SIP Trunking

Preview

SIP

SIP Trunking

Enterprise voice connectivity over SIP. Connect your PBX, contact centre, or SBC to the public telephone network with carrier-grade origination, termination, and REST management APIs.

Full PSTN Access

Originate and terminate calls to any phone number worldwide through Sendexa carrier routes.

SIP + REST Control

Signal over SIP (UDP/TCP/TLS) and manage trunks, numbers, and CDR via the REST API.

Dual Authentication

SIP digest credentials plus optional IP access-control lists for zero-trust trunk security.

Carrier-Grade Quality

G.711, G.729, and Opus codecs with multi-region media anchors and sub-second setup.

Built-in Failover

Primary and secondary SIP URIs with sequential hunt for high-availability inbound routing.

Detailed CDR

Per-call detail records with duration, hangup cause, direction, codec, and billed minutes.

TLS & SRTP Ready

Encrypt signaling on port 5061 and optionally enable media encryption end-to-end.

PBX Friendly

Drop-in configs for Asterisk, FreeSWITCH, 3CX, FreePBX, and commercial SBCs.

SIP Host

sip…

sip.sendexa.co

Signaling Ports

5060/61

UDP · TCP · TLS

Codecs

G.711 · G.729 · Opus

Billing

/min

Per-minute, 6s increments

Authentication

All REST requests use HTTP Basic Auth with your dashboard Base64 token: Authorization: Basic <token>. Auth guide →

How SIP Trunking Works

Sendexa SIP Trunking bridges your private voice infrastructure to the public switched telephone network (PSTN). You keep your PBX, contact-centre platform, or softswitch; Sendexa provides the SIP edge, media path, and carrier interconnects.

Origination (Inbound)

A caller dials your Sendexa DID (local or international number). We receive the call from the PSTN and send a SIP INVITE to your configured SIP URI or registered contact.

PSTN → Sendexa → your PBX (sip:user@pbx.example.com)

Termination (Outbound)

Your PBX sends a SIP INVITE to sip.sendexa.co with an E.164 destination. After authentication we route the call across carrier partners to the handset.

Your PBX → Sendexa → PSTN (+233… / +1… / …)

SIP Connectivity

Edge Endpoints

Parameter	Value	Notes
SIP Host	`sip.sendexa.co`	Primary multi-region anycast edge
Port (UDP / TCP)	`5060`	Standard SIP signaling
Port (TLS)	`5061`	Encrypted signaling (recommended)
Transports	UDP, TCP, TLS	Prefer TLS in production
Auth	Digest + IP ACL	SIP username/password and optional allowlist
Codecs	G.711 (PCMU/PCMA), G.729, Opus	Negotiated via SDP offer/answer

REST Management API

Voice media and signaling stay on SIP. Use the REST API to provision trunks, assign numbers, and pull CDRs. Base URL: https://api.sendexa.co

POST

/v1/sip/trunks

Create trunk

Provision a SIP trunk with credentials, ACL, codecs, and capacity.

GET

/v1/sip/trunks

List trunks

List all trunks on the business account with status and limits.

POST

/v1/sip/numbers

Assign number

Attach a DID to a trunk and set the inbound SIP destination URI.

GET

/v1/sip/cdr

Query CDR

Retrieve call detail records with filters for date, direction, and status.

Quick Start — Create a Trunk

Provision via REST

Bash

curl -X POST 'https://api.sendexa.co/v1/sip/trunks' \
  -H 'Content-Type: application/json' \
  -H 'Authorization: Basic YOUR_DASHBOARD_BASE64_TOKEN' \
  -d '{
    "name": "Primary HQ Trunk",
    "auth": {
      "username": "trunk_hq_01",
      "password": "use-a-long-random-secret"
    },
    "ipAllowlist": ["203.0.113.10/32", "198.51.100.0/24"],
    "codecs": ["PCMU", "PCMA", "G729", "opus"],
    "transport": "tls",
    "maxConcurrentCalls": 50,
    "origination": {
      "primaryUri": "sip:pbx.example.com:5061;transport=tls",
      "failoverUri": "sip:backup.example.com:5061;transport=tls"
    }
  }'

