a2billing/asterisk/centos 5.5
$30-250 USD
Dibayar saat pengiriman
we had an live server with centos 5.5/asterisk/a2billing/openvpn it was up and running fine but suddenly got a problem and now i am unable to make more the 2 calls when i am sending calls upto 30. getting an message 603 declined in sip debug. i need an troubleshooter who can give me an solution instantly in not more that 4 hours if you are interested please
ID Proyek: #6052099
Tentang proyek
15 freelancer rata-rata menawar $225 untuk pekerjaan ini
Hi there,we specialise in asterisk servers. please see my past reviews, about Linux and Asterisk Solutions. lets complete this project
Worked with numerous asterisk projects. I can troubleshoot the issues and fix them. Lets discuss and finish this project.
Hello, I've plenty experience with Asterisk and VoIP in general. I do on a daily basis troubleshoot and debug issues with different technologies. I can diagnose the issue in a short time, and if a solution is available Lebih banyak
Hello. There are few possibilities why is that. If you give me the access I can look at it and only if I find the problem and know how to fix it, you can award the project. As a part of the deal I will provide you with Lebih banyak
Greetings I am interested in troubleshooting your issue. I am a developer with 10+years experience and extensive experience with wordpress magento and more. I can achieve your task in a timely and efficient manner. Lebih banyak
I have extensive experience that meets the qualifications, and would be happy to complete this work for you. We can speak via telephone ( 2569192001), skype - cesurasean, yahoo im, aim, msn messenger, or irc chat. T Lebih banyak
Hi there, I've been working with Asterisk for years. Ready to help you with your issue. Thank you.
12 years of asterisk and sip protocol experience. I also owned an a2billing server for more than 3 years and know very well how to deal with tose problems. Promise to fix in a couple of hours. Thank you
I am working in a live asterisk environment which uses various methods to make Automated voice calls(via Analog ,Digital PRI,IAX2,SIP) .Hence it ll be easier for me to troubleshoot this problem.
