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Tektrix

VOIP, Networks & IVR Specialists

Username: tektrix1

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Location: Lahore, Pakistan

Member since: May 2009

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4.9/5

(19 reviews)

6.6
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  • $1705.00 USD
    5.0
    Profile image for Seller bluethunder2008

    bluethunder2008

    May 12, 2014

    The develop is super, when it comes to delivering a high quality project. Thanks for the good work , am looking to work with him in future projects.

    Project Description:Requirement: - Use CentOS. ( it will be ready for you to access remotely ). - Configure Asterisk , Link it with our provider for 60 lines ( SIP Trunk ). - Configure FreePBX to handle and manage Asterisk...
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  • $1000.00 CAD
    5.0
    Profile image for Seller mbagajati

    mbagajati

    Feb 23, 2014

    great job and good professionalism and understanding.

    Project Description:remaining costs for the project slick90210.
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  • $277.00 USD
    5.0
    Profile image for Seller webaoousa

    webaoousa

    Aug 20, 2013

    tektrix1 is definitely a very good worker, available to help and efficient. He provide a very good work and respect the timeframe. Thanks ! Choose this worker for all VOIP related works is definitely the best option !

    Project Description:Hi, I need to have someone setup Freeswitch / Plivo (http://www.plivo.org/) and configure it to work with a live SIP trunk that I can provide. I will also provide root access to a CentOS server. ...
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  • $1000.00 USD
    5.0
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    pirat2012

    Apr 1, 2013

    THIS IS A REALLY ONE OF THE KIND VOIP MASTERS!!!! A ++++++++++++++++++

    Project Description:I WANT TO DEVELOP A CLICK TO CALL SERVER Functionality: Caller will see a website which has " contact Us" button and caller will press the button. As soon as he presses the button, Caller receives the call from our server...
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  • £1600.00 GBP
    5.0
    Profile image for Seller Chrisinjapan2

    Chrisinjapan2

    Jun 16, 2012

    Have to say exceptional professional and talented guys at tektrix. AHmed and his team delivered on time and on budget. We will defi itself be using them as our preferred developer I. Fact we are in the process of giving them more work hopefully. Fantastic and always on hand via Skype to help test and walk through the system. Works long hours and weekends even testing with me at night.Great company and very professional even when are requirements changed I. The spec they accommodated us. I would recommend tektrix to anyone considering a VoIP asterisk or freeswitch system. ACtually I would just recommend them for at type of software development as the are absolutely fantastic.Regards Dr Christopher J WoodcockTechnical Direcotr and Shareholder of Dialogsolutions.uk.com

    Project Description:We are looking for a freeswitch installation engineer to install and configure freeswitch and our SMS provider on our server. We need the freeswitch to be configured with a sip connection to our sip provider...
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  • $750.00 USD
    5.0
    Profile image for Seller bmc74

    bmc74

    May 30, 2012

    It has been a pleasure to work with this freelancer. He is very knowledgable with all that we asked and even when we had problems with our firewall he helped us through. Will use again .... Highly recommended.

    Project Description:[This is a Private Project. You must be logged in to view the Project Description]
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  • $250.00 USD
    5.0
    Profile image for Seller Nastiezo

    Nastiezo

    May 28, 2012

    Very prompt and easy to work with. We even had a few issues caused by our 3rd party carrier, tektrix1 worked with us to resolve all issues very quickly. Thanks!

    Project Description:I currently use Twilio for simple PHP / REST driven telephony. However, I need to have someone setup Freeswitch / Plivo (http://www.plivo.org/) and configure it to work with a live SIP trunk that I can provide...
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  • £400.00 GBP
    5.0
    Profile image for Seller davebrooks

    davebrooks

    Apr 2, 2012

    Delivered a fast and very smart solution to our problem, I would not hesitate to recommend or work with in the future.

    Project Description:Reconfiguration of existing FreePBX installation to disable responces to all DTMF tones. We have an existing FreePBX installation that we would like to make some minor changes to, specifically how it respondes to DTMF tones...
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  • $600.00 USD
    0.0
    Profile image for Seller kimosrolling

    kimosrolling [ Incomplete Report ]

    Mar 28, 2012

    Just a waste of time and money.

