Alexander Merkulov

• Asterisk PBX Phone system

  [Sealed]
  5.0
  

  jjsmd

  Apr 21, 2013

  “ Meral was responsive and was more then willing to make changes and completed the job to my satisfaction.Meral was responsive and was more then willing to make changes and completed the job to my satisfaction. ”
  Project Description: I need an asterisk pbx ( on a standalone device) that would function in the following manner: 1) call comes in and PBX examines the CID 2) If CID is in contact database, it will announce to caller to...
  [more]

• Get AsteriskNOW to register with Broadvoice

  $91.3 USD
  5.0
  

  ernestwilliams

  Apr 18, 2013

  “ Excellent work! Happy with results.Excellent work! Happy with results. ”
  Project Description: I have an AsteriskNOW system up and running. I am unable to get the system to register with Broadvoice. I have used all the examples on the Broadvoice website without any luck. I would like someone to get the asteriskNOW configuration files to such a point that my system is recognized...
  [more]

• FreePbx module required

  £150 GBP
  5.0
  

  seanmattysl

  Apr 16, 2013

  “ TOP MAN, did all i asked and very very quick. Will always only use this man againTOP MAN, did all i asked and very very quick. Will always only use this man again ”
  Project Description: Hi, I need a module to be made to allow me to do the following: Place a call into music on hold and randomly terminate the call within a set amount of minutes. e.g. randomly terminate the call...
  [more]

• Custom Project - Sample Setup and Sample Report - repost

  $750 USD
  5.0
  

  JFoshee

  Mar 30, 2013

  “ Great to work with and got the job done. Will be using in the future when we are ready to add more.Great to work with and got the job done. Will be using in the future when we are ready to add more. ”
  Project Description: This project is to allow sample both our data and see what it would take to build the reports we need. We will provide mysql access to our data and then provide example site and reports that we need to have...
  [more]

• Configure Asterisk Server 1.8

  $250 USD
  4.8
  

  garbageyunk

  Mar 6, 2013

  “ I great person and so Professionalism! Thks so much! for your great job!I great person and so Professionalism! Thks so much! for your great job! ”
  Project Description: I have a project to configure Asterisk, are the basics of Asterisk so do not think you have problems. I can not do due to time If you are not asterisk guy please dont waste your time and my time. You must have experience with this...
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• Create a simple voicemail system on Asterisk/A2Billing

  $150 USD
  5.0
  

  cmic

  Mar 5, 2013

  “ Alexander is always great to work with. Hire him! You won't be disappointed. Always on time, always on budget.Alexander is always great to work with. Hire him! You won't be disappointed. Always on time, always on budget. ”
  Project Description: Simple voicemail system on Asterisk/A2Billing Version 1.4.2.1 (Radix) - Released : 02 October 2009 Works as follows: 1) Detects busy or no-answer calls 2) Records a voicemail file (low quality, mono,...
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• FreePBX Trixbox email script on reboot/reload

  $100 USD
  4.8
  

  oiltycoon

  Feb 27, 2013

  “ Did a great job on our little project. Made things work and was impressed.Did a great job on our little project. Made things work and was impressed. ”
  Project Description: FreePBX script change Basically I would like the system to email me anytime somebody makes a change that requires a reload, or the server is rebooted with the IP address of the server that was reloaded or rebooted(Either GUI or command line reload/reboot)...
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• OpenSIPs Load Balancing utilising multiple Asterisk Instances all built with AWS EC2 servers

  $1500 USD
  4.2
  

  jacko67

  Feb 10, 2013

  “ Extremely happy with the work done and the effort put in. Despite language barrier, requirements and tasks were understood with a little extra effort and more importantly delivered. Flexibility, Patience & Professionalism second to none. Will hire again for similar work.Extremely happy with the work done and the effort put in. Despite language barrier, requirements and tasks were understood with a little extra effort and more importantly delivered. Flexibility, Patience & Professionalism second to none. Will hire again for similar work. ”
  Project Description: We're looking for an expert to help prove our proposed design by carrying out a technical proof-of-concept (PoC). Essentially, we want to see multiple Asterisk instances running on individual EC2 server...
  [more]

