SIP Jobs

25 were found based on your criteria

  • Fixed-Price – Est. Budget: $100.00 Posted
    I need a Perl script which takes a long lists of phone numbers and detects what phone numbers are valid. Script has CGI interface, allows to load CSV file, run calls and download result of calls. Calls made by SIP protocol using sip trunk. Source file has one phone number in each line. Result file has in each line: phone number;sip result code;Q.850 result code. Q.850 code is fetched from sip response message in header Reason ...
  • Fixed-Price – Est. Budget: $350.00 Posted
    We need to implement SRTP+directmedia support on Asterisk. Using SIP+RTP (encryption=no), with direcmedia=yes, the RTP goes direct between the two endpoints, without pass through Asterisk. But when we enable SRTP+TLS, the parameter directmedia is ignored and the reinvite message is not sent by Asterisk. We need that the media goes direct between the two endpoints, either when using TLS+SRTP.
  • Hourly – More than 6 months – 10-30 hrs/week – Posted
    Let us start with asking you to apply only if you have REAL experience in the field, including practical knowledge of all the subjects mentioned. Please write each section if you have the knowledge to deal with, and suggestions of what you would use: Your duty is: 1. decide about the best FREE software for auto-dialing of pre-programmed announcement, with interacting with the customer (please suggest which one is your personal favorite to use, when you apply, between Asterisk; vicidial ...
  • Fixed-Price – Est. Budget: $150.00 Posted
    i have server with staticip , want someone install and configure freeswitch on my server, and also a billing solution for Freeswitch , future paid support is expected from that guy, VNC user and password details will be provided to do everything remotely , VOIP Gateway will also be provided task will be completed when a test outgoing call is to be made from freeswitch and billing is to be done again i want exact alternate solution for VoIP Switch Freeswitch OR Asterik ...
  • Fixed-Price – Est. Budget: $20,000.00 Posted
    Instant Messaging, VoIP, and video conferencing based on SIP, XMPP, STUN, TURN, and ICE. Programmability and Integration: C and C++ APIs C and C++ sample application source code Android, iOS, Linux. Standards Compliance and Certification: Compliant with IETF standards SIP and XMPP Compliant with IETF, 3GPP, and CableLabs standards STUN, TURN, and ICE Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, SirenTM, Speex, and more Plug-and-play video codecs including EyeStreamTM, H.263, H.264, Background support for TCP ...
  • Fixed-Price – Est. Budget: $500.00 Posted
    We need an integration tool built. To access the VOIP SIP server and make an individual web page pop for the DID number that the call comes in on. Not the callers number. I can provide examples of a similar tool and provided login to the SIP server for testing http://www.gointegrator.com/ is a tool that connects to the SIP server and integrates. Works well but only pops a screen related the to callers number. We want it ...
  • 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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