Redirect a call to SIP URI when endpoint is unavailable

Dialing a URI via PJSIP must be done using an endpoint - see Dialing PJSIP Channels - Asterisk Project - Asterisk Project Wiki

In FreePBX - Asterisk SIP Settings, enable Allow SIP Guests so that it generates the PJSIP “anonymous” endpoint:

Then for your custom extension, use as the Dial string:

PJSIP/anonymous/sip:[email protected]


Alternatively, set up a PJSIP trunk with no authentication or registration, just specifying the domain of sip.ippi.com:

then if you named that trunk “ippi” for example, use as your custom Dial string:

PJSIP/nickname@ippi

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