• VOIP Swich

  $250 USD
  5.0
  

  Rdline

  Feb 5, 2013

  “ No complaints. Alexander is a professional, who is going deep and thinking about your whole solution, not only about particular task from project description. Recommendations.No complaints. Alexander is a professional, who is going deep and thinking about your whole solution, not only about particular task from project description. Recommendations. ”
  Project Description: Dear friends, I am looking for a simple SoftSwitch without billing working on Windows or Linux platforms. This Softswitch has to manage trunk accounts between Soft/Hard IP phones and GSM Voip Gateway http://www.hybertone.com/uploadfile/download/20101125010256336.pdf ...
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• Fix A2B Callback Daemon 3

  $88 USD
  4.8
  

  cmic

  Jan 27, 2013

  “ I wish there were more developers like Alexander: very skilled, fast, accurate!I wish there were more developers like Alexander: very skilled, fast, accurate! ”
  Project Description: The stupid website ignores periods, that's why it was $882. Terrible. I'm trying to fix it...costs me money. Terrible website.
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Alexander Merkulov has not completed any projects.

• Asterisk FreePbx setup, config, firewall and support

  $50 USD/hr In Progress

  Our company looks for general support our Freepbx Asterisk. We need to do some configuration changes, integrate new parts, reconfigure trunks, review the Firewall setup (cisco) and extend some functionality. We look for a freelancer that supports us over time. We look forward hearing from you.Our company looks for general support our Freepbx Asterisk. We need to do some configuration changes, integrate new parts, reconfigure trunks, review the Firewall setup (cisco) and extend some functionality. We look for a freelancer that supports us over time. We look forward hearing from you.

  [more]

• 386737 Voip Software

  $230 USD In Progress

  Voip Software Needed more details on pmb...Voip Software Needed more details on pmb...

• Desktop Application / Asterisk CID Lookup

  $1500 USD In Progress

  We require a Desktop Application to interface with our Asterisk Phone System and Client Database (ConnectWise / SQL). This will be an ongoing development project with the successful applicant to further develop additional features.The immediate requirement is to capture incoming Caller ID and reference our Client Database (ConnectWise / SQL) and then pop-up a screen with the client/company details.We require a Desktop Application to interface with our Asterisk Phone System and Client Database (ConnectWise / SQL). This will be an ongoing development project with the successful applicant to further develop additional features.The immediate requirement is to capture incoming Caller ID and reference our Client Database (ConnectWise / SQL) and then pop-up a screen with the client/company details.

  [more]

• Custom Project Jun 15 2012 10:14:18

  $50 USD/hr In Progress

  We are requiring a IVR credit card theatre ticket bookin system.We need to give the caller options for choosing the show they want to watch, the date of the show, the time of the show and what type of ticket they require(child adult etc) linked to a database that can be updated by the customer. Once this has been selected to then go on and take payment by credit card. It would need to be able to be used by multiple customers. Ability to import ivr recordings for each stage. We have a credit card script wich is tested with our credit card provider which we can supply the script for.Calls can arrive into the system either from one of our existing ivr's or direct. We are requiring a IVR credit card theatre ticket bookin system.We need to give the caller options for choosing the show they want to watch, the date of the show, the time of the show and what type of ticket they require(child adult etc) linked to a database that can be updated by the customer. Once this has been selected to then go on and take payment by credit card. It would need to be able to be used by multiple customers. Ability to import ivr recordings for each stage. We have a credit card script wich is tested with our credit card provider which we can supply the script for.Calls can arrive into the system either from one of our existing ivr's or direct.

  [more]

• Music on Hold Server

  $200 USD In Progress

  We would like to have a server so when we place a call on hold the call to gets the music from the server.We are currently getting the music on hold from our platform but the licensing is very expensive therefore we would like to use the alternative with our own server.You will need to examine how the current call flow with the music on hold to our platform works and replicate the same thing to the new music on hold server. We are currently using Polycom phones and they have an option to direct the music on hold to a different server the following is an example reg.1.musicOnHold.uri='sip::5120'When bidding on this project please include your experience in SIP, similar projects you had done and how fast you could complete this project.We would like to have a server so when we place a call on hold the call to gets the music from the server.We are currently getting the music on hold from our platform but the licensing is very expensive therefore we would like to use the alternative with our own server.You will need to examine how the current call flow with the music on hold to our platform works and replicate the same thing to the new music on hold server. We are currently using Polycom phones and they have an option to direct the music on hold to a different server the following is an example reg.1.musicOnHold.uri='sip::5120'When bidding on this project please include your experience in SIP, similar projects you had done and how fast you could complete this project.

