With regard to “allow anonymous calls” the calls come in to a context that does not have access to the internal dialplan. Login to your Elastix GUI; Click on the Down arrow which is Next to Reports > Click on Security; Click on Advance Settings > Turn ON “Enable direct access (Non-embedded) to FreePBX” and “Enable anonymous SIP calls” Asterisk PJSIP Troubleshooting Guide - Asterisk Project Wiki More information about encrypting SIP calls can be found in the section called “Encrypting SIP calls”. If you really want anonymous calls, then you will have to setup your dialplan with a guest/anonymous context for the calls to drop into. After this, all dubious calls will be sent (more or less directly) to the Nirvana. externip takes an IP address as its argument. SIP trunk desetination was 109.224.23.36 before now i changed it to 172.16.4.2.. outboud calls are going fine. Set Allow SIP Guests to NO. Asterisk SIP Settings (FreePBX wiki - Asterisk SIP Settings User Guide): Set Allow Anonymous Inbound SIP Calls to NO. To help \ understand how this works, set verbose up to 10 in the Asterisk CLI and then call \ into your PBX using a SIP … P-Asserted-Identity: sip:[email protected]. From the navigation bar at the top of the page, click on Connectivity >> Trunks. So of course we're now getting blasted with spam/hack attempts. This would not be a normal situation at all. It is not very dangerous unless you have something configured in an insecure way to … Dial Patterns : 2XX ( Replace with the format of your IP Office extension ) Trunk Sequence: SIPIPO. Getting Started with Asterisk/FreePBX [SureVoIP Support] I installed the Asterisk/FreePBX distro, and enabled TCP and UDP pjsip, both on port 5060. Asterisk

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