6

VOIP. Voice over IP. It requires quite a few more things to be configured than plugging a phone into ethernet, but that puts you in the neighborhood.


4

You ask multiple questions, I can answer some, but not all of them. why asterisk is running on loopback address and not on 192.168.32.181:5038 As far as I can tell from a few searches port 5038 is the asterisk management interface. So I assume it binds to 127.0.0.1:5038 for security reasons. You wouldn't want that to be accessible from everywhere. Why ...


4

I'm currently doing the same on my organization. The first thing to know is that there is two ways to use Skype for Business as the phone system: Using PBX Connector to join existing On Premises telephony system. Using Cloud PBX and a PSTN Calling Plan. We were using sip lines hosted directly on our provider by the past, so as we never had on premises ...


1

Yes we experienced an Asterisk issue years ago where DTMF received over a trunk would cause Asterisk to drop the call. It's been long enough that I don't recall the fix but it was hell troubleshooting to find why the calls were being dropped. Usually from an accidental keypress on a cell phone while driving.


1

You can try tSIP softphone. There are few event types like "on call state" to which Lua scripts can be assigned. From Lua script you can use ShellExecute + curl or other similar utility. If you need SIP Call-ID or other usually obscure line then it can be extracted from initial INVITE text that Lua can access.


1

Like you I know nothing of the voip world but try the following in sip.conf: [general] directmedia=no If it works then great, but I cant offer an explanation as I haven't checked into it, start here though.


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