  [more]

• Asterisk System for conference / Fax to email / email to fax / IVR / DISA Modules

  $824 USD In Progress

  Looking for experienced Asterisk developers only!Background:This asterisk based application requires NO devices to register, this is for virtual services only. This is very similar to a grasshopper solution.As scope exceeds character limit for freelancer, please see attached file for full scope. Please ready entire scope before bidding.Deliverable- Application is to be programmed on developers system- application needs to pass testing controls on all features. developer must display all features working- application installation instructions for cent os 6 and asterisk version selected by developer.Looking for experienced Asterisk developers only!Background:This asterisk based application requires NO devices to register, this is for virtual services only. This is very similar to a grasshopper solution.As scope exceeds character limit for freelancer, please see attached file for full scope. Please ready entire scope before bidding.Deliverable- Application is to be programmed on developers system- application needs to pass testing controls on all features. developer must display all features working- application installation instructions for cent os 6 and asterisk version selected by developer.

  [more]

• asteriks project - repost

  $100 USD Today

  hi I have a asterisk server and I have client on extension and i wan to give link for my client to view concurrent call and also i want to give them access to add or change their terminator IP. with web link, can you do this for me?here is my mail admin@dial2all.comhi I have a asterisk server and I have client on extension and i wan to give link for my client to view concurrent call and also i want to give them access to add or change their terminator IP. with web link, can you do this for me?here is my mail admin@dial2all.com

  [more]

• a2billing / php customize - repost

  $515 USD Today

  A2billing CustomerUI customiseWhen a Customer logs into their A2Billing Customer Portal Currently, their negative balance displays. Below the negative balance is a drop down to select the amount of money you want to add to your account. You select the amount and it redirects you to Paypal. Then redirects back adding the funded amount to your account. Instead of this we want when the Customer logs in to his portal he sees the negative balance and a link below to pay that negative balance amount, that says PAY BILL NOW. NOT a drop down to select an amount. This Link to Pay Bill Now should send this amount to our Authorize.net gateway and return marking the database paid.A2billing CustomerUI customiseWhen a Customer logs into their A2Billing Customer Portal Currently, their negative balance displays. Below the negative balance is a drop down to select the amount of money you want to add to your account. You select the amount and it redirects you to Paypal. Then redirects back adding the funded amount to your account. Instead of this we want when the Customer logs in to his portal he sees the negative balance and a link below to pay that negative balance amount, that says PAY BILL NOW. NOT a drop down to select an amount. This Link to Pay Bill Now should send this amount to our Authorize.net gateway and return marking the database paid.

  [more]

• Installing and configuring Kamailio May 19 2013 - repost

  $240 USD/hr Today

  We are interested in installing kamailio and configuring it as re-routing SIPWe are interested in installing kamailio and configuring it as re-routing SIP

• Centos OS installed, freepbx installed & sugar crm installed

  $412 USD Yesterday

  I need a server set up with centos OS Asterisk pbx and sugar crm. This should be a straight forward project. I will also need you to configure my rougher so that the ports are all configured and I will need a Cisco phone set up to work via the PBX.I need a server set up with centos OS Asterisk pbx and sugar crm. This should be a straight forward project. I will also need you to configure my rougher so that the ports are all configured and I will need a Cisco phone set up to work via the PBX.

  [more]

• Fix Asterisk Autodialer report

  [Sealed] Yesterday

  I have an instance A of an Autodialer -Asterisk server where campaign report is not longer working. I also have another instance B of the same Autodialer Asterisk server where report is properly working. The objective of this project is to fix report for instance A. You could refer to Instance B to see report working and to determinate how to make it work for Instance A again.I have an instance A of an Autodialer -Asterisk server where campaign report is not longer working. I also have another instance B of the same Autodialer Asterisk server where report is properly working. The objective of this project is to fix report for instance A. You could refer to Instance B to see report working and to determinate how to make it work for Instance A again.

