An in-house Grandstream PBX in boxed Bitrix24 without a paid SIP connector

The company runs its own Grandstream PBX, but boxed Bitrix24 knew nothing about calls: no client card on an incoming call, conversations did not reach the CRM, and clients had to be dialed from the handset by hand. The stock route is a paid SIP connector with cloud telephony.

There was no live PBX during development — no network access to it, no accounts. Calls had to be recorded through the core's stock telephony so that regular cards, reports and call details work, and a lost connection to the PBX must not lose events.

• An AMI listener keeps a persistent connection to the PBX, puts events in a queue and assembles a call by Linkedid: incoming, outgoing, pickup, transfer
• Calls are recorded through the core's stock telephony: the call card for the employee, a "Call" activity, call details, a lead for an unknown number
• "Call" from the CRM dials from the browser softphone, and if it is closed — via AMI, first to the employee's desk phone, then to the client
• A WebRTC softphone on JsSIP through the portal's nginx: incoming calls, hold, DTMF, transfer; SIP passwords are stored encrypted in the database
• Call recordings are streamed from the PBX; anything missed during an outage is backfilled from the PBX call log with retries
• An admin panel for listener and queue status, a PBX simulator on Asterisk 20 in Docker, and tests on recorded AMI sequences