SIP URI support

The ability to call in and out through the VSTG using SIP URI's might be very useful. i.e., allow inbound/outbound SIP calls as vocerausername{.site?}@myserver or the like. We juggle a lot of telephone numbers / DID's and struggle with effective allocation. This could open up some new avenues.

Beyond this ...

* Ability to customize URI userpart and domain part to some degree, maybe per site, and/or allow override/multiple userparts, at the user level.
* Not sure if some means of setting incoming permissions (can't let just anyone call "") is called for, or if this should be at the IP PBX.
* Permit assigning a URI/alias to a group. Otherwise, the only unique field I am aware of for groups at the moment are extensions.
* Outbound, allow control over URI vs TEL calling as well as caller ID behavior (I know basic CID control is already in there) on a per-destination basis.
* A little further out there ... probably a separate idea entirely ... interoperate with other "presence" system(s).

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  • Feb 1 2020
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