Hire the Best Freeswitch Specialists
in Pakistan

I build, integrate, and debug VoIP and calling systems on Asterisk from a single SIP trunk to multi-tenant calling platforms. Whatever the calling problem is setup, integration, broken audio, or a full build I take it to production and document it for your team. What I help with: → Asterisk, Wazo, FreePBX, FusionPBX — setup, configuration, and production debugging → SIP trunking with any provider — Telnyx, Twilio, Plivo, Bandwidth → Inbound/outbound call routing, IVR, queues, CDRs, recordings, webhooks → Kamailio + RTPEngine — SIP routing and media bridging → WebRTC dialers & softphones — JsSIP, SIP.js, MicroSIP, Zoiper, Linphone → WhatsApp Business / Meta Calling — SIP gateway, DTLS-SRTP, ICE, RTP → NAT, codec, and one-way-audio troubleshooting on live SIP traces (sngrep, Wireshark) → Multi-tenant VoIP SaaS with REST API management → Voice AI & call-automation pipelines on top of SIP (LiveKit, agents) → Self-hosted on DigitalOcean, Azure, AWS — any Linux host Recent proof: I built multi-tenant WhatsApp Business Calling on a Wazo/Asterisk platform — a Kamailio → RTPEngine → Asterisk media bridge handling DTLS-SRTP from Meta's SIP gateway, with a WebRTC dialer, per-tenant routing, IVR, recordings, CDRs, and full API control. The client kept me on for ongoing work. I've also deployed self-hosted Asterisk on DigitalOcean with full trunk and dial-plan config — production, not a test lab. Tell me what you're building or what's broken, and I'll tell you straight if I'm the right fit. — Zahid

Reman Ali is VOIP and IT Professional with more than 5 year+ of in-depth knowledge & expertise in leading Cisco, Asterisk VoIP, Freeswitch, Vicidial and FreePBX. Vast experienced in the support and development of VOIP & telecommunications projects. Excellence communication skills with highest level of customer support. Diverse expertise on operating systems (Linux & Windows) and multiple CRMs.

Hello, Thanks for visiting my profile. This is Abid Ur Rehman, a certified Asterisk Developer DCAP, VoIP Consultant, VICIDial, FreePBX SCE, 3CX v18 and v20 Certified Advanced Engineer, Elastix ECE, GoAutoDial, The SIP School SSCA, PBXWare, A2Billing, FreeSWITCH, Twilio, FusionPBX, ASTPP, Network CCNA, Windows Servers MCSE, and Linux RHCE certified expert. I graduated with a BS in Telecommunication and an MSc in Telecommunication Networks. Having over 13 years of VoIP/Linux/IT/network planning, design, development, deployment, optimization, administration, and support experience in top-notch organizations such as Telappliant Ltd., Cybergen, SCB Global, Dialer360, and Astericom. Skills and Certification: IP PBX: 3CX, Asterisk, FreePBX, Elastix, PBXware MT, FreeSWITCH, FusionPBX, Issabel, VoIPOffice AI Agents/Integration: ElevenLabs, Vapi SMS Blast/Broadcast using SignalWire, VoIP.ms, Telnyx, Anveo, Bandwidth, Plivo, Skyetel, Twilio, VoxTelesys, BulkSolutions, Flowroute, Intelepeer, and BulkVS Telecom Billing: A2Billing, ASTPP, VOS3000, and Magnus Billing. Dialer: VoiceBroadcast, VICIDial, GoAutoDial, VICIBox Multi-Server Clustering, Dialer360, Primo Dialer IP Phones: Aastra, AND phone, Algo, AudioCodes, Cisco, Cisco/Linksys, CloudTC, Cortelco, CyberData, Digium, H-Tek, Incom, Intelbras, Konftel, Mocet, Norphonic, Obihai, Panasonic, Patton, Phoenix Audio, Polycom, Snom, Thomson, Unidata, Uniden, VTech, Xorcom, Yealink, Yealink/Dreamwave IP PBX (hardware): Sangoma PBXact UC / FreePBX, Elastix IP PBX, Grandstream UCM, Yeaster S-series, Renegadepbx-1U-Appliance-with-FreePBX, Patton's gateways VoIP Providers: USA and Canada: Astericom, Twilio, VoIPdotms, Switch2VOIP, SignalWire, Anveo, Alcazarnetwork, voxbeam, Telus, RingCentral, Call Centric, Voyant (Vitelity), Nextiva, bandwidth, Altotelecom, FlowRoute, QuestBlue, sipdotus, DIDLogic, Nexmo, Plivo, siptrunkdotcom, Telnyx, IDTExpress, Zadarna SMS and Bulk SMS: I can write code for you to send and receive SMS using any Asterisk and FreeSWITCH server and through popular providers like Telnyx, Twilio, Bandwidth, VoIPdotms, and SignalWire. For bulk SMS, I can set up a code with the front end for sending bulk SMS from a CSV . UK: Fuse 2, BT, VoIPTalk, DrayTEL, Telappliant, Gamma, sipgate, VoIPFone, VoiceFlex, Voicehost, Brightstar, Yellowgrid, Gradwell, T4com Ltd, Voxbeam, and Spitfire Linux Distros: CentOS, Redhat EL, Ubuntu, OpenSUSE, Debian, Back Track (Kali Linux) Windows machines: Windows Server 2012 and 2016, Windows Desktop 11, 10, 8, 7, and XP. Cloud Platforms: AWS EC2 and LightSail, GCP, Vultr, BlueHost, HostGear, Hetzner, and Digital Ocean Coding: PHP, MySQL, Lua, XML, JSON, HTML I would be very happy to assist start-ups as well as SMEs in achieving their goals and business objectives. Thank you!!

