Last seen: Apr 16, 2014 6:15 PM EDT
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Rate: $60.00 USD/hour
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Alexander Merkulov

best quality voip experts

Username: meral

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Location: Kiev, Ukraine

Member since: April 2007



(205 reviews)

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3 users have recommended this freelancer.

My skills:

  • $618 USD
    Profile image for Seller atsak


    5 days ago

    Pleasure to work together on this.

    Project Description:I'd like someone with experience setting up A2Billing to setup a billing platform for me on a new PBX we'll be using for clients who buy communications services for us. I will provide a PBX In a Flash server setup you can SSH to and setup, and we can work together to determine how to set it up...
  • $115 AUD
    Profile image for Seller olod


    Mar 16, 2014

    Project finished successfully and on time.

    Project Description:Need to check settings for CISCO 7970 advise what to change to phone to register. What you be budget for this job?
  • $240 USD
    Profile image for Seller calvinbradshaw


    Mar 3, 2014

    Excellent follow up work. Thanks.

    Project Description:Hi Alexander. I have put a copy into in production; The team have reviewed, and they have 2 requests; (1) We need a version without "Total queue time" column...
  • [Sealed]
    Profile image for Seller calvinbradshaw


    Feb 25, 2014

    Excellent work, very skilled and talented with our complex Asterisk / Elastix / SQL / VOIP programming needed. Great communication. Highly recommended.

    Project Description:[This is a Private Project. You must be logged in to view the Project Description]
  • $115 USD
    Profile image for Seller cmic


    Feb 22, 2014

    Excellent as always. Got the job done fast and taught me some things too.

    Project Description:As discussed in email, here's the project for the two tasks that we discussed: TASK 1: Simple IVR System We just need a simple IVR to handle two button presses ("1" for Sales, "2" for Support). We have .gsm sound files ready...
  • $532 USD
    Profile image for Seller mda2011


    Feb 9, 2014

    One of the best voip professionals we have worked with, thank you Alex. Will work with you again!

    Project Description:Hi we got CIsco 7960's with SIP firmware and we would to work with someone to make it work with end point manager on freepbx. we are very serious about finding a experienced engineer to do this task. we are willing to accept TFTP, HTTP, FTP provisioning...
  • €103 EUR
    Profile image for Seller ateneahost


    Jan 28, 2014

    High knowledge of Unix - Linux - A2billing

    Project Description:I am looking for an A2Billing Click2Dial Script: 1. Creation of a PHP script that will initiate a “Call-Back” to an A2Billing subscriber’s registered CID and connect the call to a requested phone number 2...
  • $500 USD
    Profile image for Seller Hillburg


    Jan 12, 2014

    Working with Alexander is very nice as he knows what he is doing...

    Project Description:Hi, I saw that you did a project before relatively close to my inquiry, so i am offfering this project to you directly. You can find the brief information as follows: The call will hit to asterisk, it will route the call to a trunk...
  • $150 USD
    Profile image for Seller ricx31


    Dec 19, 2013

    Great programmer, have used him several times and he always delivers.

    Project Description:Write a program to send a dial string to softphone from Dialcmd.exe.
  • $257 USD
    Profile image for Seller michaellancaster


    Dec 19, 2013

    Superb developer, understood exactly what we needed..

    Project Description:We are looking to develop a feature on asterisk that will allow us to define a random hold time for answered calls from say 1 to 10,000 seconds. So a call will arrive on the asterisk, drop into extensions.conf where we can define a random call length for the number dialed...
    Alexander Merkulov has not completed any projects.
  • [Sealed] In Progress

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  • $670 USD In Progress

    Meral as discussed, The following is the work to be done:\r\n1. Bind CID to DID\r\n2. Bind PIN to DID\r\n3. Daily limit per user, \r\n4. Blocked calls

