Vanilla chan_pjsip
No custom modules, no patches. Pure standards-compliant SIP.
DIDHub for Asterisk
Add DIDHub as a SIP trunk in Asterisk via chan_pjsip — full pjsip.conf and extensions.conf examples below. Works with Asterisk 18, 20, 21 and beyond.
Why DIDHub + Asterisk
Asterisk is the open-source telephony framework underneath FreePBX, FusionPBX, Issabel, AsteriskNOW, and many commercial PBXs. DIDHub connects via the standard chan_pjsip driver — vanilla SIP, no custom modules. If you're running Asterisk directly without a GUI, this guide gives you the bare config you need.
Most direct-Asterisk users come to DIDHub for embedded telephony in software products (a SaaS that needs to make/receive calls), for AI voice agent infrastructure built on Asterisk + LLM stacks, or for in-house contact-center builds where a GUI like FreePBX would be more constraining than helpful.
Setup — 15 minutes
chan_pjsip-based config. Five steps from a fresh Asterisk install to a working DIDHub trunk.
Verify chan_pjsip is loaded
module show like pjsip — should list res_pjsip, chan_pjsip, res_pjsip_session loaded.
Edit pjsip.conf
Add transport, endpoint, aor, auth, identify, registration sections (see config below).
Edit extensions.conf
Add a context that matches inbound DID and an outbound dialplan that uses the DIDHub endpoint.
Reload pjsip + dialplan
asterisk -rx 'pjsip reload' and 'dialplan reload'. Check 'pjsip show registrations' for OK status.
Test inbound + outbound
Call into the DID; have the dialplan exec to a Playback or to a SIP/extension. Place an outbound through the DIDHub endpoint.
SIP trunk configuration
The exact trunk values DIDHub provisions for your account. Pre-fill these in the Asterisk trunk-config UI:
; pjsip.conf — DIDHub trunk endpoint [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/your-cert.pem priv_key_file=/etc/asterisk/keys/your-key.pem [didhub] type=endpoint transport=transport-tls context=from-didhub disallow=all allow=ulaw,alaw,opus outbound_auth=didhub-auth aors=didhub from_user="didhub_xxxxxx" from_domain=sip.didhub.io media_encryption=sdes rtp_symmetric=yes force_rport=yes rewrite_contact=yes direct_media=no [didhub-auth] type=auth auth_type=userpass username="didhub_xxxxxx" password="(provisioned)" [didhub] type=aor contact=sip:sip.didhub.io:5061;transport=tls [didhub-reg] type=registration transport=transport-tls outbound_auth=didhub-auth server_uri=sip:sip.didhub.io:5061 client_uri=sip:[email protected]:5061 [identify-didhub] type=identify endpoint=didhub match=sip.didhub.io ; extensions.conf [from-didhub] exten => _X.,1,NoOp(Inbound DID: ${EXTEN}) exten => _X.,n,Dial(PJSIP/100,30) exten => _X.,n,Hangup() [from-internal] exten => _NXXXXXXXXX,1,Dial(PJSIP/${EXTEN}@didhub,60) exten => _NXXXXXXXXX,n,Hangup()
The pjsip endpoint uses outbound auth tied to your DIDHub-provisioned credentials. Replace didhub_xxxxxx with the username DIDHub provisions (visible in your DIDHub dashboard under Trunks). For TLS, you'll need a cert/key pair — DIDHub accepts standard public CAs.
Capabilities verified with Asterisk
TLS + SRTP
Encrypted signaling and media via standard pjsip transport-tls + media_encryption=sdes.
Asterisk 18+, 20, 21
All currently-supported Asterisk LTS lines work.
AMI / ARI events
Asterisk Manager Interface and Asterisk REST Interface emit standard events for DIDHub trunks — wire to your monitoring.
AGI / Stasis
AGI scripts and Stasis applications work normally for advanced dialplan logic.
STIR/SHAKEN headers
DIDHub injects SHAKEN identity headers on US outbound; Asterisk's AddHeader can be used for compliance customization.
Frequently asked questions
Which Asterisk version?
Asterisk 18 LTS, 20 LTS, or 21+. chan_pjsip is the only supported driver. chan_sip is removed in 21.
Can I use chan_sip with DIDHub?
Technically yes for older Asterisk, but strongly discouraged. chan_sip lacks modern NAT handling, SRTP via DTLS, and is removed from Asterisk 21.
Do I need a public IP?
No. DIDHub handles NAT-traversal via standard SIP rport / Symmetric RTP. Asterisk behind NAT works fine — just set rtp_symmetric=yes, force_rport=yes, rewrite_contact=yes on the endpoint.
How do I handle multiple country trunks?
Multiple endpoint sections in pjsip.conf, one per country. Each has its own auth + aor + registration. The dialplan picks which endpoint to dial out via.
Does DIDHub work with PJSIP-based VoIP frameworks (FastAGI, asterisk-Java)?
Yes. DIDHub speaks standard RFC SIP; any framework that integrates with chan_pjsip works.
Related
FreePBX · FusionPBX · 3CX · Softphones
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