All integration guides
Connect a SIP platform to TelecomsXchange
In short: TelecomsXchange is SIP peering only: it never accepts REGISTER. To buy, send INVITEs over UDP to sip01.telecomsxchange.com port 5060 from an authorised IP, or answer the digest challenge with the account's VoIP login, and dial the tech prefix before the E.164 number (80039#447700900123). To sell or receive numbers, accept SIP over UDP from 174.136.44.213, source ports 5060 to 5072.
Both directions through TelecomsXchange
- Your platformAuthorised IP or digest login; no REGISTER
- then, 80039#447700900123, TelecomsXchangesip01.telecomsxchange.com, UDP 5060
- then, 447700900123, Seller's switchReceives from 174.136.44.213, ports 5060-5072
On this page
TelecomsXchange connects to your platform as a SIP peer. This page collects every value an engineer needs, in both directions, and links to step-by-step guides for the most common platforms.
Warning
TelecomsXchange does not support SIP REGISTER. There is no registration to configure on any platform. Your account is recognised by the source IP address of each INVITE, or by digest authentication on the INVITE with the account's VoIP login and password. Turn registration off on the trunk or gateway that points at TelecomsXchange.
Before you start#
- A SIP account with your platform's public IP address, a VoIP login and password, or both.
- To send calls: a purchased route and its tech prefix from My Interconnections, and funds on the account.
- To receive calls on routes you sell: a route whose IP Address:Port points at your platform, and the values in Interconnect Info.
- To receive calls on numbers you bought: a phone number with its SIP endpoint set to your platform.
How do I send calls to TelecomsXchange?#
| Setting | Value |
|---|---|
| SIP host | sip01.telecomsxchange.com |
| Port and transport | 5060, UDP |
| Registration | None. Peering only. |
| Authentication | Source IP authorised on the account, and/or digest authentication on the INVITE |
| Digest credentials | The account's VoIP Login and VoIP Password (Edit Account → Add SIP Digest Auth) |
| Extra source IPs | Edit Account → (Additional: n) → Add New IP |
| Dialled number | Tech prefix + E.164 without + or 00, for example 80039#447700900123 |
| Routing group | Group prefix starting with 444, for example 444101#447700900123 |
| Call rate | 10 calls per second per authorised IP |
| Codecs on SIP routes | G.729, G.723 and G.711, as listed in the Purchased Route Confirmation email |
Every platform guide below does the same three things:
- Define a peer, trunk or gateway pointing at
sip01.telecomsxchange.com:5060over UDP, with registration off. - Authenticate it: either send from the IP address authorised on the account, or add the VoIP login and password so the platform answers TelecomsXchange's 401 or 407 challenge.
- Rewrite the number: strip
+or00and put the tech prefix in front, so+44 7700 900123leaves as80039#447700900123.
Tip
Use one tech prefix per destination or route. If you buy several routes to the same country, a routing group lets your platform dial one 444…# prefix while TelecomsXchange picks the route.
Some SIP stacks, and CPaaS platforms such as Twilio and Telnyx, write # as %23 in the Request-URI. TelecomsXchange accepts both: it decodes %23 back to # before matching the tech prefix. If calls fail with 404, check the dialled number in the CDR first.
When TelecomsXchange challenges an INVITE for digest authentication, the realm is its platform IP, 174.136.44.213. It answers SIP OPTIONS only within a call, so leave keepalive pings (qualify) off for the trunk.
How do I receive calls from TelecomsXchange?#
TelecomsXchange delivers calls on the routes you sell, and on phone numbers you bought, from its platform IP.
| Setting | Value |
|---|---|
| Source IP | 174.136.44.213 |
| Transport | SIP over UDP |
| Source ports | 5060 to 5072 |
| Codecs | Passed through, no transcoding |
| Number format | E.164 international without +, for example 14165551234 |
| Destination | The host and port on your route (IP Address:Port), or the SIP endpoint on your number |
- Allow
174.136.44.213on your firewall for SIP (UDP, source ports 5060 to 5072) and for RTP to your media ports. - Add
174.136.44.213to your platform's ACL, trusted peer list or IP authentication rule, and send those calls into a dedicated inbound context or route. - Route the E.164 number. If your route has an Outgoing Prefix or strips digits (Edit Route → Outgoing Number Translation Parameters), match the number in that format instead.
Warning
Accept calls only from 174.136.44.213 on the context that receives TelecomsXchange traffic. An open inbound context on a public IP gets scanned and abused within hours.
How do I receive calls on numbers I bought?#
TelecomsXchange delivers each voice number to the SIP Endpoint (Primary) set on the number, in the form sip:user@host:port, for example sip:14165551234@203.0.113.10:5060, with an optional SIP Endpoint (Secondary) for failover. Change it in the number's DID Settings or in bulk by CSV. Whitelist 174.136.44.213 as above.
If you sell numbers, point them at sip01.telecomsxchange.com, UDP port 5060, and add your sending IP or VoIP login on the Incoming DIDs Authentication page. See Sell virtual numbers.
