Asterisk pbx işler

Su siparişi ip pbx santralden gelenleri otomatik cevaplouor tuslayana son aldığı urunden siapris giriyor. Müşteri bize bağlanmak ister ise yada yeni müşteri ise telefon cebime yönlendiriliyor. Sipariş geldiğinde istediğimiz kuryeye yonlendirme, ürün grupları ürünlerde fiyatlandirma. Gün sonunda hangi kurye kaç adet satmış nakit kaç adet kredi kartı kaç adet gibi özellikler. Admin tüm siparisleri görecek duzenleyecek. Yapılmış örnek var isteyene örnek gösterilir.

Google Dialogflow kaynağını kullanarak Asterisk veya Freeswitch ile çalışaçak Etkileşimli Sesli yanıt IVR oluşturma.

PBX telefon centrali uzmanı Genel Nitelikler - İleri Düzey PBX Sistemleri ve Voip Gatewayler hakkında bilgi sahibi. - İleri Düzey FusionPBX, Freeswitch, 3CX telefon centrallerinde uzman. - Sip telefonları provisioning yapabilen. - Tercihen linux scriptleri yazabilen. Atomasyon yapmaya meraklı. - Network bilgisine sahip (TCP/IP, LAN, WAN, VPN) - VoIP proje ve uygulamaları konusunda en az 3 yıl deneyimi olan - WAN uygulamaları ve cihazları, internet, firewall konularında deneyimli. - İngilizce bilen. Teknik döküman takip edebilecek düzeyde ingilizce bilgisine sahip - Evinden çalışabilecek. (Home ofis) - Çözüm odaklı olan, analitik düşünme yeteneği gelişmiş, sorumluluk bilinci yüksek, - Yoğun iş temposuna ayak u...

Asterisk santrali c# dili kullanarak ve kaynak kodlarıyla beraber bir yönetim programı oluşturmak istiyoruz. Daha önce asterisk santrali üzerinde çalışmalar yapmış, konuya hakim kişiler ö santrali c# dili ile yapılacak görseller projemize eklenmiştir. Buradan inceleyebilirsiniz. Bu tarz bir yönetim paneli talep ediyoruz. Bu projede bizlere yardımcı olacak ve proje ile ilgilenen kişileri teklifleriyle birlikte bekliyoruz!

Merhaba, 40.000 Kullanıcı kapasiteli, eş zamanlı 10.000 çağrıyı kaldırabilecek Asterisk kurulumu için destek verebilecek bu konuda IP Telephony, Call Center, Voip, Network konularında tecrübeli çalışma arkadaşları arıyoruz.

Elastiks veya İssabel pbx santral üzerinde bir kaç ayar yapılması gerekiyor. Asteriks bilgisi olan uzman arkadaşa ihtiyacım olacak.

Cihaz, kurulum ve bakım maliyetleri olmaksızın; gelişmiş bir santralin (PBX) sağladığı tüm özellikleri; bulut teknolojisi, “Bulut Santral” veya “Sanal Santral” olarak da bilinen ve VoIP ses hizmeti sağlanan İşteSantral ürünümüz ile sizlere sunuyoruz. Üstün özellikli sanal santral ses hizmeti çözümüne; uygun maliyetle ve aylık ödemelerle sahip olmak istiyorsanız, yeni lokasyonlarınız ile büyümeyi planlıyorsanız, şirket içi görüşmelerinizin ücretlendirilmeyeceği şekilde, bulut santral çözümümüzden faydalanabilirsiniz. Uygun Maliyet İlk yatırım maliyeti, santral bakım ve onarım masrafları olmadan kullanıcı başına aylık ücretlen...

Hello there I want to create a callcenter api with Fusionpbx I want to do with this api. 1- Create Callcenter. 2- Add and delete agents in Callcenter Queue. 3- Agent creation. 4- Agent delete . 5- Changing the status of the callcenter agent (Logged Out - On Break-Available) 6- Callcenter queue status.

Yusuf Bey Merhaba, STH lisanslı bir telekom firmasında teknik departman müdürüyüm , şöyle bir sorunumuz var daha önceden çalışmakta olduğumuz aculab prosody s üzerine kurulu voip sisteminden asterisk platformuna taşınmaya çalışıyoruz test amaçlı kurduğumuz server üzerinde denemeler yapıyoruz, şu anda çağrılarımızda bir sorun yok istediğimiz şekilde çalışıyor olmasına rağmen dtmf detection sorunumuz var, ne yaptıysak asteriskin dtmfleri yakalamasını bir türlü başaramadık, bu konuda bize yardımcı olma şansınız var mıdır?Teşekkürler.

Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.

Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.

Asterisk üzerinde çalışan crm yazılımını geliştirmek için proje desteği istiyoruz yapmış olduğumuz crm yazılımına data arama ve data kayıtlarının tutulduğu arayüz entegrasyonu yapılıcak.

Merhaba Benim bir voip softswitch im var. itelbilling kullanıyorum. Bazı hayal edip istediğim özellikler burada mevcut değil. mesela IP PBX , CALL CENTER , SESLİ ANKET , HOTEL , VOD AND DEAMOND , IP TV. Bütün bu sistemin çalışmasını tek ekranda yapabilecek var mı ? portaone gibi bir sistem istiyorum. detaylarını daha fazla anlatacağım

I’m ready to bring my Grandstream environment online and need a specialist who can jump straight into configuration. Both the PBX and the VoIP handsets are on-site, factory-reset, and reachable for remote access. Your task is to register every phone to the PBX, build the dial plan, and then load my custom call-waiting package—this includes a spoken message I’ve already recorded as well as a separate music-on-hold track. Key deliverables • Register and provision all Grandstream VoIP phones with the PBX • Configure extensions, inbound/outbound routes, voicemail, and caller ID as needed • Upload my provided audio files, assign the custom message to call-waiting, and set the music track for hold scenarios • Verify the announceme...

ONLY BUILD IF YOU HAVE STRONG EXPERIENCE IN THESE AREAS SIP, NAT Traversal, Asterisk/FS, React and strong backend servers scaling. I’m building a browser-based VoIP platform dedicated to business communication and I need an experienced React developer to take it from architecture to live deployment. The entire feature list—covering everything from secure voice and video calling to messaging and call-related utilities—is spelled out in the requirements document I’ve attached, so you’ll have clear, granular guidance from day one. SIP, NAT Traversal, Asterisk/FS Tech expectations You’ll craft a responsive single-page app in React (TypeScript preferred) that connects to a SIP/WebRTC back-end. I’m open to your preferred server stack ...

...routing communication through the unified system. 3. Technology Stack Layer Technology ERP Odoo 18 Enterprise Hosting Telephony Twilio Voice SMS Twilio SMS WhatsApp Twilio WhatsApp AI OpenAI Middleware NodeJS + Python Database PostgreSQL (Odoo default) 4. System Architecture Client Communication Channels Phone calls WhatsApp messages SMS ↓ Twilio Communications Layer Twilio Voice (PBX + IVR) Twilio SMS Twilio WhatsApp ↓ Middleware Integration Layer ↓ Odoo 18 Discuss (communications hub) CRM Recruitment Service workflows Projects Contact database 5. Twilio Configuration The system will use a Twilio EU phone number as the central communications endpoint. The existing public mobile number will forward incoming calls to this Twilio number. Requ...

Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File:...incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" including Asterisk/FS Skills: PHP, Software Architecture, Asterisk PBX, MySQL, JavaScript See more: making money with international premium rate numbers, iprn telecom, iprn providers, international premium rate numbers providers, international...

I already have a Linux server online and reachable; what I need now is a clean installation and full configuration of Asterisk, FreePBX, and A2Billing so I can run a reliable business phone system. Your scope covers: • Installing the latest stable releases of Asterisk, FreePBX, and A2Billing on my existing server • Bringing the stack to a “ready-to-use” state—SIP extensions, VoIP trunks, IVR, call recording, voicemail, and billing profiles must all function out of the box • Hardening the server (firewall rules, Fail2Ban, strong passwords/keys, SSL where applicable) • Running test calls to confirm inbound and outbound traffic, rate decks, CDR logging, and balance deductions in A2Billing • Handing over a concise post-deployme...

I want my new AWS account configured to host a USA-based SIP trunk through Amazon Chime. The job includes creating or hardening the AWS environment, provisioning an Amazon Chime Voice Connector, assigning U.S. numbers, and enabling both inbound and outbound calling. The trunk has to register cleanly with our existing on-premise PBX and soft-phone apps, so please allow for SIP signaling tweaks, codec selection, and basic security (TLS, SRTP, firewall rules). Once the link is live, I’ll need brief documentation of the key settings so we can maintain it internally. Deliverables • AWS account prepared with least-privilege IAM roles • Amazon Chime Voice Connector fully configured and licensed for U.S. PSTN access • SIP trunk integrated and test calls comple...

