Use pjsip or sip w/ FreePBX14/Asterisk13?


(Mvogel4949) #1

Is everyone using pjsip for extensions now? I’m thinking of making the switch but wondering if there are any pitfalls I’ll run into. SIP so far has been pretty plug and play but if it isn’t being supported than I suppose it’s best to switch. Not sure if it matters but I’m using FreePBX14 with Asterisk 13 on my systems.


(Joshua C. Colp) #2

Speaking from the Asterisk side we’ve seen a continued move of people from chan_sip to chan_pjsip and there’s been no blockers for quite some time. The problems we’ve seen have really been configuration related, and the fact that it’s just configured differently using a different philosophy by embracing standard SIP concepts.

Just as a nice point - Sangoma’s commercial product, Switchvox, has exclusively used PJSIP for the last 4 years for both endpoints and providers. The chan_sip module isn’t even built or present.

disappears back into the Asterisk void


(Lorne Gaetz) #3

The only pitfall you might see now is a recent framework issue that defaulted the ‘rewrite contact’ parameter for pjsip extensions to no, when it should be yes. Issue is resolved, but the framework may still be in edge.

I would not expect you to see any difference. You will probably want to start a little bit at first, which means having both drivers enabled and bound to different ports. You need to be very disciplined when dealing with SIP ports, always being aware which driver is using which port, and configuring firewall rules and sip devices accordingly.


(system) closed #4

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