Which guide do I follow?#
| Platform | Guide | Send to TCXC | Receive from TCXC |
|---|---|---|---|
| Asterisk (PJSIP) | Connect Asterisk | Yes | Yes |
| FreeSWITCH | Connect FreeSWITCH | Yes | Yes |
| Kamailio | Connect Kamailio | Yes | Yes |
| OpenSIPS | Connect OpenSIPS | Yes | Yes |
| Sippy Softswitch | Connect Sippy | Yes | Yes |
| PortaSwitch | Connect PortaSwitch | Yes | Yes |
| Twilio | Connect Twilio | <Dial><Sip> with digest |
Phone numbers you bought |
| Telnyx | Connect Telnyx | TeXML or Call Control with digest | Phone numbers you bought |
| 3CX | Connect 3CX | Yes, IP authentication | Yes, and numbers you bought |
| FreePBX | Connect FreePBX | Yes | Yes |
| Yeastar P-Series | Connect Yeastar | Yes, IP authentication | Yes |
| Grandstream UCM | Connect Grandstream UCM | Yes, IP authentication | Yes |
| Cisco CUBE and CUCM | Connect Cisco CUBE | Yes | Yes |
| VICIdial | Connect VICIdial | Yes | Inbound numbers |
| Genesys Cloud | Connect Genesys Cloud | Yes, BYOC Cloud trunk | Inbound numbers |
| Amazon Connect | Connect Amazon Connect | Through your own SIP server | Through your own SIP server |
| Microsoft Teams | Connect Microsoft Teams | Through a certified SBC | Through a certified SBC |
| Zoom Phone | Connect Zoom Phone | Through a certified SBC | Through a certified SBC |
| VOS3000 | Connect VOS3000 | Yes | Yes |
| VoipSwitch | Connect VoipSwitch | Yes | Yes |
| MOR (Kolmisoft) | Connect MOR | Yes | Yes |
| AudioCodes Mediant SBC | Connect AudioCodes | Yes | Yes |
| Ribbon SBC | Connect Ribbon | Yes | Yes |
How do I test it?#
- Place a call to a number covered by the route, dialled as
TECH_PREFIX+ E.164. - Watch it in Live Calls while it rings.
- When it ends, find it in Logs → CDR (Voice) with its SIP code and cost. See call records and live calls.
- On your side, capture the SIP exchange with
sngreportcpdump -n -i any udp port 5060to see the exact INVITE and reply.
What if something goes wrong?#
| You see | What to do |
|---|---|
| 401 or 407 that never completes | The platform isn't answering the digest challenge. Check the VoIP login and password on Edit Account, and that the trunk has outbound authentication enabled. |
| 403 Forbidden | The source IP isn't authorised and no valid digest credentials were sent. Check your public (NAT) IP against the account's IP Address and Additional Allowed IPs, and that the account isn't blocked or out of balance. |
| 404 Not Found | Missing or wrong tech prefix, or the number isn't E.164. Copy the prefix again from My Interconnections and strip + or 00. |
| 408 Request Timeout, or no reply | Outbound UDP 5060 to sip01 is blocked, or DNS doesn't resolve on your platform. Test with sngrep and check your firewall. |
| 503 Service Unavailable | The route can't carry the call: blocked by you or the seller, out of capacity, or the seller's side failed. Check the route status in My Interconnections and try another route or a routing group. |
| One-way or no audio | Your platform advertises a private IP in SDP, or the firewall blocks RTP. Set your platform's external media address to its public IP and open your RTP port range on the firewall. |
| No inbound calls | 174.136.44.213 isn't allowed on the firewall or in the platform's ACL, or the route's IP Address:Port or the number's SIP endpoint is wrong. |
| Calls arrive but fail to set up | The caller's codec isn't enabled on your platform; TelecomsXchange doesn't transcode. |
Common questions
Can my platform register to TelecomsXchange?
No. TelecomsXchange is peering only and does not accept SIP REGISTER. Your account is recognised by the source IP of each INVITE, or by digest authentication on the INVITE with the account's VoIP login and password.
Which host and port do I send calls to?
sip01.telecomsxchange.com, port 5060, SIP over UDP.
What do I dial?
The purchased route's tech prefix, then the number in E.164 without + or 00, for example 80039#447700900123. A routing group's prefix starts with 444, for example 444101#447700900123.
Which IP do I whitelist for calls from TelecomsXchange?
174.136.44.213, SIP over UDP from source ports 5060 to 5072. It sends calls on the routes you sell and on phone numbers you bought.
Does TelecomsXchange transcode?
No. Codecs are passed through unchanged, so the two ends must share a codec.
- Can my platform register to TelecomsXchange?
- No. TelecomsXchange is peering only and does not accept SIP REGISTER. Your account is recognised by the source IP of each INVITE, or by digest authentication on the INVITE with the account's VoIP login and password.
- Which host and port do I send calls to?
- sip01.telecomsxchange.com, port 5060, SIP over UDP.
- What do I dial?
- The purchased route's tech prefix, then the number in E.164 without + or 00, for example 80039#447700900123. A routing group's prefix starts with 444, for example 444101#447700900123.
- Which IP do I whitelist for calls from TelecomsXchange?
- 174.136.44.213, SIP over UDP from source ports 5060 to 5072. It sends calls on the routes you sell and on phone numbers you bought.
- Does TelecomsXchange transcode?
- No. Codecs are passed through unchanged, so the two ends must share a codec.
Finished this guide?
This guide matches platform build 83b4ebc9. Markdown version.