SIP Jobs

73 were found based on your criteria

  • Hourly – Less than 1 month – Less than 10 hrs/week – Posted
    I have a client with an IP-PBX (trixbox CE). They have two locations. They used to be able to connect from one site to the other over the internet. They just changed their internet service provider. Now, the data goes out one gateway (10.2.5.254) and the voice goes out another gateway (10.2.5.252). The voice gateway does not allow internet traffic so now they cannot connect the site-to-site trunk. I am looking for an effective ...
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    Hi. I'm looking for someone to install FreeSwitch on Fedora, make it talk to my SIP Trunk provider (I have connection details) and make the FreeSwitch provide soft-modems using span_dsp for HylaFax to make T38 Fax calls. The Freelance should know FreeSwitch, ssh, screen, wireshark, routers & TCP/IP, SIP Protocol, Linux in general, etc. The job will be done over the internet using ssh after we agree on the details. The project is finished when Hylafax makes relaiable calls ...
  • Hourly – Less than 1 week – 10-30 hrs/week – Posted
    We are looking for an Asterisk expert to help us configure existing Asterisk/FreePBX to work with a SIP Redirect Server based on 302 messages. The Asterisk/FreePBX is already installed and working. SIP Redirect Server is already installed, configured and working. The job is to configure Asterisk/FreePBX to forward the calls based on the responses from SIP Redirect Server.
  • Fixed-Price – Est. Budget: $200.00 Posted
    We are a VoIP telecom provider and we would like to start offering fax solutions to our customers. We have built an Asterisk server (FreePBX) and have successfully implemented fax-to-email for receiving faxes and have had some success with T.38 pass-through however we need the ability to use an ATA on a traditional machine and have the faxes transmit/receive reliably every time. The best solution we can come up with is a relay (store and forward) server that ...
  • Hourly – 1 to 3 months – Less than 10 hrs/week – Posted
    We have a softphone that compliments a chrome application. Lately we have been having networking / routing issues connecting to some of the bigger US based SIP providers (8x8, RingCentral). When we use the native softphone applications that most of these vendors provide (eg 8x8, RingCentral), the connection works fine. Thus, it is less of a networking / routing problem from the users router and more of an internal networking issue in the way the softphone is registering. We are looking for ...
  • Hourly – Less than 1 month – 10-30 hrs/week – Posted
    Our Small business needs a Zyxel USG20 security appliance configured for use. We will need a LAN configured for workstations, a LAN for an Asterisk server, and a DMZ/LAN for an Apache server. We will also need L2TP VPN accounts setup for multiple users to access all resources. We need a freelancer with experience to configure our firewall to support our Asterisk server connecting to our WAN side SIP trunk, connect to softphones/IP phones on the workstation LAN ...
  • Hourly – Less than 1 week – 30+ hrs/week – Posted
    These are APIs i need API to: 1. Create new account (sign-up) 2. Create a call conference room 3. Delete a conference room 4. Download voice mail 5. Get balance info 6. Get rates (for all countries) 7. Get plan (unlimited minutes, ...) 8. Deactive acccount 9. API to make conferene callback (I have python source code, I can send if need) Group chat problem/solution: Problem: 1. Sip server supports voice conferene but not group chat 2. XMPP (ejabberd) support ...
  • Hourly – Less than 1 week – Less than 10 hrs/week – Posted
    Google Voice announced they wouldn't work with ObiHai VOIP adapters after May 15, 2014. I have multiple ObiHai 202 adapters that ring at multiple locations. I've subscribed to PhonePower.com as the service provider as a test but can't figure out how to program the adapters. I also want to know if I should buy a inhouse VOIP PBX such as the "Yeastar YST-U100 MyPBX IP PBX" to allow more than two analog phones at the main ...
  • 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.
  • Hourly – Less than 1 month – 10-30 hrs/week – Posted
    The intention is to develop a web based application and a server based application: The objective is to develop a SIP ALG Detector client and server. The client must be a web based App, it should send REGISTER or INVITE messages to the server using the specified UDP ports, the server will report back to the client providing all the details for the messages received, finally the client will compare the original messages sent, against the ones reported by back ...
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