PBX / VoIP
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Your LYLIX PBX VPS came up with FreePBX® already installed, configured, and reachable on its public IP. This…
Telnyx is one of the most popular SIP-trunk carriers for FreePBX® deployments — API-first, credit-based pricing,…
Flowroute is a long-running US-based SIP-trunking provider, known for IP-based authentication (no credentials to…
VoIP.ms is a popular low-cost SIP-trunk provider — pay-as-you-go or unlimited per-DID plans, lots of POP…
Your LYLIX PBX VPS provisioned with FusionPBX® came up with FreeSWITCH® running, FusionPBX accessible, admin…
"I can hear them but they can't hear me" — or the reverse — is the most-asked SIP question on the…
Hand-configuring a SIP phone through its web UI is fine for one phone. For five, ten, or fifty — you want…
Dial patterns are FreePBX®'s way of matching dialed numbers to trunks and matching inbound DIDs to destinations.…
STIR/SHAKEN is the framework US carriers use to cryptographically attest that the caller ID on an outbound call is…
In the United States, every phone number capable of making a 911 call must have a registered physical address that…
Fax over VoIP is one of those problems the industry quietly admitted it never fully solved. T.38 makes it work most…
"Which codec should I use?" is one of those VoIP questions where the wrong answer doesn't break the call — it just…
You've got a SIP trunk that comes up, runs fine, and falls over at the same interval — every 30 minutes, every…
Polycom phones (VVX series and the newer Poly Edge) can pull their full configuration from a provisioning server on…
Yealink's "AutoP" provisioning is similar in shape to Polycom's but with a couple of important differences: the…
The FreePBX® Endpoint Manager (EPM) is a commercial module that handles phone provisioning from inside the PBX GUI.…
If you're using a US 10-digit phone number to send SMS from your PBX or application, the carriers now require you to…
WebRTC lets a browser act as a SIP phone — no client install, no extension provisioning, just open a URL. FreePBX®…
FreePBX® is excellent for managing complex deployments, but if you just need Asterisk® to bridge one carrier to…
The FreePBX® Backup & Restore module is the in-PBX backup tool — it captures the FreePBX configuration cleanly…
Most FreePBX® call queues start as "add agents, set ring strategy, done" and become an unmaintainable knot of…
Voicemail-to-email is one of those features that "just works" until your VPS IP changes, or your mail relay swaps, or…
DTMF (the keypad tones — touch-tones) seems trivial until you hit a voice menu that ignores your key presses, an…
Moving a PBX from a physical box in a closet to a hosted VPS is one of the most common LYLIX onboardings. The good…
A PBX with port 5060 on the public internet receives SIP brute-force traffic from the moment it boots. The attackers…
"They can hear me but I can't hear them" (or the reverse), or audio that's there but unusable — choppy, robotic,…
FreePBX® distro upgrades have moving parts: framework, modules, underlying OS packages, PHP, MariaDB, Asterisk®…
The PBX boots, Asterisk® runs, but the GUI throws FATAL ERROR: DB Connection Failed or MySQL connect ERROR, or…
LYLIX offers PBX VPS images built on three platforms — FreePBX®, FusionPBX®, and bare Asterisk®. The…
The Asterisk® CLI (asterisk -rvvv) is where you actually see what's happening when a call doesn't behave the way…
The two open-source telephony engines that matter are Asterisk® (1999-) and FreeSWITCH® (2006-). Both can do…
A compromised PBX is the worst infrastructure incident a small business can have. International toll fraud runs up…
If your PBX records calls, you're in legal territory the moment those calls cross a state line. The US is split…
Conference bridges turn a PBX into a meeting platform — dial-in numbers, PIN-protected rooms, moderator controls,…
Queue callback (sometimes called "virtual hold") replaces the "all our agents are busy, your call is important" hold…
A ring group is a list of extensions that ring together when one of several inbound destinations is hit. FreePBX®…
Asterisk® ships two SIP channel drivers: chan_sip (legacy, deprecated, removed in Asterisk 21+) and chan_pjsip…
Your VPS PBX needs a SIP trunk to make and receive calls on the public phone network. LYLIX doesn't bundle one —…
The Asterisk® CLI (asterisk -rvvv) is where you actually see what's happening when a call doesn't behave the way…
The two open-source telephony engines that matter are Asterisk® (1999-) and FreeSWITCH® (2006-). Both can do…
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