I am a VoIP and network security expert with over 7 years of hands-on experience delivering custom communication solutions for businesses worldwide. I specialize in deploying and supporting VoIP systems such as Avaya, Asterisk, 3CX, Mitel, FreePBX, FusionPBX and Kamailio, with deep expertise in SIP routing, SIP trunking, and cloud telephony. I’ve worked with major providers including Twilio, Voip.ms, Telnyx, and Voxtelesys, and offer reliable integration and troubleshooting services. I also have a strong background in VPNs, firewalls, and secure networking — including Fortinet, pfSense, OPNsense, WireGuard, OpenVPN, and IPSec setups — especially for remote access and restricted regions. Additionally, I offer AI Voice Assistant setup and integration with VoIP systems, as well as AI chatbots for websites and WhatsApp Business automation. My clients benefit from project-based deployment, long-term support, and a live lab environment where I demo solutions in real time before final delivery and deliver reliable, well-tested solutions designed around your business needs. Let’s work together to secure and optimize your communications systems.

CAREER OBJECTIVES: To perform and grow in a challenging environment with many opportunities to learn and grow both in experience and as a person

𝗔𝗜/𝗠𝗟 𝗘𝗻𝗴𝗶𝗻𝗲𝗲𝗿 | 𝗦𝗶𝗴𝗻𝗮𝗹 𝗣𝗿𝗼𝗰𝗲𝘀𝘀𝗶𝗻𝗴 𝗔𝗹𝗴𝗼𝗿𝗶𝘁𝗵𝗺𝘀 | 𝗥𝗲𝗮𝗹-𝗧𝗶𝗺𝗲 𝗦𝘆𝘀𝘁𝗲𝗺𝘀 | 𝗩𝗼𝗶𝗰𝗲 𝗔𝗜 & 𝗖𝗼𝗻𝘃𝗲𝗿𝘀𝗮𝘁𝗶𝗼𝗻𝗮𝗹 𝗔𝗜 𝗲𝗻𝗴𝗶𝗻𝗲𝗲𝗿 (𝗦𝗧𝗧/𝗧𝗧𝗦 + 𝗿𝗲𝗮𝗹-𝘁𝗶𝗺𝗲 𝗮𝗴𝗲𝗻𝘁𝘀) | 𝗪𝗲𝗯𝗥𝗧𝗖/𝗟𝗶𝘃𝗲𝗞𝗶𝘁 | 𝗦𝗜𝗣/𝗩𝗼𝗜𝗣 (𝗧𝘄𝗶𝗹𝗶𝗼/𝗔𝘀𝘁𝗲𝗿𝗶𝘀𝗸) I build 𝘃𝗼𝗶𝗰𝗲 𝗮𝗴𝗲𝗻𝘁𝘀 and 𝗹𝗼𝘄-𝗹𝗮𝘁𝗲𝗻𝗰𝘆 𝗮𝘂𝗱𝗶𝗼/𝘃𝗶𝗱𝗲𝗼 𝘀𝘆𝘀𝘁𝗲𝗺𝘀 that actually hold up on real calls, in production, under load. If you're fighting latency, 𝗰𝗮𝗹𝗹 𝗱𝗿𝗼𝗽𝘀, 𝗡𝗔𝗧/𝗧𝗨𝗥𝗡 𝗶𝘀𝘀𝘂𝗲𝘀, 𝗿𝗼𝗯𝗼𝘁𝗶𝗰 𝘁𝘂𝗿𝗻-𝘁𝗮𝗸𝗶𝗻𝗴, 𝗿𝗲𝗰𝗼𝗿𝗱𝗶𝗻𝗴 𝗳𝗮𝗶𝗹𝘂𝗿𝗲𝘀 or a 𝘃𝗼𝗶𝗰𝗲 𝗮𝗴𝗲𝗻𝘁 that breaks the moment a real customer interrupts it, I can take it end-to-end and ship something stable. I focus on what client's measure, response latency, barge-in handling, RTT, jitter, packet loss, concurrency and cost per call. 