aqeel.sip
available for hire
SIP/2.0 · establishing session

Aqeel Abbas

I build the voice
that connects the call.

Every connection begins with a single request.Scroll to trace the session — from INVITE to 200 OK.

carrier signal
scroll to trace the session ↓
INVITEclient → you

The call is being placed to you

A team needs telephony that actually works — carriers, call control, recordings, real-time voice. That's the INVITE. I'm the endpoint that answers: a VoIP & backend engineer who has built the whole path, from the SIP trunk to the product on top.

100 Tryingyou → client · provisional

Working through the stack

Provisional response received. Here's the toolkit I reach for while the session sets up.

Telephony

AsteriskFreePBXSIPRTPAMIARISIP Trunking

Backend

JavaSpring BootAsterisk-JavaREST APIsWebSocketsWebhooks

Voice / AI

Realtime Voice AIDeepgramAssemblyAIGoogle Cloud STTGoogle Cloud TTSLive Transcription

Automation

Make.comn8nTelegram Bot APICrypto Payments
180 Ringingyou → client · provisional

Ringing — this is what you're calling for

Real-time media

RTP, recording and streaming audio pipelines.

Event-driven control

Calls driven by AMI/ARI events, not guesswork.

Low-latency voice AI

Speech in, speech out, fast enough to feel human.

200 OKyou → client · answered
session established

Connected. Here's what I do.

  • Design and run SIP trunks, extensions, dial plans and IVRs on Asterisk / FreePBX.
  • Control live calls programmatically — initiate, route and end — over AMI & ARI.
  • Wire telephony into real-time voice AI, cloud TTS/STT and live transcription.
  • Ship the GUIs and REST backends that turn all of it into a usable product.

0+

clients connected

0+

projects delivered

0+

years on voice & backend

0%

answered on first ring

ACKclient → you

Handshake complete — the media channel is open. Now the real payload flows. Here's the work I've shipped.

RTP MEDIAclient ↔ you · bidirectional

Media stream: selected work

Six sessions I've built end-to-end — each one a real telephony or real-time voice system.

RTP/01200 OK

Crypto-Metered Calling Bot

Telegram bot + VoIP billing engine

A Telegram bot that provisioned talk-time for callers based on crypto payments, then metered live calls against their balance in real time.

  • Credited caller minutes automatically on confirmed crypto payments
  • Consumed AMI call events to decrement balance per second of talk-time
  • Hung up gracefully the moment a balance ran out
Telegram Bot APIAsterisk AMICrypto PaymentsJavaSpring Boot
RTP/02200 OK

Make.com Outbound Call Automation

Integration & IVR flows

Automated outbound campaigns wiring a programmable voice API and FreePBX into Make.com — placing calls, collecting caller input, and persisting results.

  • Triggered outbound calls from Make.com scenarios
  • Collected DTMF tones as structured caller input
  • Fired an HTTP request to save each recording/result to storage
Programmable Voice APIMake.comFreePBXWebhooks
RTP/03200 OK

Recording + Google Cloud Speech

Call media + speech pipeline

A recording pipeline that captured call audio and ran it through Google Cloud for text-to-speech prompts and speech-to-text transcription.

  • Recorded call legs and stored media for downstream processing
  • Generated dynamic voice prompts with Cloud Text-to-Speech
  • Transcribed conversations with Cloud Speech-to-Text
Google Cloud STTGoogle Cloud TTSAsteriskAudio Pipeline
RTP/04200 OK

FreePBX Dialer Platform

Full telephony platform

End-to-end platform: a SIP trunk on Asterisk/FreePBX, an Asterisk-Java backend driving calls over AMI/ARI, and a GUI for bulk campaigns.

  • Configured SIP trunk, extensions and dial plans on the PBX
  • Controlled calls (initiate / end) from a Java backend via AMI & ARI
  • Built a GUI to attach an audio file and upload a CSV of numbers
  • Connected GUI to backend over REST with real-time call state
AsteriskFreePBXAsterisk-JavaSpring BootAMI / ARIREST
RTP/05200 OK

Real-Time Voice AI for Orders

Conversational voice agent

A low-latency voice agent that answered calls, understood spoken orders, and completed them in a natural back-and-forth conversation.

  • Handled live orders end-to-end over the phone
  • Streamed speech in and out with minimal perceptible latency
  • Confirmed and finalized structured orders from natural speech
Realtime Voice AIDeepgramAssemblyAITTSTelephonyLLM
RTP/06200 OK

Live Communication Transcription

Real-time transcription service

A real-time transcription service that turned ongoing conversations into live, streaming text for agents and downstream systems.

  • Transcribed live conversations as they happened
  • Streamed partial and final results with low latency
  • Fed transcripts into downstream tooling in real time
DeepgramAssemblyAIStreaming STTWebSocketsSpring Boot
QoSquality report

Call quality: what callers say

Verified 5.0 rating across 19 sessions on Fiverr. Client handles are hidden by request.

QoS · 5.0 / 5.0
worked with a few times now and he’s great
B Verified client · United StatesVoIP & Telephony
was able to understand the request in one go, no need to explain anything in detail, was self understand on what the requirement was from the client, has good knowledge of VOIP and is ready to adjust as per the requirement of the client
T Verified client · IndiaVoIP & Telephony
Highly professional, sincere, and efficient. I wholeheartedly recommend their services.
U Verified client · TunisiaVoIP & Telephony
Good communication with great professionalism
K Verified client · United KingdomVoIP & Telephony
BYEclient → you

Ready to hang up — or start the next call?

If your product needs voice, telephony or real-time backends that hold up under real traffic, let's talk. I'll answer on the first ring.

200 OKyou → client · call complete

Session terminated cleanly. Thanks for tracing the whole call. // EOF