...Experience working with flash-based storage systems Linux system optimization for embedded hardware VoIP / SIP / Networking Experience implementing or maintaining SIP-based communications systems Knowledge of RTP / RTCP media streaming Familiarity with VoIP codecs such as G.711, G.729, Speex Understanding of NAT traversal, STUN, QoS (DSCP), and SIP registration Experience integrating with PBX systems (Asterisk, FreeSWITCH, etc.) TCP/IP networking, DHCP, DNS, NTP Audio Processing Experience working with real-time audio streaming Knowledge of ALSA or similar Linux audio frameworks Audio mixing, buffering, and jitter control Experience with microphone, speaker, and audio codec hardware Signal processing basics (tone generation, filtering) Embedded Hardware Integration Exp...

I need an IVR built and configured so callers reach the right help fast. Its sole purpose is customer support, and it must offer three options in every relevant branch: an automated response for common questions, the ability to transfer to a live agent, and—when queues ar...of the call flow with all customer touch-points • All audio files (or TTS scripts) in WAV and MP3 • Working configuration files or portal access so my team can maintain the menus • A brief test report confirming that automated responses, live-agent transfers, and call-back requests all succeed under real call conditions If you’ve set up similar customer-service IVRs before—especially on Asterisk, Twilio, FreePBX, or a comparable platform—let me know. I’m ready t...

...The system must greet callers, then automatically route each call to one of five departments through a simple, voice-prompt menu. In addition to this core routing, I also want the IVR to trigger bulk messages (SMS or WhatsApp blasts) and hook directly into Meta Business Suite so our phone flows, social inbox, and chat automations all stay in sync. You are free to choose the underlying platform—Asterisk, FreePBX, Twilio, or a comparable cloud service—so long as it supports reliable call routing, bulk-messaging APIs, and Meta’s official integrations. If a ready-made template speeds things up, that is fine; the voice prompts, language, and menu logic must still be customised to our brand. Deliverables • Fully functional IVR map with five departmental route...

...skonfiguruje połączenie z moim dostawcą SIP trunk – Zadarma – tak, aby system bezbłędnie realizował zarówno połączenia wychodzące, jak i odbierał przychodzące. Zakres prac: • dodać i zweryfikować trunk Zadarma, • utworzyć odpowiednie dial-plany i reguły routingu, • przeprowadzić testowe połączenia w celu potwierdzenia jakości oraz stabilności, • wprowadzić ewentualne korekty konfiguracji serwera Asterisk wchodzącego w skład GoAutoDial. Środowisko obejmuje jeden VPS, więc konfiguracja dotyczy pojedynczego serwera. Po zakończeniu proszę o krótką instrukcję krok-po-kroku, abym mógł samodzielnie zarządzać podstawowymi zmianami w przyszłości. Jeżeli masz doświadczenie z GoAutoDial i integracjami SIP trunk, daj znać, ile cza...

Troubleshoot asterisk trunk TLS/SRTP

Project Title: Full Issabel 5 PBX Deployment: Installation, Trunks & Extensions Project Overview I am seeking a VoIP specialist to perform a complete installation of Issabel 5 on an AlmaLinux 8 cloud instance. Beyond the base installation, the freelancer will configure the initial telephony architecture, including SIP trunks for external connectivity and internal extensions for users. Detailed Scope of Work 1. Server Installation & Hardening Perform a clean installation of Issabel 5 on AlmaLinux 8 using the official net-install script. Configure Fail2ban and firewall rules to block unauthorized SIP and SSH attempts. Set secure passwords for the Linux root, MariaDB, and Issabel web admin. 2. SIP Trunk Configuration Connect the PBX to my chosen VoIP provider usin...

Necesito conectar nuestro PMS Cloud con la centralita Asterisk para cubrir funcionalidades de gestión de hotel. El objetivo es que el personal pueda manejar desde el PMS: • Registro de huéspedes: que cada check-in o check-out actualice automáticamente el estado de la extensión telefónica asignada. • Asignación de habitaciones: que al cambiar una habitación en el PMS se reprograme la extensión correspondiente en Asterisk sin pasos manuales. • Facturación y pagos: que las llamadas salientes e internas se registren en la cuenta del huésped y se reflejen en la factura final. Ya contamos con un software específico de gestión hotelera en la nube; requiero que el integrador traba...

