![\"Mascote](\"\/wp-content\/uploads\/2026\/10\/freepbx-17-debian-12-capa-v1.webp\")
<\/p>\n<\/p>\n
Setting up an IP phone system on Linux does not need to start with manually editing the dialplan. The FreePBX 17<\/strong> provides a web interface to manage Asterisk: extensions, trunks, routes, queues, and auto-attendant are organized into modules. The important change in this generation is in the foundation: Debian 12, installation via script or ISO, and attention to migrating from chan_sip to PJSIP<\/strong>.<\/p>\n In this guide, we will separate the versions involved, choose the installation method, and set up a lab with two extensions. Then, we will look at what changes when migrating an existing PBX and the precautions before connecting to a carrier. The references are the official Sangoma documentation<\/a>, the installer maintained by the project, and the two videos indicated at the end.<\/p>\n Think of two layers: Asterisk<\/strong> runs the telephony; FreePBX<\/strong> provides the panel that organizes your configuration. The first processes calls and dialplan applications. The second lets you manage the PBX without writing every configuration by hand. This is the relationship that the official installer repository<\/a> describes.<\/p>\n There is also a difference between open-source software and a telephone service: installing FreePBX does not provide you with a public number or free external calls. Two local extensions can talk to each other without a carrier; to reach the public telephone network, you will need an appropriate service, such as a hired SIP trunk.<\/p>\n The ecosystem combines open components with optional commercial modules<\/a>. So, confirm the license of the desired feature before assuming that a function shown in a demo is included in the basic install.<\/p>\n These points are listed on the official FreePBX 17 page<\/a>. The final version, or GA, was made available at 2 August 2024<\/strong>. The current installer code<\/a> confirms Asterisk 22 and PHP 8.2 as defaults and rejects non-Bookworm systems.<\/p>\n In the lab, two SIP clients register their extensions on Asterisk. The administrator configures the PBX through FreePBX. The trunk with a carrier is a later step: it is not required for the first internal call<\/strong>.<\/p>\n Signaling is not audio.<\/strong> SIP negotiates the call; RTP carries the media. Thus, an extension can register and make the other one ring, but still have no audio if the network is incorrect. The diagram represents media passing through the PBX; direct media settings can change that path.<\/p>\n Going to virtualize? See also the history of Proxmox<\/a> and the history of VirtualBox<\/a>. To understand the distribution that underpins the PBX, it's worth reading the history of Ian Murdock and Debian<\/a>.<\/p>\n A official downloads page<\/a> specifies the minimum of 2 GB of RAM and 20 GB of disk<\/strong> for the Debian host or VM 12. This is not a production sizing: simultaneous calls, transcoding, and recordings change the resource requirement.<\/p>\n As a starting point for this small lab, I suggest 2 vCPUs, 4 GB of RAM and 40 GB of disk<\/strong>. It is a lab choice, not a capacity guarantee or an official requirement. Use a dedicated VM, a stable IP address or DHCP reservation, working DNS, and internet access to download the packages.<\/p>\n Leave the VM and both clients on a lab network that allows communication between them. On networks isolated by the hypervisor's NAT, verify connectivity before investigating SIP. Do not place the PBX directly on the internet to simplify the test.<\/p>\n Use this path on a freshly installed, dedicated Debian 12<\/strong>. The installer sets up dependencies and repositories: do not run it on the server that already hosts your website, database, or other PBX. Open an administrative session on the VM; the commands below are for it, not for your workstation. If you use SSH, prepare a persistent session with Inside the If your minimal installation does not have If the installer indicates that a newer version exists, download the file again instead of bypassing the check with A installation documentation<\/a> explains the options. Prefer to boot a VM from a ready-made media? Get the current link and the SHA-256 at download page<\/a>, verify the integrity of the image and follow the official ISO guide v17<\/a>. The installation still depends on the network to download components.<\/p>\n Set the VM firmware to UEFI<\/strong>. The ISO menu offers installation profiles, called spice levels<\/em>. For a SIP-only lab, without physical telephony cards, the guide suggests Basic \u2192 FOG<\/strong>.<\/p>\n The ISO directory<\/a> also contains historical images. Being available for download does not mean it is still supported. To get started, follow the file indicated on the current page, not an old ISO chosen by name.