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Freepbx Jobs

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Fixed-Price - Est. Budget: $ 100 Posted
Tehnology we use: -Asterisk -FreePBX -WebRTC (Tried with Sip5ML, JsSip, Sip.js) - 4 different trunks - One stationary number (our ISP) - 4 GSM numbers Problem we have: When calling trough stationary trunk on stationary trunk we get sound delay when answering late (Delay is not always the same it depends on the time when user pick up the call) Explanation: Ok so we start off with what is working. If we make a phone call through GSM trunks on any number (GSM or Stationary) everything works fine. The important thing here is that when connection is established and call starts to progress we get response 183 (With early media) and so on... The 183 response is sent by our GSM VoIP terminal. The problems appear when calling through our ISP trunk, since provider can't fake 183 response if it is not sent to them from dialed number. When we call through ISP trunk on GSM numbers things work fine, when connection is established GSM providers send 183 response with early media...
Fixed-Price - Est. Budget: $ 100 Posted
i am using asterisk 1.8. recently found a problem. i have 9 trunks. every trunk have 16 gsm port. my carrier send his call using "77761XXXXX" prefix. i need to distribute all calls equally using 1 (first) priority. So i need to distribute all calls like round-robin and equally. Thanks.
Fixed-Price - Est. Budget: $ 400 Posted
Hello, We need you to build a new a2billing server to handle our inbound and outbound call routing and billing. DO NOT APPLY UNLESS YOU HAVE USED A2BILLING, ASTERISK AND FREEPBX IN THE PAST AND CAN PROVE THIS We will provide a centos 6.5 server (hosted on AWS), and you will need to perform the following tasks; - Install asterisk 12 and freepbx 12 (chansip not pjsip) - Install and configure trunks as per our requirements (both inbound and outbound) - Install and a2billing 2.1 (latest version) - configure a2billing to our requirements - perform appropriate testing - be available for support once server is made live We will provide you a document detailing our requirements. some of the requirements for a2billing are - All accounts are post paid, no prepaid accounts - The database is to be backed up daily to a drive mounted on the server (we will set this up for you) - Caller ID's must be passed through, without alteration - A billing report must be emailed each month -...
Fixed-Price - Est. Budget: $ 10 Posted
Hello. Today I'm looking for a professional Asterisk/FreePBX programmer, that knows about DialPlan writing and maybe other interfaces of Asterisk. I need a simple application that allows users to log-in the voice menu, and be able to record a message, or to scroll (using the touch-tone phones) through messages others were left for the community. It's like a primitive BBS or Internet forum, but for use by people over the phone. It seems to me it can be achieved by dialplan programming. but if an external application server is better - I think it will be good for me also.