  • $1030 USD Today

    We require an IVR phone survey module that can be used to evaluate the quality of customer support calls and agents. At the end of the call an agent will ask the customer if they are prepared to answer a short survey and the call will be transfered to the survey module. The customer will be asked questions and asked to press the appropriate number key to indicate their response.Detailed requirements:* Must be delivered as a FreePBX module compatible with FreePBX and asterisk 1.8.7* Must be configurable through a web page on the FreePBX web interface* Configurable features must include: number of questions, range of values for response* Results will be stored in a MySQL database with information on the call, the agent, the questions and the answers to the questions* There will be a web interface to display the result for each question across multiple agents as a bar chart* Changing the questions will not invalidate the data in the database* All source code will be provided

  • $103 USD/hr Yesterday

    This posting is for an Asterisk expert to provide architectural overview of the viability of an Asterisk project.The project is meant to allow a main site and several (potentially over 10) sites to be on one Asterisk network using the Internet as the fabric.This part of the project is simply to review the design with an Asterisk expert--a freelancer-- to ascertain the viability of the architecture.The main project, to follow this phase, will be to implement the design.The ultimate intent is to also award the implementation of the design to the Asterisk expert selected for this project, provided that this first phase (the architectural review) is mutually satisfactory.The protocol for this phase will be to connect with the freelancer for a dedicated time (say an hour or two to start) and to exclusively spend the time reviewing the design using the available tools at to communicate.

  • ₹38144 INR Yesterday

    We have Asterisk based vicidial call center dialer which have MySQL 5.0.95 and PhpMyAdmin version 2.11.10 we have to upload our customer database into our Asterisk based vicidial call center dialer and if any incoming call hit to the server it has to check the database first if number is exist in the database then it has pass the call to one particular campaign if number is not present then it has to pass the call into the fresh call campaign.... so i want to script to happen these things...

  • £2061 GBP 3 days ago

    I need someone who know pbx,inside out. i want someone to test the security level of it.

  • $515 USD 5 days ago

    [This is a Private Project. You must be logged in to view the Project Description]

  • $144 USD 6 days ago

    i have a Cisco CP-7960G that i would like to use with regular SIP account. i would be willing to pay if someone can teach me how to convert cisco SCCP to sip.

  • ₹134020 INR 6 days ago

    We would like to get a real time monitoring system built to track agent status in a call center environment. We use asterisk based telephony system on to which we would like to build an interface to display agent status. Whether the agent is on call, idle, hold etc.. and corresponding duration.Interface must also have the functionality to map agent extensions to various users,display historic reports and basic analytic capabilities.

  • $60 USD/hr 7 days ago

    Hello Merle, I have a Vicidial system did not need to have debugged. It crashed a few days ago and I am not really sure why it crashed.I would like for you to log into the system reviewed the logs and see if you can find any issues.Also, Vicidial is reporting more channels used and what I am seeing in the asterisk consult. I would like to see why there is a difference here as well.

  • £257 GBP 9 days ago

    Hello, we are looking to upgrade our current Asterix VoIP server to the latest version and test to ensure everything is still working. We have a number of AGI scripts that handle diverting calls to mobiles, looking up permissions to see if a user can dial externally and other database lookup queries. An example is a member of public cal call into the system and dial an event ID number - the system then looks up the leader of the event in the database, finds the number and forwards the call to their mobile

  • $2061 USD 12 days ago

    Hello,We are looking some one who will able to complete a chan dongle with asterisk project. This project include following features1.Installation and configure chan dongle for incoming and outgoing call2.Should have a user friendly web interface. The web interface should contain the following options•Option to check mobile balance through USSD for all connected dongles on same time.•Option to check port status like dialling, ringing or connected like active call.•Option to active and inactive dongles individually•Option to check dialy or hourly or custom time basis cdr and acd•Option to call each dongle to others dongle on certain condition. Like after certain success call or certain duration.•Same should have option to send sms•For above both option need to put sim card number along with dongle number to handle automatic call when fulfil any condition.

  • $144 USD 12 days ago

    I need to integrate AteriskNOW or Elastix with Active directory.

  • [Sealed] 13 days ago

    We would like to automate invoice creation from an a2billing instance. We use freshbooks for a billing platform. Freshbooks has APIs exposed to automate the billing process, a2billing is open source and there are publicly available scripts for querying the mysql database for billing information. All of the components seem to be available, we simply need someone to put the pieces together.

