I am building cloud PBX platform and need Kamailio behind it.
It should accept call invites (both registrations and calls) and forward them to cloud server with following requirements for registration and call-routing:
All accounts of single customer must be registered at single server. Calls should be routed to server, based on registration.
If that server became unavailable (unresponsive), all SIP registrations must be relocated to another server.
Sample:
Customer A has accounts sipA-1, sipA-2, desired servers Ast1, Ast3;
Customer B has accouns sipB-1, sipB-2, servers: Ast2, Ast3.
Sip registrations (and calls) of sipA-1 and sipA-2 first should be forwarded to Ast1, if filed, to Ast3. For customer B it is Ast2, then Ast3.
Other conditions:
– RTP proxy must be installed (same with Kamailio server) and calls should be routed through.
– All types of SIP protocols must be accepted: udp, tcp, tls, ws, wss;
– Please include separate routing rules for ws/wss, they may get exceptions in future, i.e. different servers priorities or exceptions regarding RTP proxy;
– How I can manually switch customer registrations and calls (can only update DB? Perform some action?)
Comments:
– Cloud PBX servers will be both Asterisk and Freeswitch;
– Configuration should be designed with routing rules(i.e. route[TOSOMEWHERE] { ... } ), and they should be clearly documented (commented)
Simple integration article:
[login to view URL]:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb, and I want only to add multi-server support.
Don't need to integrate Asterisk DB (I have freeswitch too) to Kamailio, it must be different databases, I will sync data.
Development workflow:
– I will provide pre-configured accounts (4-6 accounts) on different servers and desired (Debian, Ubuntu, Centos) server for kamailio with full root access.
– Servers with SIP accounts will get private IP addresses, Kamailio will get both private and public address.
– Will release milestone payment (50%) once will be able to test SIP and WebRTC call through Kamailio
Budget:
– 200$. It is one-day work
Hi,
We have 6 year + experience with Open source CTI tools, like Asterisk, Freeswitch etc.. And we are fully confident about your satisfaction in this project if you choose us.
Thanks for sharing your requirement in such details and so clearly. however I don't think that I can complete it within 1 day. I will take 3 days and will charge you USD 400 as total.
Please reply If you want to proceed.
Regards
It's seem to very intresting task. But at i'm busy with other same project. If you can wait i can work with you from 1 oktober. We may discuss technical details at now.
Hi,
I have very good understanding of VOIP and have developed applications on it. I can give you demo for my applications.
I have 7 yr experience in mobile app development and have worked on various applications.
Please check my profile and portfolio at https://www.freelancer.com/u/NZTSolution.html to get idea about my expertise and quality of work I provide.
Looking forward to hear from you.
Regards Tushar
Hello, my name is Robert Wilson with etollfree. We are standing by and ready to complete this bid for you as efficiently and quickly as possible. If you have noticed we have very little reputation as a freelancer bidder but we do have much reputation as a freelancer provider. We are trying to increase our bidder reputation, this is why we are willing to do this job as low as possible. We want to make sure that we fulfill all of your needs so we can receive a good reference from you.
George here based from Los Angeles, USA. I am expert in desired skills for this project and have done similar tasks already. Please get back to me, so I can show you some of the work I have done. I will not be asking for any upfront only pay me when you are satisfied with the progress.