    Project Description:I need a predictive dialer that can diale several numbers at once, and put them to a freepbx queues About the predictive dialer: I have been looking at vicidial and elastix,both with a call center module...
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  • $250.00 USD
    5.0
    Profile image for Seller mrjrhill

    mrjrhill

    Feb 26, 2012

    Excellent communications and easy to work with. I will be using this freelancer in the future. Job completed better than expected.

    Project Description:We are looking for someone who is experienced with installing freeswitch, GUI interface Lua, Lua-mysql and doing all needed sip/freeswitch configuration. Install and configure: 1) latest version of freeswitch, make all maximum performance config, config sip and all other needed configuration...
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    Tektrix has not completed any projects.
  • $3777 USD In Progress

    Please don"t use generic proposals, we need your analysis and proposed solutions. A brief document is appreciated.We are in the process of developing a website application that will act as a platform of connecting users with each other via call. The concept is that a user can talk to a random user by calling the person and has to pay for that. We will taking care of whole website but we need someone to take on the VOIP part of the project from start in the end. Off course we"ll need that contractor to integrate that VOIP module with the main website. Below are some key points that needs to be implemented.VOIP Switchboard Software to route telephone calls anonymously from customer to service provider automaticallya.Allow for prompts for messagesb. Need to connect to the telephone payment processing as well as the database.All of these systems need to be connected to each other so that payment and connection are seamless.Phone System Needs Prompting service: update the customer on information on their accountPhone payment processing: functionality for the customer needs to add to their balance Directory Services: for general listing of available service providersCall back service: call back service for automatically calling back a customer when the service provider is free so customer do not need to waitVoice Recording / Playback: Functionality for recording each conversation and store in different audio format for playbackCustomers can have a option to have their conversations recorded and stored for streaming playbackPersonal library for all recorded conversations for each customerThese are just brief instructions we would appreciate serious contractor only.

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  • $41 USD/hr In Progress

    Hey guys I am looking for a WebRTC voice chat feature. The main functions I want are...Create room (system creates a 6 digit hexadecimel room number)Join room (join one of these user created rooms by manually adding room number)Up to 5 users in one callPeer-to-peer calling (as secure as possible)Other users have a volume slider (to adjust volume)Mute function to allow user to mute other usersSignalling (allow a user to know who is speaking real time)We are on a tight schedule and are hoping to get started this weekend and would ideally have it completed in one week. Our project budget is roughly $2000.I WILL ONLY RESPOND TO MESSAGES WITH "SOCHI OLYMPICS" IN THE SUBJECT. PLEASE INCLUDE THIS IN YOUR MESSAGE SO I KNOW YOU ARENT A BOT.Thank you

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  • $477 USD In Progress

    1) We have a customized Cisco Phone script which need to make some additional changes!2) We need to block international calls with Pin on Cisco Call Manager EXPRESS which is a module in Cisco 3800 router (we don"t have call manager server)

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  • $666 USD In Progress

    We are looking to setup a cheap hosted IVR phone verification system/application with voice prompts, that can be called by phone. Should have good natural text to speech functionality, fetch information from database and authenticate caller, transfer live call. This is a small project and project duration should not exceed three days. We prefer to host this application to safe cost and should have web interface to forward calls to any phone number. The script will be provided after awarding the project. This project is very time sensitive and project duration should not exceed 3 days (4 days at most). We want to deal with only knowledge expert in hosted IVR/PBX solutions only that has access to voice artist.

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  • $666 USD 25 days ago

    Implement a scalable VoIP Telephone System that will interconnect 3 colleges.- User in college A should be able to communicate with user in college B from their local networks (even if the users don"t have internet).- IVR configuration is needed to cater for the college administrative needs.- A web interface to register the SIP users and Billing - Implement QoS (on the router) for VoIP within the college to prioritize voice.IMPORTANT NOTICE* You are only setting up for one college* The users are using only Softphones

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  • $777 USD 25 days ago

    We have a Programmer from Bangladesh disappear. We had operating system but now it is broken :( Need it running for a demo and minor changes. Thomas@txtCOINSnow.comSKYPE: [The administrator removed this message for containing contact details which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

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  • $1000 USD Jun 27, 2014

    We need a Freeswitch based system which can handle basic IVR and outbound calling and dialer functionality thru a SIP line. The system should be capable of handling high volume of calls.We would require your assistance with installing, configuring and building an outbound call bridging system to provide add based calling to our users.The system will also have a basic IVR system.We would provide remote control access to the server with the SIP line.Once the system is built, we need support for a limited time and training of our technical staff on the built system.We are open to both hourly billing or fixed price models.Part 2: We would also require an advertiser portal where the user can log in and add advertisements to the phone call.

