Real-Time Voice Test & System Diagnostics
100% reliable internal extension dialing, WhatsApp Cloud API audio, and dual-channel recording.
Sub-millisecond SIP signaling with SRTP-SDES media proxying on TLS 5061 and UDP 5060.
Trunk: wa.meta.vc:5061 with PCMU codec and Coexistence mode enabled.
Left Channel: Customer Audio | Right Channel: Agent Audio (WAV 16-bit PCM).
Call Detail Records (CDR) & Dual-Channel Audio
Complete audit log of inbound, outbound, and internal calls with playable waveforms and AI summaries.
| Call UUID / Time | Direction | Channel | Caller ID | Destination | Duration | Quality | Stereo Recording | Action |
|---|
Agent & Extension Matrix (Softphone Provisioning)
Total 111 accounts (Main Admin: hola, WhatsApp: 01-10, Botanika: BB01-BB100). Auto-generate extensions or reassign old extensions.
| Extension(s) | Agent Name / Email | Role | Telephony Permissions | Presence | Actions & Softphone Provisioning |
|---|
Meta WhatsApp Cloud API Direct Audio SIP Trunk
Carrier-grade SIP bridge interconnecting Meta VoIP cluster (wa.meta.vc:5061) with FreeSWITCH.
tgrp=wacall
graph.facebook.com
183 Session Progress
Coexistence Mode (WhatsApp Business Mobile App + CloudPBX)
Agents can take voice calls via WebRTC / FreeSWITCH softphone while continuing to chat on WhatsApp Business mobile.
Registered WhatsApp SIP Calling Directory
All production WhatsApp numbers configured across Botanika Bharat, AWEBCS, SMPPKing, Rishabh Instruments, Farm Culture Agro, and DATAMATICS.
| WhatsApp Number | Profile Name | Business Entity | Phone ID / WABA ID | Username | Trunk Route | Actions |
|---|
Growth Ka Raasta Corporate Jingle
Streams continuously into caller's ear while ringing agents using FreeSWITCH SIP 183 Session Progress. Terminates the exact millisecond an agent picks up (0ms audio bleed, zero caller billing).
<action application="set" data="instant_ringback=true"/>
<action application="set" data="bridge_early_media=true"/>
<action application="set" data="transfer_ringback=/var/lib/freeswitch/sounds/tenants/default/Growth_Ka_Raasta_8k.wav"/>
Automated FFmpeg Audio Transcoding Pipeline
Converts consumer audio files into carrier-grade, uncompressed broadcast streams conforming to ITU-T and EBU standards.
Upload any format: MP3, WAV, AAC, M4A, FLAC via Web Portal.
FFmpeg -ac 1 eliminates phase cancellation over telephony channels.
Generates 8kHz PCM for G.711u and 16kHz for Wideband G.722 HD.
Loudness locked to -16 LUFS with -1.5 dBTP ceiling to avoid ear fatigue.
Dialplan updates dynamically via mod_xml_curl without restarting FreeSWITCH.
Corporate Audio Library
2 Assets| Title | Category | Format | Loudness | Action |
|---|
Upload & Transcode New Audio
Phone Numbers & Inbound Routing
| Number (E.164) | Type | Provider | Label | Routing Target | Status |
|---|
Contact Center ACD Queues
| Queue Name | Strategy | Assigned Extensions | Max Wait Time | Virtual Hold Callback | Total Calls Served |
|---|
Live Call Wallboard & Quality Monitoring
Supervisors can listen silently, whisper guidance to agents, or barge directly into calls.