What is the supported method to have one user and multiple devices

[fceledon]

In the past, I created individual per device, 6 digit long extensions numbers, grouped on a four digit ‘group’ matching the last four DID digits. This worked fine, no issues, all sharing the same mailbox. Internally, users only dial 4 digits.
Today, with PJSIP, what is the accepted/supported method?
Just one 4 digit user and multiple 4 digit devices registering with the same authorization Id?

[cynjut (Dave Burgess)]

@fceledon

Today, with PJSIP, what is the accepted/supported method?
Just one 4 digit user and multiple 4 digit devices registering with the same authorization Id?

Yup - you’ll need to adjust the ‘instance’ counter, but that’s the approach. As long as you’re using regular SIP devices, you should be good to go. Note that ‘WebRTC’ is an outlier on this, so if you are using that, PJ-SIP may not be the best method and using your Ring Group approach will probably work better.

Personally, I’d not use WebRTC and go with the PJ-SIP solution, but I’m adverse to pain.

[fceledon]

Thanks. Indeed if Web_RTC is required at one point, we can create a legacy SIP device.

[TheWebMachine]

@cynjut

Note that ‘WebRTC’ is an outlier on this, so if you are using that, PJ-SIP may not be the best method and using your Ring Group approach will probably work better.

I thought that WebRTC connections were just creating/using the same PJSIP virtual (99XXX) extension numbers like a normal device registration now?

[cynjut (Dave Burgess)]

There is an active discussion about this going on right now - I don’t use it, so I’m just repeating what I understood the issue to be.

[fceledon]

It’s hard to ignore your advise, but pjsip It’s much simple than the old ring/extensions method and easy to maintain.
On my test I was able to connect a softphone w/Linphone (using WIFI and Cell data), local (LAN) LinksysSPA phone and a remote phone through NAT.
All worked ok. Now, I may miss some functionality (unknown to me).
The only service needed is phone and voicemail.
I’m missing something?

[cynjut (Dave Burgess)]

@fceledon

All worked ok. Now, I may miss some functionality (unknown to me).
The only service needed is phone and voicemail.
I’m missing something?

To be clear, ChanSIP is deprecated and will be removed from Asterisk in the next version release (if my notes from Astricon are true), so …

I don’t know what to tell you. If it works and works the way we’ve all suggested, then I’d say you should be good to go. I don’t know what functionality you might be concerned about missing, but I was skeptical for a couple of years but even I’ve come around.