  • $61 USD/hr 13 days ago

    Asterisk development project to allow telephone functionality through a web site. We want to develop a very sophisticated web based UI for Asterisk as well as customize the software itself to fit our purpose. We are looking for a serious programmer who can take the lead on this multi-faceted development project. We are seeking a talented, creative and very knowledgeable programmer who is available to dedicate the resources to make this project successful. You will be required to sign a non-disclosure agreement.

  • $515 USD 22 days ago

    Hi,i am looking for someone who can install Asterisk on my server for wholesale IP2IP traffic with Codec Transcoding , like my Gateway support only G729/G723 only and my carrier client will send traffic with codec "G729 A, G729 B, G729, G729 A/B, G711 A, G711µ" so it is possible that Asterisk cover the codec ->G729 A, G729 B, G729, G729 A/B, G711 A, G711µ to G729 ? and pass the calls successfully ?Thanks

  • $309 USD 22 days ago

    I need someone to modify the call center module of elastix.We are running an outbound call center and generate many calls per day.The elastix Call center module is great for us but there is one drawback that we need to get fixed.We are looking for an asterisk/elastix experienced programmer to modify the script.Here is the problem we are looking to fix:Whenever a call is placed and ended, the agent on the call do not have enough time to enter any notes or disposition or any other details of the call. We would like to add a delay between each call so once the customer hangs up, the agent is still "in call" and once he save the form information , he will be available to receive calls again. it seems this needs to be modified within the script.If all goes well we have a few other modifications we would like to implement.Please only message me if you have asterisk or elastix.

  • $103 USD 23 days ago

    Simple install/basic config of Asterisk on an Amazon EC2 instance.

  • $309 USD 25 days ago

    I need to reconfigure 3 existing asterisk servers to do SIP/IAX2 trunking , and troubleshoot why calls drop frequently from asterisk to voip gateway.

  • $2061 USD 26 days ago

    Hello,We require the building of a VoIP Switch which focuses on inbound DIDs. It must handle 1000 concurrent calls.We have our own number range consisting of over 1000 000 DiDsIt will have multiple trunks for over 10 providers who will send calls to this switch.The switch must accept the call and RECORD the IP/TRUNK/CHANNEL where the call is originating from.It must also route the call to the relevant SIP Account the DID is allocated to.We need to allocate a cost on each channel so each call has a cost to it based on prefix.At month end, we must be able to draw out a report stating the inbound calls via the relevant channels and its cost, with a total of it.The CORE should be something sturdy, with data saved in a database.Opensource CORE is welcome. Either Kamailio, A2B, FreeSwitch, Kazoo, or any other robust switch and billing system. There is NO outbound VoIP in this switch, its purely inbound.Ultimately, we need to know which provider is calling our DIDs, and we need to allocate a cost so we can bill them for sending calls to us.We need someone with good experience to handle this project. TxMo

  • $515 USD 26 days ago

    i have something wrong with a2billing. DID Destination billing not working. call connect no problem but billing not working. actually i want to put that feature which one u made about 2-3 years before. please contact with me on skype [Contact information removed by Administrators which breaches the site Terms and Conditions]

  • $150 USD 28 days ago

    Hi,We are a small voip provider and looking to get the core network enhanced.We currently have a multi-tenant hosted PBX solution that we"ve build our self based on 2 asterisk 1.8 and mysql replicated db.What we need is a scalable solution that will have device state replicated, proxy frontend and replicated db backend.Also our existing dialplan will need to be adapted for the new environment.The system should be hosted in Amazon AWS.Please let us know if you are interested in taking this job and we will give you the full specs of the project.Best regards,Igor Toma

    Alexander Merkulov does not have any open projects.
    Alexander Merkulov does not have any work in progress.
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master voip developer

Mar 2003 - Present (11 years)

UA Prime LTD

voip developer,work as freelancer.

voip developer

May 2000 - Mar 2003 (2 years)


created billign system for voip, voip support<br />cisco/asterisk



Kyiv National Taras Shevchenko University





Cisco network level basic certification


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