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Setting up a FreePBX SIP trunk: connecting a SMS Conecta number over PJSIP

Updated 2026-09-30 · By the SMS Conecta team (SMSConecta Inc.)

Short answer: In FreePBX, head to Connectivity > Trunks, click Add Trunk and pick a SIP (chan_pjsip) trunk. Set your SMS Conecta number as the outbound caller ID, then within the PJSIP settings enter your SIP username and secret, sip.smsconecta.com as the server and 5060 as the port, with registration switched on. Wrap up with an inbound route and an outbound route.

FreePBX builds its underlying Asterisk configuration through web forms, so pjsip.conf never needs hand editing. The trunk itself takes only a few minutes; it's the routes where most of the thinking happens.

What you need first

  • A SMS Conecta number with its SIP username and password, found on that number's dashboard page.
  • FreePBX running the chan_pjsip driver (the default across current releases).
  • Admin access to the FreePBX web interface, and, when the PBX sits behind NAT, the ability to forward RTP ports.

Connection settings

SettingValue
SIP server / registrar / domainsip.smsconecta.com
Transport and portUDP or TCP 5060; TLS 5061
Username<SIP_USERNAME> (from the number's page in the dashboard)
Password<SIP_PASSWORD> (from the same page)
AuthenticationDigest (username + password), registration based
CodecsG.711 µ-law (PCMU) and G.711 A-law (PCMA)
DTMFRFC 2833 (RTP events) or SIP INFO
Inbound DID<YOUR_NUMBER> in E.164 format

Step-by-step: creating the PJSIP trunk

Labels follow Sangoma's Trunks Module user guide as of September 2026; tab names may shift slightly between FreePBX versions.

  1. Log in to FreePBX and go to Connectivity > Trunks.
  2. Click Add Trunk, then choose Add SIP (chan_pjsip) Trunk.
  3. On the General tab, name the trunk something like smsconecta and set Outbound CallerID to <YOUR_NUMBER>. Maximum Channels can stay empty, or set whatever limit suits you.
  4. Open pjsip Settings > General. Set Username to <SIP_USERNAME>, Secret to <SIP_PASSWORD>, Authentication to Outbound, Registration to Send, SIP Server to sip.smsconecta.com and SIP Server Port to 5060.
  5. Pick the transport matching your setup (UDP is the usual pick; use a TLS transport on port 5061 if one has been configured).
  6. Open pjsip Settings > Codecs and turn on only ulaw and alaw.
  7. Submit, then click Apply Config. Check Reports > Asterisk Info (or run pjsip show registrations from the Asterisk CLI) to confirm the trunk shows Registered.

Inbound route for your DID

Go to Connectivity > Inbound Routes and add a new route. Put your SMS Conecta number into the DID Number field, then pick a destination such as an extension, ring group, queue or IVR.

Should calls fall through to the catch-all route, watch the Asterisk CLI while a call comes in. The DID shown there is exactly the string your route must match; some setups show it with a leading plus, others without.

Outbound route and caller ID

Under Connectivity > Outbound Routes, create a route, add the smsconecta trunk into its Trunk Sequence, and write dial patterns converting what users dial into full international numbers.

Set the caller ID, on the route or the trunk, to your SMS Conecta number, matching it to whichever number owns the trunk's credentials. Each number carries 1,000 outgoing minutes per month.

Testing the trunk before going live

A brief test plan catches most mistakes before real callers run into them. Run it once right after setup, then again whenever a firewall or router changes.

  1. In the Asterisk CLI (fwconsole or asterisk -rvvv), run pjsip show registrations and confirm the smsconecta trunk shows Registered.
  2. Call your SMS Conecta number from a mobile phone. The inbound route's destination should ring, with both sides hearing each other clearly.
  3. While that call runs, press a few digits in an IVR or on a test extension to confirm DTMF gets detected.
  4. Place an outbound call to another phone and verify it shows your SMS Conecta number as the caller ID.
  5. Keep a call running for at least a minute; audio dropping around the 30-second mark usually points to a NAT or firewall timeout.

Troubleshooting

  • Registration rejected with 401 or 403: re-enter username and secret straight from the dashboard, and confirm Authentication is set to Outbound rather than Inbound.
  • Registered yet inbound calls never show up: check whether the FreePBX firewall module or host firewall is blocking traffic coming from sip.smsconecta.com.
  • One-way audio: set the external address and local networks under Settings > Asterisk SIP Settings, forward your RTP range, and disable SIP ALG on the router.
  • Calls drop immediately with a 488 response: a codec mismatch. Keep ulaw and alaw enabled on the trunk.
  • Keypad tones go ignored: set DTMF to RFC 4733/2833 on the trunk, the RTP events method, switching to SIP INFO only if that proves necessary.

Frequently asked questions

chan_pjsip or chan_sip, which should FreePBX use?

Go with chan_pjsip. Current FreePBX releases are built around that driver, and this page is written with it in mind.

What belongs in the SIP Server field?

Type sip.smsconecta.com, pairing it with port 5060 for UDP/TCP or 5061 for TLS.

My trunk shows registered, so why do inbound calls land in the wrong place?

The inbound route's DID doesn't match the number format Asterisk actually receives. Copy the exact DID out of the CLI and paste it into the route.

Can several SMS Conecta numbers be added to one FreePBX?

Yes. Since each number carries its own credentials, add a trunk and an inbound route per number.

Does the trunk include voicemail?

Voicemail isn't part of the SMS Conecta service. Anything of that sort would come from FreePBX itself.

Can I port an existing number into SMS Conecta for FreePBX?

Number porting isn't offered. Instead, pick a new number from any of 181 countries.

Related pages

Get a number in these countries

Sources