I need a complete on-hold production for our PBX: a warm, friendly female voice delivering the three short scripts below, separated by ambient background music beds of roughly thirty seconds each. The finished mix should run somewhere between one and two minutes in total and be supplied as a single 16-bit, 8 kHz WAV file ready to load straight into the system. Script to voice: “Thank you for calling Andrew’s Tyre and Mechanical North Lakes. All of our operators are busy assisting other customers and we will be with you shortly.” —30 s ambient music— “Did you know Andrew’s Tyre and Mechanical has been locally owned and operated for the past 30 years? Now that’s service!” —30 s ambient music— “Andrew’s Tyre...

I already have a Contabo server standing by and simply need a clean, production-ready ViciDial stack on it. The job covers: • Installing the latest stable ViciDial (with Asterisk and its dependencies) from scratch. • Optimising the underlying Linux distro you feel is most reliable for call-centre workloads. • Hardening the box with a well-tested firewall configuration—CSF, UFW, iptables, or a similar solution is fine—as long as only the ports Vicidial, SSH and web administration actually require remain open. • Verifying that the web interface, database, and telephony services all start automatically after a reboot and that calls can be placed through a demo campaign. • Supplying a concise hand-off sheet: all commands run, credentials cre...

My organization is looking for a telephony tech in the Saskatoon, SK to connect a Grandstream ATA to an Avaya IP Office system. The primary objective is to move two existing fax numbers—306-934-5787 and 306-955-3059—onto the ATA so they send and receive reliably through the PBX. The job covers: • Physically or remotely provisioning the Grandstream ATA, assigning it an internal extension, and ensuring it communicates correctly with Avaya IP Office. • Mapping both fax numbers to the ATA ports and confirming successful inbound and outbound fax transmission. • Providing a brief record of any IP Office or Grandstream settings you change so I can reference them later. Acceptance is complete when both fax lines pass test sends and receives without errors.

...interaction feels human. • Dynamic query-based routing – once the caller’s need is clear, the AI should transfer the call to the appropriate extension or external number automatically. • Clean hand-off – when the call is routed, the receiving party must get a short, accurate summary of the caller’s request so they can pick up seamlessly. I’m happy to integrate with existing VoIP platforms (Twilio, Asterisk, FreePBX, 3CX or similar) if that speeds development, but I’m also open to a custom SIP-compatible solution. Cloud-hosted, on-prem, or hybrid deployment can be discussed; reliability, low latency, and call quality are non-negotiable. For deliverables, I’ll need: 1. A working prototype handling live calls. 2. A simple...

I run a growing small business that relies on Microsoft 365 and a cloud-based VoIP phone system. I’m looking for a dependable partner who can step in as our day-to-day IT resource, keeping both environments running smoothly whi...keep us aligned. Deliverables • Same-day response to support tickets during business hours • Resolution of Microsoft 365 and VoIP incidents or escalations • User onboarding/offboarding completed within agreed timeframes • Monthly health report outlining work performed, open issues, and improvement ideas A solid grasp of Azure AD/Entra ID, Exchange Online, Teams admin, and common hosted PBX platforms will help you hit the ground running. If this sounds like your wheelhouse, let’s talk about how we can keep my tech ...

I have a WebRTC soft-phone built with JsSIP that needs to register to an Asterisk 18 server over WSS. SIP credentials are confirmed correct, yet the browser console shows an authentication failure. The signalling path is protected with TLS certificates, so the problem is somewhere in the certificate handling or the way Asterisk presents the challenge. Your job is to trace and eliminate the registration failure, then hand back a clean configuration and proof that the client can successfully register and place a test call. Environment details you will touch: – Asterisk 18 (pjsip stack enabled) – JsSIP running in a standard browser (wss://) – TLS with server and client certificates already issued Acceptance criteria: • JsSIP completes REGIS...

...configured CRM integration required with Zoho CRM The freelancer’s role is to properly configure campaign logic, IVR flow, DTMF capture, reporting, and CRM integration. 2️⃣ Existing Infrastructure Dialer: VICIdial (ViciBox 12, Asterisk 16.x) TTS Engine: Amazon Polly (Aditi voice configured) Audio: Pre-recorded WAV files ready GSM Gateway: Dinstar Server: On-prem Linux (Public IP available) CRM: Zoho CRM SSL: Not configured yet (freelancer may configure if required) 3️⃣ Scope of Work A. Audio & TTS Handling Freelancer must: Verify WAV format compatibility (8kHz, mono PCM for Asterisk) Upload & map audio properly in VICIdial Use Amazon Polly only where dynamic text is needed Implement fallback if TTS fails Optimize playback clarity B. Voice Blaster Campaign Set...