Example response

JSON

{
  "success": true,
  "code": "TRUNK_CREATED",
  "message": "SIP trunk provisioned successfully.",
  "requestId": "REQ-9f2c1a80",
  "timestamp": "2026-07-12T10:15:22.441Z",
  "data": {
    "trunkId": "TRK-8c1e2b9a-4d3f-41a0-9e7c-12ab34cd56ef",
    "name": "Primary HQ Trunk",
    "status": "active",
    "sipHost": "sip.sendexa.co",
    "ports": { "udp": 5060, "tcp": 5060, "tls": 5061 },
    "auth": {
      "username": "trunk_hq_01",
      "realm": "sendexa.co"
    },
    "ipAllowlist": ["203.0.113.10/32", "198.51.100.0/24"],
    "codecs": ["PCMU", "PCMA", "G729", "opus"],
    "transport": "tls",
    "maxConcurrentCalls": 50,
    "origination": {
      "primaryUri": "sip:pbx.example.com:5061;transport=tls",
      "failoverUri": "sip:backup.example.com:5061;transport=tls"
    },
    "createdAt": "2026-07-12T10:15:22.441Z"
  }
}

SIP Authentication Model

Field	Type	Required	Description
username	string	required	SIP digest username issued when the trunk is created. Used in Authorization headers on REGISTER and INVITE.
password	string	required	SIP digest password. Store in your PBX secret store; never commit to source control.
ipAllowlist	string[]	optional	CIDR ranges permitted to send signaling. When set, unlisted source IPs are rejected even with valid credentials.
realm	string	optional	Always sendexa.co. Configure this in your PBX authentication realm if prompted.

STIR / SHAKEN

For destinations that require caller-ID attestation (notably North America), Sendexa applies STIR/SHAKEN signing on supported outbound routes when your account and CLI are eligible. Attestation level (A/B/C) depends on number ownership verification. See Origination & Termination for CLI presentation rules.

Documentation

Guide

Getting Started

Provision a trunk, create SIP credentials, allowlist your IP, and place a test call.

Open guide

Config

Trunk Configuration

Codecs, transport, auth modes, registration, concurrent call limits, and failover.

Open guide

Routing

Origination & Termination

Route DIDs into your SIP URI and dial out to the PSTN with E.164 destinations.

Open guide

API

CDR & Reporting

Query call detail records, understand billing units, and export usage for reconciliation.

Open guide

Billing Snapshot

Metric	Rule
Unit	Billed per minute of connected talk time (answered calls only)
Increment	6-second increments after the first 30 seconds (destination-dependent)
Failed / busy / no-answer	Not billed for talk time; setup attempts may appear in CDR with zero cost
Currency & rates	Account currency; rate cards available in the dashboard under SIP → Rates

Common SIP & API Errors

Code	Layer	Cause
`401 Unauthorized`	SIP	Invalid digest credentials or missing Authorization header
`403 Forbidden`	SIP	Source IP not on trunk ACL, or CLI not allowed
`404 Not Found`	SIP	Unknown DID or unroutable destination number
`486 Busy Here`	SIP	Far end busy or concurrent call limit reached on trunk
`488 Not Acceptable`	SIP	No common codec between offer and Sendexa allowlist
`INSUFFICIENT_CREDITS`	REST/SIP	Account balance too low for outbound termination
`TRUNK_NOT_FOUND`	REST	Invalid trunkId on number assignment or update

Preview status

SIP Trunking is in Preview. APIs and edge behaviour are production-capable for early access partners; rate cards, regional media POPs, and some number inventory may change before general availability. Contact your account manager for capacity and DID ordering.

Best practices

Prefer TLS 5061 for signaling and enable SRTP when available
Always configure an IP allowlist in addition to SIP credentials
Use E.164 for outbound destinations (+233…, not local formats)
Set a secondary failover URI for inbound DID routing
Store trunkId and correlate with callId / CDR for billing reconciliation
Monitor concurrent call utilisation against maxConcurrentCalls

SIP Trunking

Full PSTN Access

SIP + REST Control

Dual Authentication

Carrier-Grade Quality

Built-in Failover

Detailed CDR

TLS & SRTP Ready

PBX Friendly

Authentication

How SIP Trunking Works

SIP Connectivity

REST Management API

Quick Start — Create a Trunk

SIP Authentication Model

Recommended production posture

STIR / SHAKEN

Documentation

Billing Snapshot

Common SIP & API Errors

Preview status

Best practices