I need a VOIP Server for maximum of 1500 simultaneous calls between Android clients and a PSTN network. I installed an Opensips 2.1 server on Ubuntu Server 14.04 but have these problems:
1. Opensips Clients can call each other but without audio transition. It seems that this is a NAT traversal problem. I installed RTPProxy but could not solve the problem.
2. How can I configure this server with PSTN network?
3. How can implement a prepaid billing for my Opensips using ASTPP? I installed ASTPP but could not connect it to Opensips.
I want you to configure my server and learn me how can I do this myself.
The PSTN server is in Iran.
6 freelance font une offre moyenne de $194 pour ce travail
Hi There, I am a serious bidder. And I can start to work with you right now. Once I accept your project, I am sure I can complete it and satisfy your desires. However, I need to know exactly details you want before Plus
I am 15 years experienced senior software developer. I have mostly worked on the VOIP projects. i have designed hybrid VOIP PBX. I have used Asterisk, i have written modules for it and configured. I used b2bua at the Plus
hi. 1. You need configure nat in opensip,cfg. 2. What do you have to connect to the PSTN? 3. Read manual. If you want I'll help you
I am Vlad Paiu, an official OpenSIPS developer and contributor. I have a vast experience in implementing and deploying OpenSIPS based platforms, in troubleshooting SIP, RTP and NAT issues and also knowledge of the inte Plus
hello, I been sysadmin cluster with two asterisks and one opensips balancer with rtp-proxy. So if you want simple load balancing that's not hard task I can do. For futher we may discuss about it, tell me your spec Plus