  [more]

• Installing and configuring Kamailio May 19 2013

  $200 USD 2 days ago

  We are interested in installing kamailio and configuring it as re-routing SIPWe are interested in installing kamailio and configuring it as re-routing SIP

• Popup window

  $1546 USD 2 days ago

  We would like to make a popup window that interacts by a web service.We would like to make a popup pop when an incoming call is entering in the system.Popup requirement:1.Popup should be able to appear on the right / left side (RTL or LTR), and should be able to show English and Hebrew text.a.The text will be imported from a local (client side) txt file based on the language preferences of the client, as well as RTL and LTR.2.Popup should appear above the taskbar.a.Specific parameters should be imported from a local (client side) txt file based on the language preferences of the client.i.For example:1.Opacity2.Font3.Font size4.Background color5.Transition timeout6.location3.The popup will show the CID (who is calling) and his name (taken from the DB). a.Name should be UTF8, and should be RTL or LTR.4.On the popup will be a logo of our company that will link to our website. This should be hard coded.We need to have a configuration file that we can config these elements:1.Font type2.Font size3.Font color4.Popup sound5.Popup speed6.Popup size (width and height)7.Popup type (up and down / fade in fade out)8.Background color9.Exit buttonWe need to build a web service, that gets information from our server (https), and when it gets a string it needs to open the popup and show what is inside the string.In addition, we want to add a reject button for a phone call, forward a call to another number. Example:We would like to make a popup window that interacts by a web service.We would like to make a popup pop when an incoming call is entering in the system.Popup requirement:1.Popup should be able to appear on the right / left side (RTL or LTR), and should be able to show English and Hebrew text.a.The text will be imported from a local (client side) txt file based on the language preferences of the client, as well as RTL and LTR.2.Popup should appear above the taskbar.a.Specific parameters should be imported from a local (client side) txt file based on the language preferences of the client.i.For example:1.Opacity2.Font3.Font size4.Background color5.Transition timeout6.location3.The popup will show the CID (who is calling) and his name (taken from the DB). a.Name should be UTF8, and should be RTL or LTR.4.On the popup will be a logo of our company that will link to our website. This should be hard coded.We need to have a configuration file that we can config these elements:1.Font type2.Font size3.Font color4.Popup sound5.Popup speed6.Popup size (width and height)7.Popup type (up and down / fade in fade out)8.Background color9.Exit buttonWe need to build a web service, that gets information from our server (https), and when it gets a string it needs to open the popup and show what is inside the string.In addition, we want to add a reject button for a phone call, forward a call to another number. Example:

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• Custom Project May 18 2013 16:58:25 - repost

  $3092 USD 2 days ago

  I am looking for voice application developers to build an IVR/ chatline system using C and asterisk for voip integration.The system would consist of a voice personals section , a live chat section , and a multi user conference section.every new caller would get a 6 digit mail box number and password to use on each login. The system would have voice mail and live 1on1 connect features. There are lots more features and rules associated wit this program and will be explained to to best qualified person/s .If you are an expert at building IVR systems you would already have an idea of what is involved in this project. To be considered, you must have a portfolio of similar work to show me and you must be able to communicate 24/7.I am looking for voice application developers to build an IVR/ chatline system using C and asterisk for voip integration.The system would consist of a voice personals section , a live chat section , and a multi user conference section.every new caller would get a 6 digit mail box number and password to use on each login. The system would have voice mail and live 1on1 connect features. There are lots more features and rules associated wit this program and will be explained to to best qualified person/s .If you are an expert at building IVR systems you would already have an idea of what is involved in this project. To be considered, you must have a portfolio of similar work to show me and you must be able to communicate 24/7.

  [more]

• VoIP-PSTN Trial

  $5000 USD 4 days ago

  We want to build a very simple proof of concept to help us raise capital. The proof of concept involves developing an App and the supporting infrastructure to divert call internationally. The plan is to create the App to divert calls on the GSM network to a local PSTN to VoIP gateway. The server will than route the calls to another server located in the destination country. The destination server and VoIP-PSTN Gateway transfer the call to its final destination. The phone number on the last leg of the call routing should be spoofed with the original number. From a scope perspective we are looking at two to three end points in the UK. If you are interested in such a project please let me know so we can setup a call and discuss the details and the actual cost.We want to build a very simple proof of concept to help us raise capital. The proof of concept involves developing an App and the supporting infrastructure to divert call internationally. The plan is to create the App to divert calls on the GSM network to a local PSTN to VoIP gateway. The server will than route the calls to another server located in the destination country. The destination server and VoIP-PSTN Gateway transfer the call to its final destination. The phone number on the last leg of the call routing should be spoofed with the original number. From a scope perspective we are looking at two to three end points in the UK. If you are interested in such a project please let me know so we can setup a call and discuss the details and the actual cost.

