connector is AWS's own SIP trunk, not a bridge to another carrier. To use TelecomsXchange numbers, deliver them to a SIP server you run (Asterisk, FreeSWITCH or an SBC) and forward each call to a number claimed in Connect, through a TelecomsXchange route over UDP.

## TelecomsXchange numbers into Amazon Connect

1. **Caller**: Dials a number bought on TCXC
2. **TelecomsXchange** (via `14165551234`): From 174.136.44.213, UDP
3. **Your SIP server** (via `sip:14165551234@203.0.113.10`): Asterisk, FreeSWITCH or SBC
4. **Amazon Connect** (via `80039#18005550100`): Number claimed in Connect

Amazon Connect is a cloud contact centre with its own telephony. Unlike Genesys Cloud or Twilio, it has **no bring-your-own-carrier SIP trunk**: you can't point it at sip01, and TelecomsXchange can't deliver a SIP call straight into a Connect instance. This page explains what does work with TelecomsXchange, which only speaks SIP over UDP. For the full list of TelecomsXchange values, see [Connect a SIP platform](https://help.telecomsxchange.com/integrations/connect-sip-platforms/).

!!! warning
    TelecomsXchange does not accept SIP REGISTER, and supports SIP over **UDP only**: no TCP, TLS or SRTP. It can't be the TLS endpoint for any AWS voice service. Every pattern below keeps a SIP server you run between TelecomsXchange and AWS.

## How do Amazon Connect, Voice Connector and TelecomsXchange fit together?

| AWS service | What it is | Can it carry TelecomsXchange traffic? |
|---|---|---|
| Amazon Connect telephony | Numbers you claim or port into Connect; Connect places and receives the PSTN calls | Only as a phone-network destination: TelecomsXchange can call a Connect number |
| Amazon Chime SDK Voice Connector | AWS's SIP trunking service for your PBX or SBC: AWS numbers, **Origination** (calls to your SIP hosts) and **Termination** (calls from your allowed hosts to the PSTN) | No. It's another carrier, not a bridge from TelecomsXchange into Connect |
| Amazon Connect external voice transfer connector | Lets a Connect flow hand a live call and its data to your on-premises voice system (an SBC or PBX) without going through the PSTN | Only after a call is already in Connect; it isn't Connect's carrier trunk |

So the working pattern is: **TelecomsXchange ↔ your SIP server over UDP**, and **your SIP server ↔ Amazon Connect over the phone network**, using numbers claimed in Connect.

## Before you start

- A [SIP account](https://help.telecomsxchange.com/buyers/create-your-first-account/) with your SIP server's public IP, for example `203.0.113.10`, or a VoIP login and password from **Edit Account → Add SIP Digest Auth**.
- A [purchased route](https://help.telecomsxchange.com/buyers/find-and-buy-routes/) that reaches the country of your Connect number, and its tech prefix from **My Interconnections**, for example `80039#`.
- A SIP server you run on a public IP: Asterisk, FreeSWITCH, Kamailio or a commercial SBC. The [Asterisk](https://help.telecomsxchange.com/integrations/asterisk/) and [FreeSWITCH](https://help.telecomsxchange.com/integrations/freeswitch/) guides cover the TelecomsXchange side in full.
- An Amazon Connect instance with a claimed number, for example `+18005550100`, attached to a contact flow.
- [Phone numbers](https://help.telecomsxchange.com/buyers/virtual-numbers-dids/) bought on TelecomsXchange, if callers should dial TelecomsXchange numbers.

## How do I send calls to TelecomsXchange?

Amazon Connect itself can't. Its outbound calls, agent calls and outbound campaigns use Connect telephony and the numbers claimed in Connect.

What you can do is send calls *towards* Connect through TelecomsXchange: your SIP server dials a Connect number through a purchased route, exactly as it dials any other number (`TECH_PREFIX` + the Connect number in E.164 without `+`). In Asterisk, with the `tcxc` endpoint from the [Asterisk guide](https://help.telecomsxchange.com/integrations/asterisk/), define both as globals; the dial plan in the next section uses them:

```ini
; /etc/asterisk/extensions.conf
[globals]
TECH_PREFIX=80039#            ; route from My Interconnections
CONNECT_NUMBER=18005550100    ; number claimed in Amazon Connect, E.164 without +
```

If a Connect flow transfers calls to your own voice system with the external voice transfer connector, the receiving SBC or PBX can route those calls on to TelecomsXchange like any other outbound call. The Connect-to-SBC leg follows AWS's requirements (AWS recommends turning on encryption for it); the SBC-to-TelecomsXchange leg is plain SIP over UDP to `sip01.telecomsxchange.com:5060`.

## How do I receive calls from TelecomsXchange?

Calls come from `174.136.44.213` over UDP, from source ports 5060 to 5072, with the number in E.164 without `+`, such as `14165551234`. They go to your SIP server, never to Connect directly.

