SIP Jobs

4 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 ...
  • 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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