  [more]

• Voip/Asterisk expert Wanted Urgently

  $515 USD 4 days ago

  We are looking for a specialist who understands Vicidial,asterisk and Voip.We just changed ISP,at an offshore location.We can no longer dial on the dialer with more than 4 agents despite the fact that we have 3Mbps of dedicated internet.What we are looking for is someone that can coordinate with our ISP and help configure the dialer to work the way it should work.You will be communicating with the ISP,voip carrier to get this doneOn our softphone we get error 408,registration time outPlease BID only if you have experience with this.We are looking for a specialist who understands Vicidial,asterisk and Voip.We just changed ISP,at an offshore location.We can no longer dial on the dialer with more than 4 agents despite the fact that we have 3Mbps of dedicated internet.What we are looking for is someone that can coordinate with our ISP and help configure the dialer to work the way it should work.You will be communicating with the ISP,voip carrier to get this doneOn our softphone we get error 408,registration time outPlease BID only if you have experience with this.

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• need a A2billing expert to upload rate and test route for new voip provider

  $154 USD 6 days ago

  Hello all,I am currently have a calling card business. I need an Expert who know A2billing and Asterisk to help me upload rate card from several companies like switch2voip.us (premium rate) to my A2billing test server. After we make the test calls, if it is successful then he/she will update the rate card and start using this provider permanently on the production server.Hello all,I am currently have a calling card business. I need an Expert who know A2billing and Asterisk to help me upload rate card from several companies like switch2voip.us (premium rate) to my A2billing test server. After we make the test calls, if it is successful then he/she will update the rate card and start using this provider permanently on the production server.

  [more]

• Custom Project May 14 2013 16:52:44

  $30 USD 7 days ago

  We are in need of some astrix/trixbox services - we have two parts:1. We have a 1-877 number that is not working and we need to have the necessary modifications made to have this working again. We used trixbox hosted by voicenetwork.2. We need to revamp our entire phone system: We have multiple companies with multiple lines based out of single office - we need to have system streamlined...among other things for each company we need auto-attendant, employee extensions, and ability/training to scale up (ie add more individuals) our system when adding users. Much of this is already in place but needs to be tweaked.We are looking for a deal. Our second project we are ideally looking for a fixed fee. Some free support to get our current issue fixed giving us time to put in place our new system would be great. Please be in touch asap as our #1 part is quite urgent. I'm not familiar with this site so I'd like to discuss project budget so please ignore what is entered below.We are in need of some astrix/trixbox services - we have two parts:1. We have a 1-877 number that is not working and we need to have the necessary modifications made to have this working again. We used trixbox hosted by voicenetwork.2. We need to revamp our entire phone system: We have multiple companies with multiple lines based out of single office - we need to have system streamlined...among other things for each company we need auto-attendant, employee extensions, and ability/training to scale up (ie add more individuals) our system when adding users. Much of this is already in place but needs to be tweaked.We are looking for a deal. Our second project we are ideally looking for a fixed fee. Some free support to get our current issue fixed giving us time to put in place our new system would be great. Please be in touch asap as our #1 part is quite urgent. I'm not familiar with this site so I'd like to discuss project budget so please ignore what is entered below.

  [more]

• Billing System for hosted application

  $824 USD 6 days ago

  We are looking for someone who could help us out with developing a program that automatically exports csv files from an ftp server and convert them to a readable format in order to bill end users.. prefer all automatically and optional web portal for the end user to check their accountWe are looking for someone who could help us out with developing a program that automatically exports csv files from an ftp server and convert them to a readable format in order to bill end users.. prefer all automatically and optional web portal for the end user to check their account

  [more]

• Call recording support for Asterisk

  $309 USD 7 days ago

  We need call recording for Asterisk PBX.We need to record every incoming and outgoing call.Each conversation need to contain uniqueid from cdr.We need call recording for Asterisk PBX.We need to record every incoming and outgoing call.Each conversation need to contain uniqueid from cdr.

