SIP/$OUTNUM$ over PJSIP

Greetings to all,

I installed a FreePBX 16.0.19 Asterisk 16.25.0 that does NOT have SIP active. It only has the PJSIP.

Previously you could create a CUSTOM trunk for OUTBOUND only like:

SIP/[email protected]:5060

Currently it doesn’t work if FreePBX OLD SIP is off, I tried: PJSIP/[email protected]:5060 and it doesn’t work either.

I don’t want to activate SIP as it will disappear very soon.

Any Suggestion?

https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels

chan_pjsip allows you to override both the user and domain on an endpoint, but does require an endpoint. As such, you need to provide a placeholder endpoint with details like the transport and codecs to use.

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