SIP Jobs

6 were found based on your criteria

  • Fixed-Price – Est. Budget: $100.00 Posted
    I am running a SIP service using Kamailo SIP Server. I am using RTP Proxy. I want to record the audio of all conversation between two parties. You can use any method but I need all audio recorded and stored in database. We have enabled RTP proxy recording. 1. Need to check whether the recorded file name has caller/callee username 2. Need to convert RTP files to Mp3/Wav files and save in a server in the folder. Please ...
  • Fixed-Price – Est. Budget: $15.00 Posted
    Hi, I have an obi device I would like to configure. I am enlisted to dialnow.com The problem neede to solve - default of dialnow is to dial the whole number (+country-code area-code number) only. I want to dial as only (area-code number) so that by default the country code would be added by the sip device. In addition I want to preserve international calling so that if I add "+" sign before the number (or any other constant) the former ...
  • Hourly – Less than 1 month – 10-30 hrs/week – Posted
    We are looking for an EXPERT Linux / VoIP Systems Architect who is able to perform an integration for us using OpenSIPS into our billing system Jerasoft VCS. The objective of this installation is to gain high CPS for enterprise / call center VoIP Traffic. We have researched and found that the easiest integration would be using common radius collectors for OpenSIPS. More information about the common radius collectors is available at: http://support.jerasoft.net/index.php?/Knowledgebase/Article/View/16 ...
  • Fixed-Price – Est. Budget: $100.00 Posted
    I am running a SIP service using Kamailo SIP Server. I am using RTP Proxy. I want to record the audio of all conversation. You can use any method but I need all audio recorded and stored in database. We have enabled RTP proxy recording. Need to convert to Mp3/Wav files and save in a server in the folder. Please quote. My Server is CentOS and I can provide browser SSH.
  • Fixed-Price – Est. Budget: $130.00 Posted
    Hello, I have a running PIAF server in production. Asterisk is managed by FreePBX and the user base is managed by a2Billing. Everything is working like a charm. However, I would like to have web module to make calls through your browser. For that I tried to use sipml5 but with no success. I managed to establish a connection, but whenever I try to make a call, I get this warning in my asterisk console: WARNING[21995][C-00000fc7]: chan_sip.c ...
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