Professional VoIP • Self-Hosted Infrastructure

Enterprise-Grade SIP Communications for Your Desktop

Reliable, low-latency voice calling powered by Asterisk and custom SIP trunking. Requires 3CX or Zoiper desktop client for optimal stability.

Server voip.asherflora.com
Protocol SIP / TLS
Encryption TLS

Built for Reliability

Our self-hosted Asterisk deployment delivers enterprise-grade voice communications with native desktop application stability.

Self-Hosted Security

Full control over your communication infrastructure. No third-party data collection or cloud dependencies.

HD Voice Quality

Crystal-clear audio with G.722 wideband codec support and optimized codec selection for each call.

Desktop Stability

Native 3CX and Zoiper clients provide rock-solid reliability, native OS integration, and offline capabilities.

Secure SIP Trunking

Custom SIP trunk configuration with TLS encryption and secure authentication for enterprise-grade protection.

Always-On Uptime

Self-hosted on reliable infrastructure with proactive monitoring ensures your communications never go down.

Multi-Extension Support

Scalable extension-based routing with call queues, IVR, and concurrent call handling.

Best Experience with Native Clients

Our SIP service requires a desktop softphone for the most stable, feature-rich experience. Get started with one of these trusted applications.

Recommended

3CX Desktop App

The industry-standard softphone with unmatched stability, conferencing features, and seamless integration.

  • Crystal-clear HD audio
  • Conference calling
  • Call recording
  • Chat & presence
Download 3CX

Available for Windows, macOS, and Linux

Zoiper Desktop

Lightweight, cross-platform softphone with excellent codec support and low resource usage.

  • Low bandwidth usage
  • Multi-account support
  • Video calling ready
  • Free & paid tiers
Download Zoiper

Available for Windows, macOS, Linux, iOS, Android

Quick Setup Guide

Get connected in minutes with these simple configuration steps for your preferred desktop client.

1

Download & Install 3CX

Get the 3CX Desktop App from the official website. Run the installer and launch the application.

2

Enter Account Details

Choose "I have an account" and enter your extension credentials provided by your administrator.

3

Configure SIP Server

Set the SIP server to voip.asherflora.com on port 5060 using TLS transport.

4

Register & Connect

Enter your extension number and password. Click "Connect" to register with the PBX.

1

Download & Install Zoiper

Download Zoiper Desktop from zoiper.com. Select the free or paid version based on your needs.

2

Create New SIP Account

Open Zoiper, click the "+" button, and select "Create new SIP account" to start the configuration wizard.

3

Enter Extension Details

Enter your extension number as the username, your password, and set the domain to voip.asherflora.com.

4

Configure Transport & Register

Select "TLS" as the transport protocol with port 5060. Zoiper will automatically detect settings and register your account.

Connection Parameters

SIP Domain / Server voip.asherflora.com
Transport TLS
Port 5060

Demo Accounts

Extension 1
Username 1001
Password Password1001
Extension 2
Username 1002
Password Password1002

Frequently Asked Questions

Why can't I use the web browser?

Browser-based WebRTC SIP softphones face NAT traversal challenges and inconsistent audio handling across browsers. Desktop applications like 3CX and Zoiper provide native OS integration, better codec optimization, and superior stability for mission-critical communications.

Is my communication secure?

Yes. All SIP traffic uses WSS (WebSocket Secure) with TLS encryption. The server is self-hosted with no third-party cloud dependencies. Your calls stay private.

Which is better: 3CX or Zoiper?

3CX offers more advanced features if you need conferencing, call queues, and integrated chat. Zoiper is lighter, works on more platforms, and is ideal for simpler SIP-to-SIP calling needs.

Where do I get my credentials?

Contact your system administrator for your extension number and password. These are configured on the Asterisk PBX and cannot be self-provisioned.

Can I call regular phone numbers with this service?

Currently, this system supports direct SIP-to-SIP calling between registered extensions. To place outbound calls to regular PSTN numbers (mobile or landline), you would need to integrate a SIP trunk provider such as Twilio, which operates on a pay-as-you-go basis. This integration requires additional account setup and is not included in the default configuration.

Can I receive calls from regular phone numbers?

No, not with the current setup. This service handles only SIP-to-SIP internal calling between registered extensions. Receiving inbound calls from external PSTN numbers (mobile or landline) requires a SIP trunk or DID provider such as Twilio, which involves paid provisioning and additional Asterisk configuration.