Standard pjsip driver
No custom modules. Works with the FreePBX UI you already know.
Add DIDHub as a SIP trunk in FreePBX with the standard pjsip driver. Inbound routes, outbound routes, Caller ID, recording — all working on the standard FreePBX UI.
FreePBX is the most-deployed open-source PBX globally and the GUI on top of Asterisk. DIDHub plugs in as a standard SIP trunk via the chan_pjsip driver — no third-party modules needed. The same FreePBX UI you use for in-house extensions handles DIDHub DIDs identically: inbound routes map DIDs to destinations, outbound routes select trunks per dialed prefix, Caller ID is per-trunk or per-extension.
FreePBX users come to DIDHub for international expansion (most other SIP carriers are US-centric), for predictable per-DID flat pricing, and for the option to mix DIDHub with their existing carriers (FreePBX supports multiple SIP trunks per dial plan).
FreePBX with chan_pjsip is the recommended setup. Six steps from buying a DID to first call.
Create a DIDHub account, buy a DID in your chosen country. Activates instantly on most markets.
Add a new pjsip trunk. Name it didhub-{country}.
DIDHub provides the host (sip.didhub.io), port (5061 TLS), authentication user, secret, and outbound registration values.
Connectivity → Inbound Routes → Add. DID = your DIDHub number, destination = an extension, queue, or IVR.
Connectivity → Outbound Routes → Add. Match dial pattern (e.g. country prefix), trunk sequence = your DIDHub trunk.
Click Apply Config (red bar). Test inbound + outbound. Both should route via DIDHub.
The exact trunk values DIDHub provisions for your account. Pre-fill these in the FreePBX trunk-config UI:
# FreePBX pjsip trunk — General + pjsip Settings tab Trunk Name: didhub-us SIP Server: sip.didhub.io SIP Server Port: 5061 Transport: 0.0.0.0-tls (or pjsip-tls) Authentication: Outbound Username: didhub_xxxxxx Secret: (provisioned) Auth Realm: sip.didhub.io Send RPID/PAI: Send P-Asserted-Identity header Outbound Caller ID: "Your Co" <+15551234567> DTMF Mode: RFC 4733 (RFC 2833) Media Encryption: SRTP via DTLS / SDES
Use chan_pjsip (not the legacy chan_sip). chan_sip is deprecated in modern Asterisk and FreePBX 16+. The pjsip transport must be TLS-enabled in /etc/asterisk/pjsip.transports.conf for the 5061 endpoint.
No custom modules. Works with the FreePBX UI you already know.
Map DIDs to extensions/queues/IVRs; route outbound dialing through DIDHub trunks per country.
Per-trunk Outbound Caller ID; FreePBX's CID Superfecta enriches inbound CID.
FreePBX call recording works on DIDHub trunks. MP3/WAV files saved to the FreePBX server.
DIDHub DIDs work alongside other SIP trunks in the same FreePBX dial plan.
FreePBX exposes Asterisk Manager events for DIDHub trunks; AGI scripts work normally.
chan_pjsip. chan_sip is deprecated in Asterisk 18+ and removed in Asterisk 21. All modern FreePBX installs (16+) ship with pjsip enabled.
Yes. The hosted FreePBX offerings use the same pjsip driver; DIDHub plugs in identically.
Yes, with reduced features. chan_sip works against DIDHub but lacks SRTP via DTLS and some modern NAT handling. We strongly recommend upgrading to pjsip.
Either one trunk per country (cleanest) or one trunk with multiple DIDs assigned (lower trunk count). Both work; the former is easier for outbound routing rules.
Yes. SIP REFER, attended transfers, and blind transfers all work through DIDHub's managed SBC.
Asterisk · 3CX · FusionPBX · Softphones · Teams Direct Routing
Pick a DID in 80+ countries from $1.99/month. Activates instantly on most numbers.