<\/p>\n ISO v17 disables the root login by default. Depending on the profile, the installation may display a generated password for the user Once the installation is complete, open the VM's IP address in the browser on the lab network. Create the administrative credentials and configure secure access. In production, use HTTPS with a valid certificate and restrict administration to the management network or VPN.<\/p>\n The extension, caller ID, and password fields are described in the PJSIP extensions guide<\/a>. Some screenshots from the wiki belong to earlier versions; follow the PJSIP type, not old tutorials that choose chan_sip.<\/p>\n For a first check from the PBX terminal:<\/p>\n If you are already root and do not have The MicroSIP shown in the video is an option for Windows. To participate in the same lab from a Linux desktop, you can use one of the SIP clients below. The client runs on the user's workstation, not on the FreePBX server.<\/strong><\/p>\nReference: October 2026.<\/strong> FreePBX, Asterisk, Debian, and the ISO image each have their own versions and release cycles. Do not confuse the ISO number with the version of all the modules, and do not use an old demo as an automatic indicator of the project's current state.<\/div>\nFreePBX does not replace Asterisk<\/h2>\n
What changes in FreePBX 17<\/h2>\n
\n
Architecture: panel, extensions, and carrier<\/h2>\n
![\"Arquitetura](\"\/wp-content\/uploads\/2026\/10\/freepbx17-arquitetura.webp\")
Prepare a separate lab<\/h2>\n
Option 1: install with the official script<\/h2>\n
tmux<\/code> before starting the installation.<\/p>\n# Na VM Debian 12, abra uma sess\u00e3o root\nsudo -i\n\n# Confira o sistema e a conectividade local\ncat \/etc\/os-release\nip -br address\nip route\n\n# Instale o utilit\u00e1rio de download, se necess\u00e1rio\napt-get update\napt-get install -y wget ca-certificates less tmux\n\n# Abra a sess\u00e3o persistente antes da instala\u00e7\u00e3o\ntmux new -s freepbx<\/code><\/pre>\ntmux<\/code>, execute the following block. If the SSH connection drops, log back into the VM, open the root session, and resume with tmux attach -t freepbx<\/code>.<\/p>\n# Baixe o instalador oficial, sem execut\u00e1-lo diretamente do download\nwget -O \/tmp\/sng_freepbx_debian_install.sh \\\n https:\/\/github.com\/FreePBX\/sng_freepbx_debian_install\/raw\/master\/sng_freepbx_debian_install.sh\n\n# Leia o script; pressione q para sair\nless \/tmp\/sng_freepbx_debian_install.sh\n\n# Depois da revis\u00e3o, execute a instala\u00e7\u00e3o\nbash \/tmp\/sng_freepbx_debian_install.sh<\/code><\/pre>\nsudo<\/code> or less<\/code>, use the administrative access configured during the Debian installation and install the necessary utilities. The official README<\/a> documents the download and execution as root. Do not add --testing<\/code> for a normal installation.<\/p>\n--skipversion<\/code>. In another administrative terminal, monitor the most recent file created in \/var\/log\/pbx\/<\/code>:<\/p>\nls -lt \/var\/log\/pbx\/freepbx17-install*.log\n# Use o nome exato listado acima:\n# tail -f \/var\/log\/pbx\/freepbx17-install-DATA-HORA.log<\/code><\/pre>\n--opensourceonly<\/code> selects only open FreePBX modules; --dahdi<\/code> is relevant for compatible telephony hardware. Do not add DAHDI in a purely SIP lab without need: the drivers have specific kernel dependencies.<\/p>\nOption 2: install via ISO<\/h2>\n
# No diret\u00f3rio onde voc\u00ea salvou a ISO:\nsha256sum SNGDEB-PBX17-*.iso\n# Compare o resultado com o SHA-256 da mesma imagem no site oficial.<\/code><\/pre>\nPay attention to the disks:<\/strong> the documentation warns that multiple disks will be incorporated into RAID and that USB sticks of 100 GB or more will be treated as members and wiped. Use a USB media smaller than 100 GB; the guide also reports possible issues with NVMe. For the lab, keep only one empty virtual disk. On physical hardware, disconnect disks that will not be used and read the options before confirming.<\/div>\nsangoma<\/code>: if this screen appears, save the password. Follow the guide to set or change the credentials and use sudo<\/code> for administrative tasks. It is not necessary to resort to GRUB password recovery just because the root login is not available.<\/p>\nFirst access and two PJSIP extensions<\/h2>\n
\n
sudo asterisk -rx 'core show version'\nsudo asterisk -rx 'pjsip show endpoints'\nsudo asterisk -rx 'pjsip show contacts'<\/code><\/pre>\nsudo<\/code>, skip that prefix. These commands help check the installed Asterisk, the endpoints, and their contacts. The functional proof is still the call being completed with bidirectional audio, not just the presence of the extension in the list.<\/p>\nSoftphones for Debian and Ubuntu<\/h2>\n