SIP Trunking

Preview

SIP

Origination & Termination

How calls enter and leave the Sendexa SIP edge: map DIDs to SIP URIs for inbound service, dial out to the PSTN with authenticated trunks, configure failover, and understand CLI presentation rules.

Origination

PSTN callers dial your DID; Sendexa INVITEs your PBX or SBC at a SIP URI.

Termination

Your PBX INVITEs sip.sendexa.co with an E.164 destination; we complete to the PSTN.

Failover hunt

Primary and secondary destinations with timeouts for resilient inbound service.

CLI & attestation

Present verified caller IDs; STIR/SHAKEN on supported international routes.

Inbound

DID→SIP

Origination

Outbound

SIP→PSTN

Termination

Host

sip…

sip.sendexa.co

Numbers API

POST

/v1/sip/numbers

Authentication

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

Call directions

Origination (inbound)

A public telephone number (DID) hosted on Sendexa rings. We receive the PSTN call and create a SIP dialog toward your infrastructure.

Caller (PSTN)
→ carrier interconnect
→ Sendexa edge
→ INVITE sip:user@your-pbx:5061
→ your IVR / queue / agent

Direction in CDR:

inbound

Termination (outbound)

An agent or application on your PBX places a call. Your system sends an authenticated SIP INVITE to Sendexa with a routable destination.

Agent / app
→ your PBX / SBC
→ INVITE sip:+233…@sip.sendexa.co
→ Sendexa + carriers
→ called party handset

Direction in CDR:

outbound

Origination — DID to SIP URI

Each DID is assigned to a trunk and given one or two destination URIs. When the number is called, Sendexa builds an INVITE with the DID as the Request-URI user (or To header, depending on profile) so your dialplan can match the called number.

POST

/v1/sip/numbers

Field	Type	Required	Description
number	string	required	E.164 DID on your account, e.g. +233302123456.
trunkId	string	required	Trunk that owns billing context and optional default destinations.
destination.primaryUri	string	required	Full SIP URI. Examples: sip:233302123456@pbx.example.com:5061;transport=tls or sip:ivr@10.0.0.5:5060
destination.failoverUri	string	optional	Tried when primary is unreachable or times out.
destination.timeoutSeconds	integer	optional	Primary hunt timeout before failover (default 20, max 60).
destination.headers	object	optional	Optional custom SIP headers added to the outbound INVITE toward your PBX (e.g. X-Sendexa-DID).

Bash

curl -X POST 'https://api.sendexa.co/v1/sip/numbers' \
  -H 'Content-Type: application/json' \
  -H 'Authorization: Basic YOUR_DASHBOARD_BASE64_TOKEN' \
  -d '{
    "number": "+233302123456",
    "trunkId": "TRK-8c1e2b9a-4d3f-41a0-9e7c-12ab34cd56ef",
    "destination": {
      "primaryUri": "sip:233302123456@pbx.example.com:5061;transport=tls",
      "failoverUri": "sip:233302123456@backup.example.com:5061;transport=tls",
      "timeoutSeconds": 20,
      "headers": {
        "X-Sendexa-Trunk": "TRK-8c1e2b9a-4d3f-41a0-9e7c-12ab34cd56ef"
      }
    }
  }'

What your PBX receives

Typical INVITE shape (illustrative):

Bash

INVITE sip:233302123456@pbx.example.com:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS sip.sendexa.co:5061;branch=z9hG4bK...
From: <sip:+233555539152@sip.sendexa.co>;tag=...
To: <sip:+233302123456@sip.sendexa.co>
Contact: <sip:edge@sip.sendexa.co:5061;transport=tls>
Call-ID: 5f2a...@sip.sendexa.co
CSeq: 100 INVITE
Max-Forwards: 70
P-Asserted-Identity: <sip:+233555539152@sip.sendexa.co>
Content-Type: application/sdp
Content-Length: ...

v=0
o=- 0 0 IN IP4 ...
s=Sendexa SIP
c=IN IP4 ...
t=0 0
m=audio ... RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

From / PAI — original caller ID when available
To / Request-URI user — your DID (match in dialplan)
SDP — media toward Sendexa relay or direct as configured

