SIP trunk for Asterisk and FreePBX

Yes. didlogic SIP trunks work with Asterisk on both chan_sip and chan_pjsip modules. The setup guide on docs covers digest authentication, IP authentication, and FreePBX, Elastix, and AsteriskNOW distributions. A test call typically completes within 10 minutes of finishing setup.

Both. The didlogic docs include sip.conf examples for chan_sip and pjsip.conf examples for chan_pjsip. Asterisk 22 removed chan_sip from the official source tree, so all new deployments on Asterisk 22 must use chan_pjsip. Existing deployments on Asterisk 20 LTS or earlier can keep chan_sip running through that LTS lifecycle.

Yes. FreePBX, Elastix and AsteriskNOW all run on Asterisk underneath, so any didlogic trunk that works with Asterisk works with these distributions. Trunk creation is done from the distribution’s admin UI, using either PEER details for chan_sip or pjsip settings for chan_pjsip.

FreePBX is a web administration layer that runs on top of Asterisk. If you want a GUI for managing trunks, extensions, IVR and routing, use FreePBX. If you are comfortable editing sip.conf or pjsip.conf directly, plain Asterisk works without FreePBX. didlogic SIP trunks integrate the same way either route.

Both. Digest authentication (username and password registration) works out of the box from any didlogic SIP account. For IP authentication, email support with your Asterisk server’s static IP and we will enable it on your account. IP authentication is the standard choice for high-volume deployments where registration overhead is not desirable.

Use the regional gateway closest to your Asterisk server. Examples: sip.nl.didlogic.net for Western Europe, sip.uk.didlogic.net for the UK, sip.lax.didlogic.net for the US West Coast, sip.nyc.didlogic.net for the US East Coast, sip.sg.didlogic.net for APAC. The full list of 12 gateways is available in the portal and in the setup docs.

No. You pay per DID and per minute. There is no per-channel concurrency charge on production accounts, which matters when an Asterisk dialer or contact center spikes traffic.

Not in any production sense. Free Asterisk SIP trunks exist for testing, but they are rate-limited, shared, and not suitable for real traffic. didlogic offers free credits on signup so you can place real test calls on a real trunk before paying for minutes, which is closer to what most “free trunk” searches are actually looking for.

A 603 on outbound is almost always missing or expired SIP registration. Run asterisk -x "sip show registry" (chan_sip) or asterisk -x "pjsip show registrations" (chan_pjsip). The hostname should show as Registered. If not, your credentials, port, or hostname need a closer look. The docs page has the full troubleshooting checklist.

One-way audio is almost always RTP traffic being blocked at the firewall. Open UDP ports in your RTP range (10000-20000 is the didlogic default in rtp.conf) on your edge firewall, and set nat=force_rport,comedia in sip.conf if your Asterisk box is behind NAT.

E.164 format only. Dial 442035198131 for the UK, not 0044 or 011 or 00. Dial 12125551212 for the US, not 2125551212. Your Asterisk dialplan should normalize whatever the user dials to E.164 before sending the call to the didlogic trunk.

Yes, on verified business accounts at Plus level and above. We require proof of address before enabling Caller ID passthrough. The default behavior on lower account tiers is that the didlogic DID acts as the outbound CLI.

Yes. didlogic supports paperless LNP in 8 countries and manual porting in most others. Porting does not require Asterisk downtime. A common pattern is keeping the existing trunk live while the port completes, then switching the Asterisk dialplan to the didlogic trunk on cutover day.

Two main differences. First, no per-channel fees, which is the line item that bloats Twilio bills on outbound Asterisk dialer deployments. Second, deeper international DID coverage (130+ countries, licensed local voice in 17) versus Twilio’s smaller direct-coverage footprint. The Asterisk configuration is similar on both sides. What changes is the per-month cost at concurrency. See our Twilio alternative for AI voice page for the breakdown.

Yes. Asterisk or FreePBX can sit as a SIP gateway between the PSTN and AI voice platforms (Vapi, Retell, LiveKit, ElevenLabs). The didlogic trunk handles the carrier side, Asterisk handles routing and protocol normalization, the AI platform handles the conversation. See the AI Voice Hub for platform-specific guides.