SIPStation: registered trunks, FreePBX integration and feature boundaries

Separate the account key, SIP credentials and registration state

SIPStation's FreePBX module uses an account key, while the manual documentation identifies service-specific SIP usernames and passwords that are not the store login. The troubleshooting guide states that the PBX must register for inbound calls. Preserve each secret's owner and function, the registered contact, expiry, selected transport and the exact PBX object instead of collapsing them into one credential field.

A reset takes effect immediately according to the provider and requires a new keycode on the PBX; service can fail until that update is applied. Plan credential rotation as a controlled change, protect the values from logs and screenshots, and keep a recovery path that does not expose the old or new secret.

Treat generated FreePBX configuration as versioned state

The SIPStation module can create two trunks and basic inbound and outbound routes. Sangoma also publishes a PJSIP-mode procedure. By contrast, its manual chan_sip article is explicitly for FreePBX 16 and older and points FreePBX 17 users elsewhere. Record the FreePBX, Asterisk and module versions, choose the supported workflow for that release, apply configuration and inspect the running objects.

Before enrollment, export existing trunks, routes and DID patterns. After generation, review ordering, caller identity, emergency routes, codec policy and any overlap with existing routes. Do not edit generated files as a durable fix or assume a historical port or channel-driver default applies to the loaded system.

Test gateway redundancy and PSTN failover separately

SIPStation documents primary and secondary gateway connections and normally attempts inbound delivery through the first and then the second. It separately offers global or per-DID failover to an alternate PSTN number. Those are different controls with different targets and failure boundaries.

For each approved test, record DNS resolution, registration state, attempted gateway, SIP response or timeout, selected failover number, delay, caller presentation and restoration. These mechanisms concern new call attempts; the reviewed material does not establish preservation of an active call when a gateway, PBX or access link fails.

Keep signaling, media, fax and encryption claims distinct

The configuration guide lists G.711 and G.729 and documents UDP signaling plus an RTP port range for standard FreePBX/Asterisk deployments. It warns that gateway addresses can change and that media can come from many addresses. Capture the actual offer, answer, DTMF behavior and RTP endpoints; a successful registration does not prove two-way media.

Sangoma separately documents best-effort T.38 on SIPStation trunks and recommends qualifying fax before porting. That does not make the distinct FaxStation product equivalent. The reviewed material did not establish TLS or SRTP for SIPStation, so a firewall rule or FQDN allowlist must not be presented as transport or media encryption.

Qualify capacity, E911 and SMS as independent services

The trunk FAQ separates DIDs from simultaneous call paths and describes account concurrency, but no current public call-start rate was verified. Obtain the effective concurrent-call and burst/CPS controls and test only within an authorized limit. Two provider gateway connections do not double the purchased capacity.

Emergency configuration uses a master E911 DID plus optional DID-specific addresses, and unmatched outbound caller identity can fall back to the master record under the documented policy. SMS is limited to capable US and Canada DIDs and, in the reviewed guide, FreePBX or PBXact through UCP or Zulu. Verify each address, identity and DID capability independently and follow current provider-approved testing and registration procedures.

Price the actual order and preserve acceptance evidence

The public marketing and store showed different current amounts during this review, so neither is treated as a durable quote. Price the exact channel, DID, outbound T.38, E911, messaging and international mix at checkout, including taxes, top-ups, commitments and support terms. Confirm number inventory, portability and activation for the transaction rather than extrapolating from trial coverage or rate-center counts.

Before porting, complete bounded inbound, outbound, caller-ID, DTMF, transfer, long-call, failover, emergency and applicable fax or SMS checks. Store redacted SIP and media evidence with the configuration export and provider case. No SIPStation account, trunk, PBX, number, call, media, fax, emergency call, message, rate, availability or support response was tested for this guide.

Follow a failed SIPStation inbound call through the actual delivery order

The SIPStation inbound diagnostic keeps registration or peer mode, primary and secondary gateway behavior, configured failover, DID delivery, FreePBX route matching and media as separate evidence.