VOIP Software Jobs

23 were found based on your criteria

  • Fixed-Price – Est. Budget: $30.00 Posted
    I need to modify the Asterisk dialplan/a2billing calling card system. Need to allow a customer to be able to dial a destination number right from the beginning of an established connection to the a2billing. At the moment, if customer dial destination number during the welcome message or balance announcing, first digits of the destination number get cut off.
  • Fixed-Price – Est. Budget: $35.00 Posted
    We have a few Polycom 331 IP phones that were used with a previous VoIP carrier. They have locked the phone where we cannot use the web interface to reprogram to another VoIP provider. We need to re-flash these phone back to factory default with web interface to program. I need all the necessary files to program the phones via TFTP. We need the correct bootrom.ld, sip.ld and the rest of the files to format and flash this ...
  • Fixed-Price – Est. Budget: $10.00 Posted
    We have GoAutoDial 3.3 installed on a dedicated server with BroadVoice Sip. Everything is working fine, but for some reason, each lead is being dialed for over 50 seconds. In each campaign, I have Dial Timeout: 20 (in seconds) I cannot figure out how to have it dial each lead for only 20 seconds. It should dial the lead, if no one answers within 20 seconds, hang up and dial the next lead. I've had my Asterisk tech ...
  • Fixed-Price – Est. Budget: $250.00 Posted
    Please Provide Cost and Timeline for the completion of The Proposed Project as listed below: We have 2 servers that are currently configured with Elastix 2.5. One of the servers is a cloud based server and the other is a local based server located in our facility. 1. Configure servers to provide redundancy (exact image) between both servers, so if one server fails we will have the ability to continue providing service to our customers. 2. Configure and test ...
  • Fixed-Price – Est. Budget: $3,500.00 Posted
    - App to make VoIP to VoIP calls - VoIP to local and international landline calls - Local and international landline calls to VoIP - Also PBX routing App to have user login, and comes with dashboard stating: - call destination number log - duration of calls log We currently have Asterix PBX infrastructure, hence requiring mobile development and web dashboard development to collect to the PBX hosted on VPS.
  • Fixed-Price – Est. Budget: $5,000.00 Posted
    We are looking for a company or a team to develop a IP PBX, VoIP soft switch for our newly created IM/VoIP application. You will be involved in building a server that functions as a SIP gateway to route calls and messages. The team must have excellent knowledge in VOIP technologies. Good candidates should have substantial knowledge in open source VOIP solution e.g. Asterisk, Kamailio. We can discuss further if you are interested.
  • Fixed-Price – Est. Budget: $300.00 Posted
    I am looking for someone to work with me long term on a part time bases. you need to be pretty advance is asterisk and kamailio. you will be working under another another team member who has built the telecom system. i am looking to pay $300 a month on a constant part time bases. if you think you know what you are doing then please get in contact, must speak English on skype, many thanks
  • Fixed-Price – Est. Budget: $400.00 Posted
    We require Kamailio and Freeswitch Real Time integration. Implementation and Architecture should be well documented for us to manage the system. Architecture should include 1. N + 1 instance of Kamailio - Responsible for users registrations and user to user audio/video calls 2. N + 1 instance of Freeswitch server - responsible for voicemail, conferencing, pstn and media 3. Cgrates on Freeswitch for billing 4. Clustered MySQL DB for Kamailio/Freeswitch
  • Fixed-Price – Est. Budget: $1,500.00 Posted
    Activate a Unified System UC560 to upgrade an older IP phone system... The project will be defined in two phases due to a Building migration process and a planning of execution must be defined. Phase 1: _ The System UC560 will be feed with a T1 links+2 PSTN backups (already used only for internet), replacing the existing analog lines on a SIP gateway. _ 1 Conference phone, 8 terminals, a Central attendant and voice mail must be implemented. Phase ...
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