1. Allow `174.136.44.213` on your SIP server's firewall for UDP 5060 and your RTP range, and match it to the TelecomsXchange trunk (the `identify` section in Asterisk, an ACL in FreeSWITCH).
2. Send the calls into a dedicated inbound context and forward them to Amazon Connect, as below.

Selling routes to TelecomsXchange that terminate in Amazon Connect isn't possible: Connect doesn't accept SIP from carriers.

## How do I receive calls on numbers I bought?

1. On TelecomsXchange, set the number's **SIP Endpoint (Primary)** in **DID Settings** to your SIP server, for example `sip:14165551234@203.0.113.10:5060`.
2. On the SIP server, forward the call to the Connect number through a TelecomsXchange route, keeping the caller's number. In Asterisk:

```ini
; /etc/asterisk/extensions.conf
[from-tcxc]
; TCXC number 14165551234 -> Amazon Connect +18005550100, via a purchased route
exten => 14165551234,1,Dial(PJSIP/${TECH_PREFIX}${CONNECT_NUMBER}@tcxc,60)
 same => n,Hangup()
```

3. In Amazon Connect, the call arrives on the claimed number and runs its contact flow. Calls on several TelecomsXchange numbers can share one Connect number; to tell them apart, forward each to its own claimed number.

!!! tip
    This adds a second call leg: you pay the TelecomsXchange number's inbound rate, the route to the Connect number, and Connect's own inbound charges. Whether the original caller ID survives depends on the route; check it on a test call. Where a route doesn't keep it, buy a CLI route in **Market View**.

## How do I test it?

1. Call the TelecomsXchange number from a mobile phone.
2. On the SIP server, watch the inbound INVITE from `174.136.44.213` and the outbound INVITE to `80039#18005550100` with `pjsip set logger on` or `sngrep port 5060`.
3. In the TelecomsXchange portal, both legs appear in **Live Calls** while the call is up, and in **Logs → CDR (Voice)** afterwards.
4. In Amazon Connect, find the contact under **Analytics and optimization → Contact search** and check the caller number and flow.

## What if something goes wrong?

| You see | What to do |
|---|---|
| 403 Forbidden on the leg to Connect | Your SIP server's public IP isn't authorised and no digest credentials were sent. Check **IP Address**, **Additional Allowed IPs** and the VoIP login on **Edit Account**. |
| 404 Not Found on the leg to Connect | Missing or wrong tech prefix, or `+` left in the Connect number. Dial `TECH_PREFIX` + `18005550100`. |
| 401 or 407 repeated | Wrong VoIP login or password in the SIP server's outbound authentication. |
| 408 or 503 | Firewall or DNS blocks sip01, or the route can't reach the Connect number's country. Check **My Interconnections** or try another route. |
| No inbound calls on the SIP server | `174.136.44.213` isn't allowed on the firewall or matched to the trunk, or the number's SIP endpoint is wrong. |
| Connect shows the wrong caller number | The route doesn't pass the caller ID. Test another route or a CLI route. |
| One-way or no audio | Your SIP server advertises a private IP in SDP, or RTP is blocked. Set the external media address and open the RTP range. |

## References

- [Amazon Connect: Set up external voice transfer to an on-premises voice system](https://docs.aws.amazon.com/connect/latest/adminguide/external-voice-transfer.html)
- [Amazon Connect: Claim a phone number](https://docs.aws.amazon.com/connect/latest/adminguide/get-connect-number.html)
- [Amazon Chime SDK: Editing Voice Connector settings (termination, origination, encryption)](https://docs.aws.amazon.com/chime-sdk/latest/ag/edit-voicecon.html)
- [Amazon Chime SDK: Voice Connectors](https://docs.aws.amazon.com/chime-sdk/latest/ag/voice-connectors.html)

## Frequently asked questions

**Q: Can Amazon Connect register to TelecomsXchange?**

A: No. TelecomsXchange is peering only and does not accept REGISTER, and Amazon Connect has no SIP trunk that could register or peer with an outside carrier in the first place.

**Q: Can I use TelecomsXchange as the carrier for Amazon Connect's outbound calls?**

A: Not directly. Amazon Connect places outbound PSTN calls through its own telephony with numbers claimed in Connect. There is no setting to send them to another carrier's SIP trunk.

**Q: Can the Amazon Chime SDK Voice Connector connect TelecomsXchange to Amazon Connect?**

A: No. A Voice Connector is AWS's own SIP trunk to the phone network, with AWS phone numbers, for your PBX or SBC. It doesn't let another carrier's numbers or routes act as Connect's telephony.

**Q: Does this need TLS?**

A: Not for the TelecomsXchange side: your SIP server talks to TelecomsXchange over UDP, the only transport TelecomsXchange supports. The hop into Amazon Connect is an ordinary call over the phone network to a Connect number.