~singpolyma/sgx-jmp

ref: f571b5e726750cf8c4523c22d77e110194c3e16a sgx-jmp/Gemfile -rw-r--r-- 1.3 KiB
Initial tests for web routes
Customer Info Forms and More Info

There's a bit extra info I wanted about users, so since I was doing this
anyway I figured I may as well port the existing forms to the new form
renderer.

Then, in order to fit within the guidelines I needed subforms, so
partials were added.
Merge branch 'whitelist'

* whitelist:
  Allow whitelisting domains
  Use FormTemplate for activation form
Allow whitelisting domains

Some domains credit new users to the admin, some to no one.  We still have to
show a form to get them to pick a plan, but otherwise skip most of the process,
activate them with 1 month and go to Finish.
Lock rubocop to the version actually in use
Need latest em_promise.rb
Fix for faraday+em_http and gzip
New bundler requires these to be here to use the commands
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 'reset-sip-v2'

* reset-sip-v2:
  After SIP reset, offer to change inbound fwd
  Reset sip account using v2 API
Reset sip account using v2 API

SipAccount now uses only v2 APIs for lookup, create, update, and delete
We don't really want remote, we just want stdin to not be closed by blather...
Use fixed em-http-request
Record Voicemail Greeting command
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
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
Next