FreePbx, Asterisk y Vicidial -- 2

soy un dev full stack con mas de 10 años de experiencia en voip, con asterisk , issabel, php, linux. puedo ayudarte configurar tu freepbx

Hello, I am a specialist FULL-STACK developer with 8+ years of experience in website and mobile app development. I have worked on numerous projects with different clients all over the world to deliver quality-driven bug-free websites and mobile applications. Additionally, I offer services in many different areas including architectural designs, 2D/3D drawing/landscape/interior designs with multiple software tools. I understand that you are looking for someone to configure FreePBX for you so that you can take a SIP trunking scenario of 30 channels and pass them onto another FreePBX to an aggregation queue. Other three channels will be connected to an internal FreePBX while the remaining channels will be connected to a third FreePBX. I believe that my skillset makes me the perfect fit for this project as I have experience in architectural designs as well as 2D/3D drawing/landscape/interior designs with multiple software tools which will be required for making the configuration of your FreePBX network seamless. Additionally, I offer client support 24/7 along with regular communication so that issues can be resolved within 24 hours should they arise. With my one month..

Hi! I have read all the provided instructions, and I am the right person to work on this task, I wanted to reach out and see whether the opportunity is still available. As you can see from previous clients I deliver exceptional quality work, I’m incredibly passionate about your project, and my style compliments your requirements: System Admin Linux Asterisk PBX Cisco VoIP Click on the that__CHAT__button below so that I can ask a few questions concerning your project. If the job’s no longer available, I’d appreciate you throwing my hat into the ring for any similar opportunities in the future. Thank you for your time, I look forward to hearing from you soon, Best Wishes, Damaris

Greetings Dear Client. Welcome to my profile, Home to Professional and Quality services with 100% customer satisfaction guarantee. I'm a Certified & Experienced Expert in the respective project requirements. Dear Client, I take this opportunity to inform you that i have KEENLY gone through all your project requirements as given in your project description and I confirm to you that i can perfectly deliver as instructed. Being in possession of all CLEARLY STATED required project skills (Linux, Asterisk PBX, System Admin, VoIP and Cisco) as this is my area of professional specialisation having completed all Certifications and developed adequate experience in the same area, I hereby humbly request you to consider my bid for Professional, Quality and Affordable services always. STRICT TIMELY DELIVERY & UNLIMITED REVISIONS. Kindly Message Me We Discuss More About The Project and seal the contract. Welcome and Thank-you.

Hey I am vicidial expert > I can upload list in all campaign efficiently before dialing start > I can add numbers in dnc > I can create filter > I can create campaign > I know how to create users and phone > I can restart dialer when its need to be > I can create avatar or soundboard > I will also provide you recordings for avatar > I can remove recordings of months if we face disk full issue > I can also setup DID,s in dialer > I can set specific caller ids that you want to show to your customer

Para este escenario, lo ideal seria usar un SBC para distribuir el trafico a las 3 PBX de manera que sea muy facil hacer cambios en la mezcla cada vez que haga falta. Para poder ser concreto en la solucion a implementar seria ideal poder tener una conversacion de discusion tecnica inicial y decidir el mejor camino. Tambien seria bueno conocer a priori si se va a discriminar el destino (Cola, extensiones, etc...) en base al DID marcado. Por lo descrito en el requerimiento, pareciera que un FREEPBX esta siendo usado como distribuidor. Estoy a su orden si deciden trabajar conmigo.

Hola, tengo más de 13 años de experiencia trabajando en tosa clase de desarrollos para asterisk, puedo ayudarte en lo que necesitas, contactame para cerrar los detalles y poder comenzar a trabajar. SALUDOS

Entiendo que el sip trunk es para tráfico saliente? Y lo recibirá un único servidor sip, este sería quien finalmente limite la cantidad de canales simultáneos para los otros endpoints? Confirmar si las llamadas entrantes que se derivaran a distintas colas o pbx son DIDs diferenciados? O si mantendrá un único DID.