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Configuring a new hosted PBX install — extensions, SIP trunk, first call

Your LYLIX PBX VPS came up with FreePBX® already installed, configured, and reachable on its public IP. This article walks the first hour after that:…

Connecting a hosted PBX to Telnyx — SIP trunk setup walkthrough

Telnyx is one of the most popular SIP-trunk carriers for FreePBX® deployments — API-first, credit-based pricing, strong US/Canada coverage. This article…

Connecting a hosted PBX to Flowroute — SIP trunk setup walkthrough

Flowroute is a long-running US-based SIP-trunking provider, known for IP-based authentication (no credentials to leak) and a relatively straightforward…

Connecting a hosted PBX to VoIP.ms — SIP trunk setup walkthrough

VoIP.ms is a popular low-cost SIP-trunk provider — pay-as-you-go or unlimited per-DID plans, lots of POP servers for geographic flexibility, and a…

FusionPBX® initial setup — domains, tenants, gateways, first dialplan

Your LYLIX PBX VPS provisioned with FusionPBX® came up with FreeSWITCH® running, FusionPBX accessible, admin credentials set, and PostgreSQL and ESL…

SIP behind NAT — why your audio is one-way and how to fix it

"I can hear them but they can't hear me" — or the reverse — is the most-asked SIP question on the internet, and the answer is almost always NAT.…

Provisioning Polycom, Yealink, and Grandstream phones from a hosted PBX

Hand-configuring a SIP phone through its web UI is fine for one phone. For five, ten, or fifty — you want auto-provisioning : the phone boots, fetches…

Inbound and outbound route patterns on a hosted PBX — common scenarios

Dial patterns are FreePBX®'s way of matching dialed numbers to trunks and matching inbound DIDs to destinations. The syntax is a little obscure if…

STIR/SHAKEN — what it is and what you need to do as a VPS PBX customer

STIR/SHAKEN is the framework US carriers use to cryptographically attest that the caller ID on an outbound call is legitimately owned by the originating…

E911 setup with Telnyx — register your physical address for 911 dispatch

In the United States, every phone number capable of making a 911 call must have a registered physical address that emergency dispatchers receive when the…

T.38 fax setup — passthrough vs reinvite, MTU pitfalls, per-carrier tips

Fax over VoIP is one of those problems the industry quietly admitted it never fully solved. T.38 makes it work most of the time. This article covers how…

Picking SIP codecs — G.711 vs G.722 vs Opus, bandwidth math, codec negotiation gotchas

"Which codec should I use?" is one of those VoIP questions where the wrong answer doesn't break the call — it just makes it sound a little worse, or costs…

Why your SIP trunk drops every 30 minutes — registration TTL, NAT pinholes, OPTIONS keepalives

You've got a SIP trunk that comes up, runs fine, and falls over at the same interval — every 30 minutes, every hour, every 15 minutes. It re-registers and…

Polycom autoprovisioning end-to-end — bootstrap.cfg, MAC-specific files, TLS provisioning

Polycom phones (VVX series and the newer Poly Edge) can pull their full configuration from a provisioning server on boot — no per-phone manual entry. This…

Yealink autoprovisioning — DHCP option 66, AutoP URL, encrypted configs

Yealink's "AutoP" provisioning is similar in shape to Polycom's but with a couple of important differences: the config file format, the encryption…

Vendor desk phones with EPM — when to use the hosted PBX Endpoint Manager (and when not to)

The FreePBX® Endpoint Manager (EPM) is a commercial module that handles phone provisioning from inside the PBX GUI. It's the intended workflow for…

A2P 10DLC for SMS — brand/campaign registration and why your texts vanish without it

If you're using a US 10-digit phone number to send SMS from your PBX or application, the carriers now require you to register your brand and campaign before…

WebRTC softphones with a hosted PBX — sipml5/JsSIP, WSS transport, common breakages

WebRTC lets a browser act as a SIP phone — no client install, no extension provisioning, just open a URL. FreePBX® supports this through Asterisk®'s…

Bare PBX without a management GUI — minimal pjsip.conf for a single SIP trunk

FreePBX® is excellent for managing complex deployments, but if you just need Asterisk® to bridge one carrier to a handful of extensions — or if you…

Hosted PBX backup and restore — what the built-in module covers, what it doesn't, what you should add