...guidance. This is a fast-turnaround project. Scope of Work: Label Updates You will be working from our existing AI file and implementing the following updates: Insert updated Supplement Facts Panel (we will provide content) Change wording: “Vegan Product” → “Vegan Friendly” Add “Equivalent to” before “1500 MG High Potency” Bold the following: Gummies quantity “Dietary Supplement” Add an asterisk (*) to the title: “Longevity & Vitality*” Enlarge the vertical logo for a stronger brand presence Replace subtle grape imagery with a more accurate visual direction (product is sourced from Japanese Knotweed, not grapes). Open to hearing/seeing a couple of suggestions, and I have a quick idea....

VoIP PBX, SIP trunk, Extensions setup with US, UK numbers

...that the third-party handoff is seamless. Required Qualifications: Proven experience with Alcatel OmniPCX Enterprise (OXE) or OmniPCX Office (OXO). Proficiency in using OmniVista 8770 or mgr/mtcl command-line interface. Deep understanding of Entities, Time Ranges, and Modification Tables within the Alcatel environment. Strong track record of handling root-level Linux/Unix permissions within PBX environments....

...demonstrate a sample conversation in Hindi. 2. Viewers can open a provided URL and hear the call with <3 s delay. 3. When I speak over the bot, it stops, acknowledges, and either answers or routes to the operator logic you deliver. 4. All conversation text and events appear in the dashboard and in a downloadable JSON log. Tech is flexible—Dialogflow CX, Rasa, Kaldi, Vosk, Twilio Voice, Asterisk, WebRTC, FFmpeg, OBS-style RTMP pipelines—use whatever delivers the smoothest Hindi recognition and low-latency stream, but keep licensing clear for commercial use. Tell me how you would architect the speech pipeline, manage interruptions, and keep the audio stream in sync. If you’ve built similar multilingual voice or streaming tools before, a quick demo link...

I am upgrading an Asterisk 20 installation to version 22 I compiled and installed the code ok, but when trying to run alembic to upgrade the database it fails with various errors. I need someone that understand alembic and Asterisk to fix the database so that alembic runs cleanly and future upgrades will work. The important thing is that the existing database does not lose any data as it is a live system.

...need to know up front: • Environment: fully cloud-hosted, no on-prem gear • Media: SRTP from Asterisk, RTP from 3CX • Symptom: recipient can hear, but caller cannot Your task is to trace the SRTP→RTP path, pinpoint why one leg is silent, and adjust the PJSIP, codec, or NAT/STUN settings so we get clean two-way audio. Once it works, I also need a short write-up of the changes so I can replicate them in staging. Acceptance criteria 1. Two-way audio confirmed on at least five consecutive outbound WhatsApp calls placed from 3CX 2. Updated PJSIP/SRTP configs returned to me (or pushed to my repo) 3. Quick walkthrough or screen-share explaining the fix Tools you’ll likely touch: Asterisk 20+, PJSIP, Wireshark, sngrep, 3CX console. SS...

...Agent transfer notification Recording link notification 5. Backend System Developer should build a backend service that will act as middleware between: Custom CRM Voice Bot Platform Ameyo Dialer Server Technical Requirements Strong Linux experience MySQL database experience Backend development experience REST API development Experience with telephony systems preferred Experience with Ameyo, Asterisk, or Vicidial preferred Preferred languages: Node.js Python PHP (Laravel) Important Notes Dialing will remain in Ameyo We are building our own CRM and backend UI developer is already working on frontend SSH access to server will be provided This project requires someone comfortable working with telephony systems and Linux servers. Future work available for full dialer ...

I am looking for an experienced VoIP/PBX specialist to set up a small FreePBX-based phone system. This is a fixed-price project, and I am requesting your best offer. The system should be hosted on a reliable and cost-efficient VPS provider of your recommendation. Hosting costs and Twilio costs are not included in your project fee and will be paid separately by me. You will guide me through the account setup where necessary. Project Requirements: The PBX system should include two to three dedicated inbound phone numbers (DIDs), connected via Twilio or a comparable cost-efficient SIP provider with good call quality. Proper SIP trunk configuration and basic VoIP security (firewall configuration, fail2ban, protection against toll fraud) are required. For the first phone numbe...

