Asterisk Server Call Quality Improvement -- 2

With over a decade of experience in Network and VoIP administration, I'm well equipped to handle the challenges of your Asterisk server issues. My strong understanding of network configurations, coupled with my proficiency in the Asterisk server and VoIP technologies and my knowledge of voice codecs positions me as an ideal fit for your project. I have a track record of diligently identifying and resolving technical issues, ensuring optimal function and performance. My experience ranges from small to enterprise-sized companies, thus I understand the unique dynamics of different infrastructures. Additionally, I bring significant expertise in routing, switching, VPNs, FreePBX, 3CX, Vodia, Asterisk+Cisco CUCM+CME systems - giving me comprehensive insight into network structures. As a dedicated professional, I prioritize best practices and proactive problem-solving. I'm fully capable of investigating the root cause of your call quality issues, proposing effective solutions, and implementing necessary changes such as selecting an alternative codec if deemed necessary. Choose me for this project and let's ensure crystal-clear call quality on your Asterisk server deployed on your Wide Area Network.

Hi, I hope you are doing well. Myself Jibran and i read your project details. I'm working and using asterisk since 2013. I can quickly resolve your this issue with my expertise and experience. Let's connect to discuss. Thanks

With my extensive background in Linux, Network Administration, and VoIP technologies, I am confident that I can quickly investigate and identify the root cause of your Asterisk server's poor call quality. I understand the complexity of WAN configurations and have successfully solved numerous issues similar to yours in the past. One skill that sets me apart is my deep knowledge of different voice codecs and their impact on call quality. This understanding will prove invaluable as we navigate through potential solutions for your project. I have experience not only improving call quality on WANs, but also selecting the most suitable codec for each unique situation. In conclusion, my name is Noor, and I offer you a combination of technical proficiency and a knack for problem-solving that aligns perfectly with what you need. By choosing me for this project, you're making a smart and informed decision to tackle your Asterisk server's call quality challenges head-on.

Hello, I’m Raj Malviya, a friendly freelance developer with an awesome team. I read your requirement for Asterisk server call quality improvement and I’m super excited to work with you, With over 7 years of experience in Asterisk PBX, VoIP, and network configurations, I can help you achieve efficient communication solutions. My services include: * Asterisk PBX installation and configuration * Toll-free number management and call forwarding * Network security and optimization * Troubleshooting and ongoing support I can ensure your communication infrastructure is reliable, cost-effective, and scalable for your growing needs. Check out my past work in my portfolio here: https://www.freelancer.com/u/rajmalviyamalviy 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. Regards, Raj Malviya

Hello Sir/madam I am experienced in the field of VOIP telecom technology and managing Astrisk system either using commandline interface and configuration files or using web user interface like FreePBX and can help you with your issue described above. According to your description I'm not surprised that you are facing bad quality voice for your wan deployed system using G.711 codec. I'm not saying the codec is bad its one of the most clear audio codec in asterisk system BUT it has big bit rate compared to other audio codecs. Another issue need to be clarified is the type of wan service you are using are you using internet or another type of wan services Hope we can discuss your issue soon Feel free to contact me if you are interested. Thanks

Hi My name is Nima and I am working as Linux and VoiP administrator for more than 20 years, so from Asterisk 2, I have a projects in VoIP, and now I can help you on your problem, so if you let me to help you, it makes me so happy, because I need to run some projects here to make my profile good, I am new in freelancer and I want to make my profile good. So I am waiting for your support and let me to deliver the best to you. In your case I will debug your Asterisk to find the problem with RTP and SIP then try to capture the packet and see the problem, the sound problem in case of Jitter can be in case of Internet quality and also the bandwidth, quality of sound should be HD now a days which needs to run the best codec depend on the CPU and RAM, for exmaple if do not have a problem in resource then G729 will be the best, but our Asterisk should have this codec. With Best Regards. Nima.

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