PBXme is a VoIP and business telephony knowledge hub. We publish practical guides, comparisons, and setup tips for cloud PBX, SIP trunks, softphones, call routing, and modern business communications.

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A cloud PBX is a phone system hosted in the cloud (instead of on-premise hardware). Calls are handled over the internet (VoIP), and can ring your mobile, desk phone, softphone, or multiple devices at once—based on routing rules you control.

Not necessarily. Many businesses use softphones (apps) on computers and smartphones. If you prefer desk phones, you’ll need SIP-compatible IP phones, or an adapter (ATA) for legacy phones.

Yes. Simultaneous call capacity depends on your provider plan (channels), your SIP trunk settings, and your internet quality. You can scale up channels to support more concurrent calls as your team grows.

Typical restrictions across VoIP providers include anti-fraud / anti-spam policies, acceptable use rules, and compliance with local telecom regulations (especially for caller ID and number ownership).

Inbound calls can ring a SIP endpoint (PBX/softphone), be forwarded to mobile/landline, or follow routing rules like IVR menus, ring groups, queues, and time-of-day routing.

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Outbound calls are placed via a SIP trunk configured on your PBX or softphone. Your provider routes the call to the PSTN, and caller ID (CLI) may be controlled by configuration and local rules.

Use full international format (E.164), typically starting with + and the country code. Example: +1 (USA), +44 (UK), +972 (Israel).

Call quality depends on latency, jitter, packet loss, codec choice, and network congestion. For best results: use a stable wired connection, QoS on the router, and a provider with reliable routing.

Use a SIP softphone or PBX with test credentials (when available), validate registration, and place test calls. Check one-way audio, DTMF, caller ID, and stability over time.

A stable internet connection, SIP-compatible devices/software, and proper network configuration (NAT, firewall, QoS). Business-grade routers and managed networks often improve reliability.

Most SIP providers support popular PBXs such as Asterisk/FreePBX, 3CX, FusionPBX, hosted PBX services, and many softphones.

Yes. Many stacks support webhooks, APIs, call detail records (CDRs), and integrations with tools like Zapier/Make. This enables automation like missed-call alerts, CRM logging, and dashboard reporting.

It depends on the number type and provider. Some numbers support inbound SMS, some support outbound, and some do not support SMS at all. Always confirm SMS capability before purchasing.

Sometimes—but it’s never guaranteed. Acceptance is controlled by the third-party platform and can change without notice.

Common reasons include platform filtering of VoIP ranges, country restrictions, carrier routing issues, or SMS not being enabled on the specific number type.

Billing models vary: monthly subscriptions per user/seat, per-number rental, per-minute usage, channel-based plans, or bundled minutes. Always review rates for inbound/outbound, forwarding, and add-ons.

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KYC (“Know Your Customer”) is an identity / business verification process required by many regulators and carriers. It helps prevent fraud, spam, and misuse of phone numbers.

Depending on country and number type, providers may ask for one or more of:

1. Personal ID (driver’s license, national ID, or passport)
2. A local address
3. Proof of address (utility bill, municipal tax bill, internet/phone bill) showing name and address
4. Company registration / incorporation documents
5. An active website and an active email address on the same domain
6. A contact phone number

Verify SIP credentials, server/host, and network settings (NAT, firewall, SIP ALG, ports). Try a different network, disable SIP ALG, and confirm your IP isn’t blocked by the provider.

One-way audio is usually NAT/firewall related. Check RTP port ranges, public IP settings, and whether your PBX is behind NAT. Use STUN/TURN where appropriate and confirm symmetric RTP settings.