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Freepbx Jobs

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Hourly - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
I have an Elastix setup with a T1 card my D-channel keeps going up and down. I need someone to look at it and solve this issue. More information will be provided during interview. This needs to be solved ASAP. Please apply if you have proven experience doing this type of setups.
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...
Hourly - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
I'm looking for a Custom module on FreePbx connected to sql db, that calls a list of teachers every day at certain time, and prompts hello teacher "NAME[according to number dialed]", please enter info on students that were absent today, how many students were missing today, if "0" say thanks for the info log the to DB and hang up. if teacher prompts any other digits, it should ask, please enter student id number that were missing and press # after every student id, System should log data on sql db and hang up and send report to principal once a day of missing students. I have an existing FreePbx Server, running on Centos 6
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 -...