Improve Asterisk Server Call Capacity -- 2

I have 6 years of experience in Opensips, kamailio and freeswitch application server. I can work on your opensips script part and will distribute load properly among your all asterisk server. I have completed many features in opensips like Register, load balancer, presence server, NAT handling, proper call distribution with call count limitation for asterisk server, Stir Shaken, fail2ban and many SBC related features. There is no challenge for me to handle Sip signaling between opensips and Asterisk, i have basic idea about asterisk so i can guide in that also but main work i will perform in opensips. I am available to discuss let me know if you are ready for communication. Regards, Om

Hi, i have read your project description, and willing to start working on it please open a chat so we can discuss the details. the budget can be fixed once i have your complete requirement. thanks Best Regards, Ghani.

With over a decade of expertise in the network and telecommunications sphere, I believe I'm the perfect candidate for your project. As a Network, Cybersecurity, VoIP, and System Engineer, I possess a comprehensive understanding of Asterisk and OpenSIPS—the technologies at the heart of this project. My experience designing and implementing network infrastructures using these systems includes optimizing call capacity, making me well-equipped to enhance your Asterisk server's throughput. In addition to my proficiency with Asterisk and Opensips,I have worked extensively with Network Administration, Security & Virtualization Technologies—honing my ability to architect scalable and redundant systems, like the one you're aiming for. This translates into my being able to ensure that the SIP proxy implementation not only bolsters concurrent call capacity but also handles high volumes of calls simultaneously without sacrificing performance. On top of my technical skills, I bring a commitment to best practices in all my projects. I will not just optimize your Asterisk server for call capacity but do it in such a way as to set you up for success in the long run. Choosing me would mean getting an efficient solution from a dedicated engineer who takes pride in delivering quality work that stands the test of time. Let's connect and explore how we can improve your telephony platform together!

Hey there! I'm Raj Malviya, a friendly freelance developer with an awesome team. I've checked out your Requirement for an asterisk developer to do some customizations and I'm super excited to Develop and Design a fantastic product for you! Whether it's either an Android app or iOS app, or an e-commerce platform, I've got the skills to make it shine. You won't just get a great Mobile app and website; I'll also be your buddy for the next 12 months, offering top-notch support. I specialize in Web Apps and have a bunch of cool projects with amazing design under my belt. Flutter for Android and iOS, PHP, and Laravel are my playground, and I've been at this for over 5 years Check out my past work in my portfolio here: https://www.freelancer.com/u/betaitsolution Thanks a bunch for thinking of me for your project. I'm all set to turn your ideas into something amazing in today's competitive world. Cheers, Raj Malviya

Dear Hiring manager, I am writing to express my interest in the VOIP/SIP Engineer position as advertised. With a solid background in VoIP and (SIP), coupled with hands-on experience in network infrastructure and telecommunications, I am eager to contribute to your team's success. In my previous role at Capgemini, I spearheaded numerous projects focused on optimizing VOIP/SIP systems for enhanced performance and reliability. My technical proficiency includes configuring and troubleshooting VOIP protocols, designing scalable network architectures, and implementing security measures to safeguard communications. Additionally, my collaborative nature has allowed me to effectively liaise with cross-functional teams to deliver solutions aligned with business objectives. I have worked with SipP protocol , asterisk server, abacus for voice feature testing and development for the Adtran client ONT's , I am willing to discuss about the role and the opportunity in detail and eager to help you in achieving the goal of this project. Thank you for considering my application. I am looking forward to the possibility of discussing how my background, skills, and enthusiasm can contribute to the continued success of this role. I am available to discuss further details and to begin working on your project at your earliest convenience and can be reached at +917358516953 Regards, Beula

I Have 8 years of experience in VOIP and asterisk/freeswitch/opensip and I can complete this projects in 3 days kindly Direct message me