SipAccount now uses only v2 APIs for lookup, create, update, and delete

* bye-number-display:
  This command is redundant with the info command

All the previously-lazy BackendSgx data is now either all loaded or all not
loaded by swapping the sgx_repo used by your CustomerRepo instance.  When not
loaded the fields are filled with bottom values that explode when used.  When
loaded the values are present in RAM and not promises at all.  Most code paths
do not need any of the data, a few need most of it, so this seems like a good
trade-off.  Most code using this object will simply never touch those fields or
care about how they are loaded, etc.

Of course, most of this data isn't even SGX related and should move out of here,
but that would take a data model refactor/migration on the catapult_* schema.

Currently we tell users to use the web activation for this, but we'd like the we
activation to go away very soon and will still need a path for them.

Command execution is normally two stage, execute, response, then submit.  We had
a hack here that allowed submitting the form directly, but this allows a blind
two-stage client to also at least work.

So transform reasonable things into what we know Bandwidth expects.

* sip-outbound:
  Support transcription disablement option
  Port in inbound calls + voicemail
  Allow fetching fwd timeout as well
  Get OGM for a customer
  Helper to fetch customer's vcard-temp
  Make Disposition more real
  Allow constructing CDR for an inbound or outbound event
  Outbound calls from v2 SIP endpoint work and save a CDR

The craziest part of this is the workaround for a serious bug in Bandwidth's
HTTP voice API (which they may yet fix, still negotiating with them about that).

When a call comes in, every 10 seconds that it is not "answered" the inbound
call gets cancelled by their upstream peer and then get retried.  The caller
sees only one oubound call for this, so it doesn't look odd to them, but to us
it looks like they keep hanging up and trying again every 10 seconds.  So what
we do for now is we wait 2 seconds after they disconnect before we decide
they're really gone.  If they call back in those 2 seconds we just connect the
eventual bridge or voicemail to this new call and everything works out.

Ew.

* number-search:
  Do not direct back to website to pick new number
  Run rubocop after tests
  Add reference to option to show where the tel is
  Format tel for human reader
  Allow user to search for numbers over XMPP
  s/WebRegisterManager/TelSelections

Instead, if number is no longer available keep them in flow and use the number
search tool.

* finish-btc:
  No more legacy session for BTC
  Do not lose WebRegisterManager on retart

Set the same key as web register manager, so that on next register jmp.chat the
tel we were using in this flow will be used.   Not needed if they came from web
register, but will still extend expiry in that case and no harm.

Clean up pending tel key on Finish.

Store web registrations in redis.  Set an expiry so they don't grow in RAM
forever as they previously would have without a restart.

We moved some of the currently private things to be public, like currency, and
then included things like how much the monthly price is as well.

Execute works, but next is more right