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PortalPBX

FreePBX 15+ (PJSIP trunk)

Configure the SIP Trunk on FreePBX

Adding the PortalPBX SIP trunk in FreePBX with chan_pjsip, plus inbound and outbound routes.

Illustration: FreePBX with PortalPBX
Illustrative image.

Before you start

  • FreePBX 15 or later and an admin user.
  • Trunk details provided by PortalPBX.
  • NAT reviewed in Settings → Asterisk SIP Settings.

Your trunk details

PortalPBX sends them when the trunk is activated. Replace each placeholder with your real value.

SIP server (host or domain)
<servidor-sip>
Port and transport
<puerto> · UDP / TCP / TLS
SIP username
<usuario>
SIP password
<contraseña>
Assigned DID number(s)
<DID>
Authentication mode
Registro / IP
Enabled codecs
G.711 (alaw/ulaw) · G.729*

* G.729 only if enabled on your trunk.

Step by step

  1. 1

    Create a PJSIP trunk

    Connectivity → Trunks → Add Trunk → Add SIP (chan_pjsip) Trunk. In General: name portalpbx and Outbound CallerID with your DID.

  2. 2

    pjsip Settings → General

    Username <usuario>, Secret <contraseña>, Authentication: Outbound, Registration: Send (or None for IP auth), SIP Server <servidor-sip>, SIP Server Port <puerto>, Context from-pstn.

  3. 3

    pjsip Settings → Advanced

    From Domain <servidor-sip>, From User <usuario>, Qualify Frequency 60, Contact User <usuario>. In Codecs put alaw and ulaw first.

  4. 4

    Routes

    Connectivity → Inbound Routes: DID Number <DID> → destination. Outbound Routes: dial patterns and Trunk Sequence = portalpbx.

  5. 5

    Apply and verify

    Click Apply Config and check Reports → Asterisk Info → Registries, or via console:

    asterisk -rx "pjsip show registrations"

Recommendations

  • In Asterisk SIP Settings use “Detect Network Settings” to load the external IP and local networks.
  • Prefer G.711 alaw/ulaw; use G.729 only if it is enabled on your trunk and supported by your equipment.
  • Disable SIP ALG on your router or firewall: it is the most common cause of one-way audio and dropped registrations.
  • If your PBX is behind NAT, set the public IP and local network, and open the RTP range your system uses.
  • Restrict SIP access with a firewall and use long extension passwords to prevent toll fraud.
  • Dial numbers in the format stated in your activation sheet (national or international) and check the caller ID.

Common errors

Registration Rejected

Check username, password and that Registration is “Send”.

Llamadas entrantes no llegan

The inbound route DID does not match the received format; test with “Any DID” to diagnose.

488 / sin audio

Codecs or NAT: check codec order and NAT settings.

Need a hand?

We configure it with you, remotely

If the manual setup does not work, contact support or open a ticket. We review registration, routes, codecs and NAT with you through a secure remote session.

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