• asterisk cdr viewer

  $257 USD 8 days ago

  On a working free pbx installation:Free PBX 2.10 and asterisk 1.8 we installed asterisk-cdr-viewerhttps://code.google.com/p/asterisk-cdr-viewer/We need to enable the report to download the call recording (if enabled in the extension)Thank you.On a working free pbx installation:Free PBX 2.10 and asterisk 1.8 we installed asterisk-cdr-viewerhttps://code.google.com/p/asterisk-cdr-viewer/We need to enable the report to download the call recording (if enabled in the extension)Thank you.

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• http://www.freelancer.com/projects/Asterisk-PBX-VoIP/Billing-Asterisk-Configuration.html

  $773 USD 9 days ago

  Hi we need A2billing test and some other work on asterisk to work smoothlyWe are facing issues in a2billing of logging off etchttp://www.freelancer.com/projects/Asterisk-PBX-VoIP/Billing-Asterisk-Configuration.htmlHi we need A2billing test and some other work on asterisk to work smoothlyWe are facing issues in a2billing of logging off etchttp://www.freelancer.com/projects/Asterisk-PBX-VoIP/Billing-Asterisk-Configuration.html

  [more]

• Kamailio Install

  £824 GBP 15 days ago

  The project is to repair and fully document an existing Kamalio AWS Install required for - SIP Registration - Load Balancing with failover - SIP SIMPLE Presence - IP whitelists passthrough - Full logs for all calls - GUI front endI am happy to discuss ONLY with those who have done Kamailio work in the past. happy bidding!The project is to repair and fully document an existing Kamalio AWS Install required for - SIP Registration - Load Balancing with failover - SIP SIMPLE Presence - IP whitelists passthrough - Full logs for all calls - GUI front endI am happy to discuss ONLY with those who have done Kamailio work in the past. happy bidding!

  [more]

• Asterisk: Playback During Queue Call and Listen for \"1\"

  [Sealed] 20 days ago

  Summary: We need a solution to allow our call center agents to click on a button in a PHP page which then causes the Asterisk server to barge into the current queue call of the agent and play back a recorded message. After the playback is complete, Asterisk needs to 'listen' for the external caller to press '1' and must ignore any key presses by the queue agent.More detail...1) Patient calls in and speaks with a queue agent in our call center.2) When the agent reaches the right point in the call using our web-based script system, the agent presses a button on the web page.3) The press of the button sends a command to the Asterisk server to barge the agent's existing call and playback a recorded message--audible to both agent and the patient. We will need help understanding how to pass the necessary variables (1) agent number or unique call id, and (2) patient ID number) from the PHP-based web page on a web server to the Asterisk server.4) When playback is completed, Asterisk continues to 'listen' on the call until the patient presses '1' or the call is ended by either the patient or the agent. 5) Asterisk must ignore DTMF codes pressed by agent on their local phone--and accept only touch tones from the patient.6) Once the '1' is detected from the caller, Asterisk must play one more pre-recorded sound clip and add an entry to a MySQL table noting the acceptance, the agent number, the patient ID (provided earlier via the remote call from the web page), and the exact time and date of acceptance.7) Asterisk does not need to do anything further on the call. The agent or patient continue talking as needed until one of them hangs up.IDEALLY: This process would work regardless of who initiated the call (whether we answer the patient's call or if we call the patient)-- but for the purpose of this project, it is inbound calls which we need to make this work for.Summary: We need a solution to allow our call center agents to click on a button in a PHP page which then causes the Asterisk server to barge into the current queue call of the agent and play back a recorded message. After the playback is complete, Asterisk needs to 'listen' for the external caller to press '1' and must ignore any key presses by the queue agent.More detail...1) Patient calls in and speaks with a queue agent in our call center.2) When the agent reaches the right point in the call using our web-based script system, the agent presses a button on the web page.3) The press of the button sends a command to the Asterisk server to barge the agent's existing call and playback a recorded message--audible to both agent and the patient. We will need help understanding how to pass the necessary variables (1) agent number or unique call id, and (2) patient ID number) from the PHP-based web page on a web server to the Asterisk server.4) When playback is completed, Asterisk continues to 'listen' on the call until the patient presses '1' or the call is ended by either the patient or the agent. 5) Asterisk must ignore DTMF codes pressed by agent on their local phone--and accept only touch tones from the patient.6) Once the '1' is detected from the caller, Asterisk must play one more pre-recorded sound clip and add an entry to a MySQL table noting the acceptance, the agent number, the patient ID (provided earlier via the remote call from the web page), and the exact time and date of acceptance.7) Asterisk does not need to do anything further on the call. The agent or patient continue talking as needed until one of them hangs up.IDEALLY: This process would work regardless of who initiated the call (whether we answer the patient's call or if we call the patient)-- but for the purpose of this project, it is inbound calls which we need to make this work for.