The FreePBX® Backup & Restore module is the in-PBX backup tool — it captures the FreePBX configuration cleanly and can restore to another FreePBX…

Call queues that actually work — priority routing, agent skills, fallback destinations

Most FreePBX® call queues start as "add agents, set ring strategy, done" and become an unmaintainable knot of overflow destinations, weird priorities, and…

Voicemail-to-email when it never arrives — DKIM/SPF, msmtp, sendmail tracing

Voicemail-to-email is one of those features that "just works" until your VPS IP changes, or your mail relay swaps, or a major email provider tightens its…

DTMF that doesn't register — RFC2833 vs SIP INFO vs inband, and when each breaks

DTMF (the keypad tones — touch-tones) seems trivial until you hit a voice menu that ignores your key presses, an IVR that half-works, or a recording…

Migrating from an on-prem PBX (Elastix, legacy distros, hardware appliances) to a LYLIX VPS PBX

Moving a PBX from a physical box in a closet to a hosted VPS is one of the most common LYLIX onboardings. The good news: with FreePBX® or FusionPBX® on…

Hardening a public-facing PBX — fail2ban, allowlists, pjsip security, port 5060 reality

A PBX with port 5060 on the public internet receives SIP brute-force traffic from the moment it boots. The attackers are looking for weak extension…

One-way audio and choppy calls — a structured diagnosis walkthrough

"They can hear me but I can't hear them" (or the reverse), or audio that's there but unusable — choppy, robotic, breaks up — is the second-most-asked…

Hosted PBX upgrade failed — recovery, rollback, and how to avoid it next time

FreePBX® distro upgrades have moving parts: framework, modules, underlying OS packages, PHP, MariaDB, Asterisk® itself. When one fails, the symptoms…

"FATAL ERROR: DB connect failed" on a hosted PBX — diagnosing MariaDB problems

The PBX boots, Asterisk® runs, but the GUI throws FATAL ERROR: DB Connection Failed or MySQL connect ERROR , or the SQL backend won't load.…

Choosing an open-source PBX platform — feature comparison

LYLIX offers PBX VPS images built on three platforms — FreePBX®, FusionPBX®, and bare Asterisk®. The right choice depends on what you're…

PBX CLI essentials — the commands you'll actually use

The Asterisk® CLI ( asterisk -rvvv ) is where you actually see what's happening when a call doesn't behave the way the GUI suggests it should.…

FreeSWITCH vs the leading open-source PBX engine — when each fits

The two open-source telephony engines that matter are Asterisk® (1999-) and FreeSWITCH® (2006-). Both can do almost anything; their architectures…

PBX under attack — incident-response runbook

A compromised PBX is the worst infrastructure incident a small business can have. International toll fraud runs up bills in the thousands of dollars per…

Call recording compliance — one-party vs two-party consent by US state

If your PBX records calls, you're in legal territory the moment those calls cross a state line. The US is split between "one-party consent" (only one party…

Conference bridge setup — multi-party PBX meetings

Conference bridges turn a PBX into a meeting platform — dial-in numbers, PIN-protected rooms, moderator controls, recording. Both FreePBX®…

Queue callback — letting callers hang up and get called back

Queue callback (sometimes called "virtual hold") replaces the "all our agents are busy, your call is important" hold experience with "press 1 to keep your…

Ring group strategies — sequential, simultaneous, hybrid, and the patterns that actually scale

A ring group is a list of extensions that ring together when one of several inbound destinations is hit. FreePBX® and FusionPBX® both expose ring…

Migrating from chan_sip to PJSIP — what changes, what breaks, what improves

Asterisk® ships two SIP channel drivers: chan_sip (legacy, deprecated, removed in Asterisk 21+) and chan_pjsip (modern, the default since Asterisk…

SIP trunk providers compared — Telnyx, Flowroute, VoIP.ms, Bandwidth, Twilio

Your VPS PBX needs a SIP trunk to make and receive calls on the public phone network. LYLIX doesn't bundle one — bring your own. Five providers come up…

Asterisk CLI essentials — the commands you'll actually use

The Asterisk® CLI ( asterisk -rvvv ) is where you actually see what's happening when a call doesn't behave the way the GUI suggests it should.…

FreeSWITCH vs Asterisk — when each engine fits

The two open-source telephony engines that matter are Asterisk® (1999-) and FreeSWITCH® (2006-). Both can do almost anything; their architectures…

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