Opensips Jobs

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Fixed-Price - Expert ($$$) - Est. Budget: $1,200 - Posted
The design goals are broadly as follows: * 99.9% availability * To be used in multi-tenant environment * End-users will have access to UCP * Currently require 10 concurrent calls, needs to be scalable to 100+ without total architecture redesign * Require one geographic PoP in Montreal hosted on OVH Public Cloud VPS * Seek to minimize monthly spend on VPS servers as much as possible while meeting design goals * Minimization of complexity to reduce ongoing management as much as possible as needs scale * Documentation of environment, configuration and technologies used The exact technologies used are up to you including database engine, use of OpenSIPS or not, etc. - we are open to all architecture suggestions.
Skills: Freeswitch VOIP Administration VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $20 - Posted
I give priority to people who can get this done quickly. Please use the word 'chaos' in your application so I know you've read this. Problem: I live in Thailand. I need to call Australian mobile and landline numbers. I need an Australian Caller ID. I need Australians to be able to call me on this same local Australian number. Looking for the best quality. Someone referred Did Logic but I'm open to other alternatives. How do you propose to go about this? Are there any other programs you recommend? I'm looking to get this job done as soon as possible. Please propose to me how long it will take and how much it will cost. The cost I have placed on this job is not real - I am open to offers. Thank you, Jordan Hogan
Skills: OpenSIPS Nexmo Plivo SIP
Hourly - Expert ($$$) - Est. Time: Less than 1 week, 30+ hrs/week - Posted
We are seeking the highest level VOIP, SIP, WebRTC, Asterisk Expert. Someone who has experience in VoIP Security and providing related security consultation service. Only the best please apply. Price is no issue if you are the best. We have an immediate voice quality problem with our Asterisk PHP cloud-based software. Our current problem is as follows: Based in Tokyo Japan. We connect to KDDI via our voip service through our data center. 1) We have been running our software for several years and generally the voice quality is acceptable although not perfect. 2) However we recently made two (2) installations and those same (2) installations have proven to be a disaster. The both have the same exact problem. 1) During at approximately the same time of day, various customers talking to customer sales agents cannot hear the agent. The result is the customer hangs up. 2) We have had various incidents every single day and cannot understand the reason. 3) No packet losses are being seen here. 4) Recently tested a codec called OPUS which helped a little but still causes repeated problems which are slightly different but in less frequency meaning the root problem is still there. 5) With OPUS we now have agent or customer or both not being able to hear each other. Again with less frequency but it is still a major issue. We need a high level problem solver and expert to help us identify the root cause and solve this issue quickly. We need you to start today until the job is accomplished. To apply, please write this in your application so I can ensure I do not have to read auto-response applications. ================================= Mr. Burns, I am highly qualified to help you fix your VOIP problem. I am available to start immediately. I am willing to take a paid test to verify my qualifications. Once you apply, my staff members Karen and or Cheza will immediately respond to you and get you started. Thank you.
Skills: OpenSIPS Asterisk SIP VOIP Administration
Fixed-Price - Expert ($$$) - Est. Budget: $5 - Posted
Hello. I would like someone to install and set up OpenSIPS on a test server that I will make available over SSH and any required ports for OpenSIPS to work. ... I would like someone to install and set up OpenSIPS on a test server that I will make available over SSH and any required ports for OpenSIPS to work. I will provide all the inbound calls from a test SIP provider to this server. ... The phone system accepts calls on UDP port 5060. It's important that the OpenSIPS demo server support multiple concurrent calls.