  [more]

• Caller ID Program Build for PBX/VOIP System

  $1030 USD 21 days ago

  New car service business looking to have a simple caller ID program built for a PBX/VOIP system. Quick turnaround needed.Requirements- Split screen, basic caller ID log one side - address book function on other- Basic Caller ID Log1. Keep a list of calls going. - Address book function:1. Ability to create record/entry - Name and Small Notes Box - for each phone number2. Once record/entry is created, should also allow to store calls from that number going forward, creating an activity log with automatic time/date stamp as well as a line to record address/short notes.3. Any time that number with record calls, bring up activity log.Quick turnaround needed!New car service business looking to have a simple caller ID program built for a PBX/VOIP system. Quick turnaround needed.Requirements- Split screen, basic caller ID log one side - address book function on other- Basic Caller ID Log1. Keep a list of calls going. - Address book function:1. Ability to create record/entry - Name and Small Notes Box - for each phone number2. Once record/entry is created, should also allow to store calls from that number going forward, creating an activity log with automatic time/date stamp as well as a line to record address/short notes.3. Any time that number with record calls, bring up activity log.Quick turnaround needed!

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• Distributed FreePBX/Asterisk System

  [Sealed] 20 days ago

  I have a main office with approximately 250 extensions on a single Elastix server, plus 4 remote sites, each with its own Elastix server and from 2 to 10 extensions. Although the system 'works', we want to move to a more fault-tolerant system which 'finds' extensions in the system rather than requiring statically defined outbound routes, etc. We would also like to switch to the FreePBX distro instead of the Elastix distro.So, there are two primary parts to this project:Part I: Main OfficeIn the main office, we want to utilize two servers (either in a master-slave or master-master configuration) to provide a high-availability solution with less than 20 seconds of downtime in the event a server fails. We have approximately 25 queues at present with a call volume of approximately 2,000 calls per day and 110,000 minutes of voice time per month. All voice traffic is routed either to another company PBX via SIP over the Internet or via SIP to an ITSP provider. Part II: Remote OfficesFor our remote offices, we are looking to switch from Elastix distro to FreePBX distro and to introduce automatic routing of calls, based on 'extension presence' on remote PBX servers. All voice traffic is routed either to another company PBX via SIP over the Internet or via SIP to an ITSP provider.Deliverables:For awarding this project, we are looking first for a proposal of how to structure our system (which technologies you propose that we use)Then once the project is awarded, we are looking for a detailed HOWTO on implementation, along with support for initial deployment. We would also want the project winner to be available for future paid support, as needed.I fully expect that more information is needed for being able to bid, so please ask questions and I will reply with project clarifications.I have a main office with approximately 250 extensions on a single Elastix server, plus 4 remote sites, each with its own Elastix server and from 2 to 10 extensions. Although the system 'works', we want to move to a more fault-tolerant system which 'finds' extensions in the system rather than requiring statically defined outbound routes, etc. We would also like to switch to the FreePBX distro instead of the Elastix distro.So, there are two primary parts to this project:Part I: Main OfficeIn the main office, we want to utilize two servers (either in a master-slave or master-master configuration) to provide a high-availability solution with less than 20 seconds of downtime in the event a server fails. We have approximately 25 queues at present with a call volume of approximately 2,000 calls per day and 110,000 minutes of voice time per month. All voice traffic is routed either to another company PBX via SIP over the Internet or via SIP to an ITSP provider. Part II: Remote OfficesFor our remote offices, we are looking to switch from Elastix distro to FreePBX distro and to introduce automatic routing of calls, based on 'extension presence' on remote PBX servers. All voice traffic is routed either to another company PBX via SIP over the Internet or via SIP to an ITSP provider.Deliverables:For awarding this project, we are looking first for a proposal of how to structure our system (which technologies you propose that we use)Then once the project is awarded, we are looking for a detailed HOWTO on implementation, along with support for initial deployment. We would also want the project winner to be available for future paid support, as needed.I fully expect that more information is needed for being able to bid, so please ask questions and I will reply with project clarifications.

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