I’m trying to wrap my head around a pjsip extension. Typically for remote extensions and sip trunks i forward port 5060 to the freepbx. If I am running a remote pjsip extension do I also need to port forward the port I am using for pjsip? Thanks
That port is what you define. In the extension page at the top you can see what port is being used.
If that extension is a remote extension do I have to have firewall rules in place to allow traffic in/out on that port?
If Responsive Firewall is enabled, it will allow registrations without adjusting the Firewall zones of the SIP services. If Responsive is not enabled, you will need to white list the remote host or re-zone the PJSIP service to Internet (NOT recommended!)
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Yup, and you need to make sure whatever port redirection from the firewall needs to be there and is working.
Without trying to confuse you, you can open and port forward any router port to the PJ-SIP port on your PBX. So, your remote extension can be configured to contact your server at UDP port 2345 and the router can redirect that port to your SIP port on the PBX. This provides some “security through obscurity” which blocks about 90% of the script kiddies from trying to break into your phone system.
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Does pjsip also use rtp ports 10000-20000 like the ChanSIP settings?
It does. The fundamental rules of SIP do not change with the driver.
So I have an odd case of a remote SIP phone connecting and having audio. A remote pjsip extension registers but with no audio. Typically I would look at the NAT setting of the extension but pjsip doesn’t have a NAT setting. Any ideas here?
What is your Asterisk version, IIRC there is one with RTP issues on remote PJSIP extensions.
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Asterisk V13.14, I’ll up them to the newest which I believe is 13.18. Thanks for the insight, that should have occurred to me.