~singpolyma/sgx-jmp

ref: 2f7a1b60192a14c86ef9a1f727ea9369cfedc2c6 sgx-jmp/web.rb -rw-r--r-- 7.1 KiB
Switch to the new Ring verb with answerCall=false

Should actually cause incoming calls to ring properly, and the bug that required
pseudo_call_id seems gone.
ogm is sometimes a promise
chmod after rename, because Tempfile is always 0600
Merge branch 'rubocop'

* rubocop:
  Additional fixes for rubocop 1.10.1
  Switch to rubocop 0.89.1
Switch to rubocop 0.89.1

This is the rubocop in new Debian stable
Merge branch 'configure-calls-v2'

* configure-calls-v2:
  New configure calls command
  Move more persistence into the repo layer
  Easy DSL for adding XEP-0122 validation to fields
  CustomerFwd uses ValueSemantics, translates old XMPP-SIP URI
CustomerFwd uses ValueSemantics, translates old XMPP-SIP URI

More of the original data is kept now, so this object could be used for putting
to persistence as well as for loading from it.
Record Voicemail Greeting command
Move CustomerFwd behind Customer

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.
Merge branch 'sip-outbound'

* 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
Support transcription disablement option
Port in inbound calls + voicemail

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.
Outbound calls from v2 SIP endpoint work and save a CDR