...handle process optimization, team coordination, and workflow setup. Key responsibilities & skills required: Manage BPO operations: process mapping (as-is & to-be), SOP creation/revision, bottleneck identification & resolution. Team handling: coordinate small-mid size teams, performance monitoring, training & knowledge transfer. Tools & knowledge: CRM/ERP basics, call center software (e.g., Asterisk, Avaya, or similar), reporting & KPI tracking (Excel, Google Sheets, Power BI preferred). Customer support/operations experience: handling inbound/outbound processes, quality assurance, compliance & escalation management. Strong communication, problem-solving, and organizational skills. Experience: 2-4 years in BPO/call center/operations management (IT...

Urgent need for an experienced Cybersecurity specialist for a confidential, short-term private project (1-3 months) in my startup. Key skills: VoIP setup and security (PBX, SIP, encryption, threat protection), VPN configuration and testing, spoofing techniques (for ethical testing/research), spoofed numbers handling/detection. Expertise in data collection from various sources (ethical/OSINT methods), advanced virus/malware detection, analysis, and simulation/creation for defensive/red team purposes (ethical security testing only – antivirus evasion, threat emulation). Experience: 2-4 years in cybersecurity (penetration testing, hardening, tools like Wireshark, Metasploit, ethical hacking). Work mode: Remote/office hybrid possible (Jaipur office visits if required, expenses...

Looking for my professor. I do have some basic knowledge of PBX over the years. I need a teacher to help me learn few things about FreePbx. I think I am pretty good at Telnyx side but Pbx side is going ok, but your expertise will make it better.

I am looking for a senior Vicidial / Asterisk Expert to perform a clean installation and professional optimization of a Vicidial system on a single server. Project Requirements: • Installation: Clean install of ViciBox v.11 (All-in-One) on a dedicated/VDS server. • Capacity: The system must be optimized to handle at least 100 concurrent calls on a single server without voice degradation or database lag. • Campaign Type: Configuration of a Survey (Press-1) Campaign. • Flow: Dialer calls the list -> Plays a greeting (IVR) -> If the user presses "1", transfer the call to a specific In-Group (Queue) where live agents are waiting. • CallerID Management: Proper configuration of Outbound CallerID to ensure CID is displayed correctly to customers. &b...

I need a production-ready softphone for both iOS and Android built on both WebRTC and standard SIP. The app will authenticate users with a simple username-and-password flow against our existing PBX or have an onboarding process for new customers, then expose a clean, corporate-style interface that matches the rest of our product line. You must be able to provide examples of apps you've made in the past which utilise both SIP and WebRTC. This might consist of screenshots, code samples or demos of apps. Core scope • Local audio mixing for conferenced/merged calls - this must be done on the device (might require native code) and will likely be the most challenging part of the project as our server does not support mixing of audio. • Ad-hoc conference/merge, BLF,...

...hardware in place, so I’ll rely on your guidance. My preference is to run the system on a VPS; please specify the exact CPU, RAM, storage and bandwidth you consider safe for 10 simultaneous agents. If you feel a dedicated machine would offer clear advantages, outline those too and I’ll weigh the trade-offs. Core tasks • Fresh installation of the latest stable VICIdial release • Server tuning (Asterisk, MySQL, Apache, networking) for smooth outbound volume • Basic security hardening (firewall rules, fail2ban or equivalent) • SIP trunk integration and configuration, including carrier recommendations that support CLI override • End-to-end testing of dialling, recordings and reports Post-install support & timeline Let me know how...

I need a small 'n' next to an asterisk removed from 6-20 pages of a PDF document. The PDF was created in Atticus. They seem to become visible when printed on kdp. The tech has to be able to id these from the pdf doc and remove. Ideal skills and experience: - Proficiency in PDF editing - Familiarity with Atticus or similar software - Attention to detail - Ability to complete the task efficiently

I need a small 'n' next to an asterisk removed from 6-20 pages of a PDF document. The PDF was created in Atticus. They seem to become visible when printed on kdp. The tech has to be able to id these from the pdf doc and remove. Ideal skills and experience: - Proficiency in PDF editing - Familiarity with Atticus or similar software - Attention to detail - Ability to complete the task efficiently

We are building a structured AI-powered call routing system in South Africa. The system must: • Integrate with existing PBX systems via call forwarding or SIP • Use a South African virtual number • Route inbound calls through an AI voice receptionist • Identify query type • Provide structured information • Escalate security-related matters • Send SMS notifications when required • Log call analytics This is NOT a chatbot project. This is a voice AI + VoIP routing infrastructure project. Technical Requirements: Developer must have experience with: • SIP / VoIP integration • PBX systems (3CX, Yeastar, Telkom, etc.) • Twilio or similar telephony APIs • AI voice agent implementation • Call forwarding configurat...