When using SIP you can direct dial to a device at a particular IP address and port using this form in your Dial statement: Dial(sip/$OUTNUM$@dotted.i.p.address:port) (where OUTNUM is replaced by the number you are sending to the device). But if you are trying to use PJSIP only, and you change SIP …

It has to be associated with an endpoint. An endpoint defines the configuration to use when talking to the given URI.

Thanks for the reply. That does not really make sense to me but could an endpoint also be a PJSIP extension? That would make a lot more sense than a trunk in this particular case.

Okay, great, I will use an extension instead of a trunk then. Thanks!

You could also just define an endpoint that you use for outbound dialing to a SIP URI.

Oh okay, that would work too. But I already have an extension associated with that device so in this case the extension would be the most logical choice, I think.

Yes, that is perfectly fine then.

Direct dial to IP address and port works in SIP but not PJSIP

Asterisk Asterisk SIP

jcolp November 19, 2019, 12:23am 4

Extensions and trunks are no different in PJSIP. They are all endpoints, with an endpoint being something you speak SIP to - an ITSP, a gateway, a phone, etc.

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Migrating to PJSIP - endpoint was not found

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Direct dial to IP address and port works in SIP but not PJSIP

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