𝗧𝗲𝗰𝗵𝗻𝗶𝗰𝗮𝗹 𝘀𝘁𝗿𝗲𝗻𝗴𝘁𝗵𝘀 𝗶𝗻𝗰𝗹𝘂𝗱𝗲: Python, NumPy, SciPy, Pandas, Scikit-learn ML model development + evaluation Audio/signal processing pipelines Feature engineering + algorithm optimization Distributed computing concepts Real-time data handling & automation 𝗪𝗵𝗮𝘁 𝗜 𝗯𝗿𝗶𝗻𝗴 𝘁𝗼 𝗲𝗻𝗴𝗶𝗻𝗲𝗲𝗿𝗶𝗻𝗴 𝘁𝗲𝗮𝗺𝘀: Research-backed algorithm design Strong mathematical and analytical foundation Production-aware AI/ML integration System-level thinking (latency, throughput, scalability) Clear documentation and reproducible workflows 𝗖𝗼𝗿𝗲 𝗖𝗮𝗽𝗮𝗯𝗶𝗹𝗶𝘁𝗶𝗲𝘀: WebRTC / LiveKit Engineering SFU scaling strategies, TURN/STUN configuration, ICE tuning Optimizing jitter, packet loss, simulcast/SVC, bitrate adaptation Debugging join failures, audio/video inconsistencies, device issues 𝗠𝗲𝗱𝗶𝗮 𝗣𝗶𝗽𝗲𝗹𝗶𝗻𝗲𝘀: FFmpeg and OBS workflows for ingest → process → record → distribute Reliable multi-track recording and egress configurations Low-latency transcoding and optimized CPU/GPU usage 𝗜𝗻𝘁𝗲𝗿𝗮𝗰𝘁𝗶𝘃𝗲 & 𝟯𝗗 𝗦𝘁𝗿𝗲𝗮𝗺𝗶𝗻𝗴: Unity, Unreal and Omniverse real-time streaming setups Digital twin + avatar pipelines with synchronized audio/video Ultra-low-latency input → render → stream loops 𝗔𝗜/𝗠𝗟 & 𝗖𝗼𝗺𝗽𝘂𝘁𝗲𝗿 𝗩𝗶𝘀𝗶𝗼𝗻: Real-time OCR, text extraction, scene understanding Background removal, segmentation, enhancement Avatar/lip-sync triggers, real-time CV analytics ML inference integrated into WebRTC with strict latency control 𝗗𝗲𝘃𝗢𝗽𝘀 & 𝗢𝗯𝘀𝗲𝗿𝘃𝗮𝗯𝗶𝗹𝗶𝘁𝘆: Dockerized deployments, environment separation Monitoring (jitter, RTT, CPU/mem, packet loss, TURN usage) Alerting, runbooks, load testing, failure-mode analysis 𝗖𝗼𝗿𝗲 𝗖𝗮𝗽𝗮𝗯𝗶𝗹𝗶𝘁𝗶𝗲𝘀: WebRTC / LiveKit Engineering Media Pipelines Interactive & 3D Streaming AI/ML & Computer Vision DevOps & Observability 𝗠𝘆 𝗺𝗶𝘀𝘀𝗶𝗼𝗻: To create next-generation communication systems by merging AI/ML, signal processing and automation and to invent efficient algorithmic approaches that solve complex real-world challenges. Need a solid solution? Drop a text anytime