OpenSIPS Jobs

45 were found based on your criteria

  • Hourly – More than 6 months – 30+ hrs/week – Posted
    I am looking for full time FreeSWITCH / Adhearsion developer I require: - good skills with FreeSWITCH development - good skills with Adhearsion - good skills in FreeSWITCH administration It would be good to have skills in: - CoreOS - docker.io - Debian / Ubuntu administration - Git knowledge
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    I need someone that can setup an opensips box to load balance/distribute calls to several freeswitch boxes. The INVITE has to pass through the opensips bos over to the freeswitch with the original ip so freeswitch can determine the account it belongs to..
  • Hourly – Less than 1 month – 30+ hrs/week – Posted
    we need an asterisk expert to join our team to complete a project. we need to build conference line and voicmail for users on a2bnbilling. You should also know about call plan/routing and experienced in opensips.
  • Fixed-Price – Est. Budget: $200.00 Posted
    I need a custom OpenSips setup. Basicly the server must manage redundancy of DID's. we have 1 main server for voip. this server can send receive forward voip traffic. so inbound calls are forwarded to sub servers / pbx's. what I want OpenSips to do is to forward the calls and watch if the sub server is gone offline ( technical failure etc ) Opensips must send that call to second backup server. so main server : Server one sends calls to ...
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    Hello, Below is our environment: Asterisk version: 11.10 Environment: Amazon EC2 Operating System: Ubuntu 14.04 (64-Bit) Servers (Individual): Asterisk1 (Amazon EC2), Asterisk2 (Amazon EC2 - Backup Server), MySQL1 (Amazon RDS), MySQL2 (Amazon RDS - Backup Server), OpenSIPS (Amazon EC2) Note: All servers are deployed in Amazon VPC. Here, OpenSIPS server will be acted as SIP proxy server and Asterisk will be acted as Media server. Scenario: SIP users table, extension (call rules) table, voice mail tables will be created in ...
  • Hourly – Less than 1 month – Less than 10 hrs/week – Posted
    We are looking for an EXPERT Linux / VoIP Systems Architect who is able to perform an integration for us using either Kamailio or OpenSIPS. The objective of this installation is to gain high CPS for enterprise / call center VoIP Traffic. We were recently notified that the easiest integration would be using OpenSIPS and Freeswitch as our billing/routing engine has existing frameworks to integrate easily with FreeSwitch. More information about this integration is available here: http://docs.jerasoft.net/display ...
  • Hourly – More than 6 months – 10-30 hrs/week – Posted
    We are a company that just had our first switch installed We are looking for someone that has great knowledge of administering and maintaing FreeSwitch, LCR and ASTPP billing. We are wanting to add new modules as we grow and expand. You must be available during US hours and be able to work 4hr days 5 days a week. You must be willing to teach the system as well, as we are eager to learn. We are looking for someone ...
  • Fixed-Price – Est. Budget: $750.00 Posted
    Hello, We require the building of a VoIP Switch which focuses on inbound DIDs. It must handle 1000 concurrent calls. We have our own number range consisting of over 1000 000 DiDs It will have multiple trunks for over 10 providers who will send calls to this switch. The switch must accept the call and RECORD the IP/TRUNK/CHANNEL where the call is originating from. It must also route the call to the relevant SIP Account the DID is ...
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