Asterisk Configuration - SIP – OnSIP Support
pfSense port settings for Asterisk FreePBX - Outside Open Jul 10, 2016 WebRTC and Asterisk 14 - Asterisk Blog Aug 23, 2017 NAT Configuration FreePBX 12 - PBX GUI - Documentation
Firewall/NAT Checklist - Digium
Troubleshooting - "AllStarLink Wiki" Jan 20, 2019 SIP and RTP Routing - Asterisk Blog
Использование Asterisk вместе с NAT. а также, адрес прокси-сервера (т. е. адрес нашего *) указывать с учетом внешнего порта, какой форвардиться на * (х.х.х.х:ext_port).
With the introduction of the Asterisk SIP Settings module, most SIP settings are made available in the GUI. This includes the all important NAT, External IP, Local Network, Enabled Codecs and Codec order. Logging In. From the top menu click Settings; From the drop down click Asterisk Sip Settings; Settings. Allow Anonymous inbound SIP Calls No audio on Asterisk SIP call - Stack Overflow Besides NAT problems I've also faced this issues on 3 cases: 1) Missconfigured parameter localnet: on /etc/asterisk/sip.conf make sure you set the network address for the phones. You can alos add multiple networks, for example: localnet=172.16.1.0/24 localnet=192.168.1.0/24