SIP Jobs

13 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: $30.00 Posted
    Hello there, Basically, I need someone to compile this library for me: https://github.com/wbsoft/python-poppler-qt4 The library needs to be compiled using SIP (http://www.riverbankcomputing.com/software/sip/intro) and I have no idea how to do it. The setup.py is not working for me because the compiler needs to be configured and SIP needs to be installed. I need the library to work with the following dependency that I've already installed: - Windows 7 32 ...
  • 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: $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: $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.
  • 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 ...
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