Questions about soft-clients

Is the best-practice to set up soft-client users with the same extension as their traditional endpoint extension number or create a new extension exclusively for soft-client usage.

I can think of advantages and disadvantages both ways so I thought tossing it out for discussion would be beneficial.

Thank you in advance for your responses.

If the endpoints were simple enough, I think it would be nice to have just one. Less objects to manage. If the endpoints were different enough, you might have to split them out due to functionality conflicts.

I have found if you use the same ext # and password in 2 different devices (ie: softphone and deskphone) they conflict and only 1 will work - usually the one that registers first.

What I do is set up 2 ext’s and then put them both in the same ring group.

It would be easier to use 1 ext for 2 different devices but I have yet to get it to work.

Hope this helps

You can do that in pjsip adjusting the max contacts field.

I use Voip Innovations for my origination and termination. I have yet to get pjsip to work flawlessly with all parties concerned. Too buggy

You can (and probably should) use chan-sip for trunking, but you equally can use chan-pjsip for your extensions, Asterisk is a B2BUA.

If you use chan-sip for trunks and pjsip for ext’s, do you use different ports or do you universally use 5060?

Asterisk will bind chan-{sip|pjsip] individually to the port so defined and on the interface(s) defined (no they can’t be the same) , VI is presumably IP based for you, so you might use iptables to “remap” VI’s port 5060/udp connection on your firewall to your bound chan-sip port on your server , no matter what port that might be.

So the answer to your question is 42 :wink:

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If you have the time and nerve it would be worth investigating those pjsip problems, posting them here and have people look at it to maybe provide a bug report to Digium.

Problem with pjsip is, that people think it has got some bugs now, but that will be different in a few years and it will improve.
No, it won’t, cause there are hardly any bug reports on the Asterisk bug tracker for pjsip.

I use VI as well and found that Chan-SIP is the way to go with them. You can set up your inbound connection from VI to any port you want, but they require their connections to be on 5060. It’s not hard to set up, and there is a little mental gymnastics you have to do to keep which-port-goes-where in the config, but it can be done.

Note that being stingy on who can connect to your system on the SIP ports is recommended. I only allow VI and my other ITSPs access to the server on port 5060 through my firewall.

VI and I actually did a lot of this work last year (maybe 18 months ago). At the time, the stumbling block was the lack of a user-less login/registration mechanism for the connection to the ITSP. It’s probably improved since then, but I still use Chan-SIP for all of me ITSP connections and use PJ-SIP and/or Chan-SIP for local connections. You just have to (once again) keep “what port is what” while you are working.

I am not sure that I understand your response. Maybe just overly-thick today.

I am using PBXact and Bria Mobile until the Sangoma soft-client becomes available.

As many have expressed, it would be ideal to have one extension per user, regardless of what endpoint they are using though I still have yet to get it to work.

Still hoping and looking.

Thank you.

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