Inbound failover behaviour

Primary response	Next step	CDR note
100/180/183 then 200	Media established on primary	destinationUri = primary
Connection refused / TLS fail	Immediate failover	failoverUsed = true
No response until timeout	Failover after timeoutSeconds	failoverUsed = true
480 / 503 / 500	Failover	hangupCause may reflect final leg
486 Busy Here	No failover (subscriber busy)	status = busy
404 / 484	No failover (routing error on PBX)	status = failed
Both fail	Caller cleared with network error	status = failed

Termination — SIP to PSTN

Outbound calls are pure SIP toward sip.sendexa.co. There is no separate REST “create call” for trunking (use the Voice API for programmable outbound). Your PBX is the call controller.

Dial string rules

Format	Example	Accepted?
E.164 with +	`+233555539152`	Yes — preferred
E.164 digits only	`233555539152`	Yes
Local leading 0	`0555539152`	Maybe — if account locale set; prefer E.164
With trunk prefix	`9+233555539152`	Strip prefix on PBX before send
SIP URI user	`sip:+233555539152@sip.sendexa.co`	Yes

Bash

# Request-URI examples your PBX may send
INVITE sip:+233555539152@sip.sendexa.co:5061;transport=tls SIP/2.0
INVITE sip:233555539152@sip.sendexa.co SIP/2.0

# Authorization: Digest username="cc_trunk_prod", realm="sendexa.co", ...
# From / PAI should present a DID you are allowed to use as CLI

CLI (caller ID) presentation

Set outbound caller ID to a Sendexa DID assigned to your account (or an explicitly authorised CLI).
Prefer P-Asserted-Identity and a matching From user part in E.164.
Unverified or spoofed CLIs may be rewritten, rejected with 403, or sent with reduced attestation.
Privacy: Privacy: id requests restricted presentation where the destination network supports it.

STIR / SHAKEN

Attestation on supported routes

For destinations that participate in STIR/SHAKEN (notably the United States and other adopting markets), Sendexa can sign outbound Identity headers when:

Your account is enabled for signed termination on that route
The presented CLI is a number you fully control (full attestation A)
Or partial knowledge yields attestation B / gateway C as appropriate

Ghana domestic and many regional African routes do not require STIR/SHAKEN today; normal CLI validation still applies. Attestation level, when present, is recorded on the CDR (attestation field).

Dialplan patterns

ini

[outbound-sendexa]
; 00 + country... international
exten => _00X.,1,Set(CALLERID(num)=+233302123456)
 same => n,Set(CALLERID(name)=My Company)
 same => n,Dial(PJSIP/+${EXTEN:2}@sendexa-endpoint,60,tT)
 same => n,Hangup()

; Local Ghana mobile 0XXXXXXXXX → +233
exten => _0XXXXXXXXX,1,Set(CALLERID(num)=+233302123456)
 same => n,Dial(PJSIP/+233${EXTEN:1}@sendexa-endpoint,60)
 same => n,Hangup()

; Already E.164
exten => _+X.,1,Set(CALLERID(num)=+233302123456)
 same => n,Dial(PJSIP/${EXTEN}@sendexa-endpoint,60)
 same => n,Hangup()

Number lifecycle

Inventory

Numbers purchased or allocated via dashboard / account team appear as available DIDs.

Assign

POST /v1/sip/numbers binds a DID to a trunk and destination URIs for origination.

Reassign / release

Update destination URIs, move trunkId, or release the number back to inventory when unused.

Billing & direction

Outbound termination is billed per connected minute using your outbound rate card. Inbound origination may include per-minute or per-number monthly fees depending on the DID product. Failed, busy, and unanswered calls typically have zero talk-time charge but still appear in CDR for analytics. See CDR & Reporting.

CDR & Reporting

Fields, filters, billing units, and export examples.

Trunk Configuration

Codecs, auth, capacity, and PBX config reference.

Best practices

Always normalise to E.164 before leaving your SBC
Set both primary and failover URIs for every production DID
Use OPTIONS monitoring from your SBC toward Sendexa and vice versa
Keep CLI inventory documented — only present numbers you own
Log Call-ID on your side and match to Sendexa callId / sipCallId in CDR