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  • $30 USD Mar 28, 2014

    hi i need to install asterisk pbx on my linux serverwith the configuration to unhide block caller id

  • $1000 USD Dec 1, 2013

    Hallo my name is nelson,i am in calling card business...my voipswitch platform generates internal pincodes with different monetary values that we sell to customers so that when they call our service number they can recharge and call their international destination number.We have limited distribution channels for our pincodes although we wish to expand beyond our borders we have therefore partnered with Paysafecard which offers vouchers with pincodes and is well distributed around the world.We therefore wish to customize our voipswitch and use paysafecard pins as the default recharge pins.The only modification to be done is creating a path to send the pincode to the paysafecard server for authentication which will reply if the card is verified or not and the monetary value and the currency of the card.I have attached Paysafe card documentation which shows how you should create that path from our voipswitch to their server.

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  • $3888 USD Jun 5, 2013

    Total Solution NeedFor a IVR Telephone Dating Service Program1 Multiple callers are connected simultaneously2 Callers choose who they want to talk with3 Voice profile setup capability4 Can detect area codes and switch lines5 System must be easy to manage and adjustexample sites: http://www.matchandtalk.com/Do you understand this business?

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  • $555 USD May 30, 2013

    3 GroupSales with 5 LinesAdmin with 2 LinesAccounts with 1lineAccount will have 1extension (300)Admin will have 2extension (100,110) with a ring group 1000Sales will have 4extension (400,410,420,430) with a ring group 2000voice mails are active for the groups goes to mail address and call out from admin department will use the 2 line onlycall out from sales department will be using the 5line respectivelyall the calls coming in to admin lines it will ring the sales groupall the calls coming in to sales lines it will ring the sales groupcall transfer should be possibleall the calls going out for 07xxxxxxxxx number will be via GSM A400P cardall the calls coming in from GSM will go to Sales line extension 200 with different ringtone so we can understand from the GSMCall coming in(British female best quality )thank-you-for-calling.:Thank you for calling.3secfirst-in-line.:your call is now first in line, and will be answered by the first available representativeafter 5secpls-stay-on-line.:Please stay on the line and your call will be answered by the next available representative.System will ask who is calling Ask for Name and Company (keep this asking part as long as possible so user can have time to talk their name better)put on Hold musicUser pick up the phoneSystem tell the caller infooptions1 answer2 send to voice mail (to caller "I am sorry all our Agents busy right now or say staff in training there is no one to answer the call etc , press 1 to leave voicemail or 2 to hangup )3 keep on hold music for 3min and ring the next extension available should be able to transfer calls to each otheradmin extension should be able to join into the call anytime using special code.should be able to pick a call from any location (if extension 400 ringing 430 should be able to pick from his location)In a case the lines are busy when 1user try to make a call it should add in a queue to dial auto as soon as the line are ready(but this can be asked "Want to add to the queue " or something ?Calls are recorded all time.

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  • $1500 USD May 24, 2013

    Our main goal to minimize the BW in client side with good quality of voice .We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B.Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario:1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards.3. Number of Server B can be unlimited.4. Number of Gateways/E1 cards per server B can be unlimited5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider)A. Any mini Linux distribution exam- puppy Linux , linux mint B. Fedora desktop distributionC. Centos 5.8 or 6 7. Server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination. we will used . A. iax trunks in trunking mode.B. Open vpn static mode and dynamic mode C. Tnic static and dynamic mode 8. Asterisk web billing gui for adding gateways. Adding client , Prefix , dialing plan viewing active calls, billing cdr ,etc.we will provide you the Dedicated server asterisk and client asterisk configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW)continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may servers.continue working on project by building up WEB interface for main server adding Billing, and other options from Item 2 like adding GW, adding client, adding IAX trunks here I add some company we need similar thinghttp://rbctechbd.com/www.syncswitch.com/content/sboplease contact with us ASAP if you can do this project

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  • $500 USD Dec 31, 2012

    hii have one task to set up voip server on asterisk technology. its urgent project.what information do your require?? please let me knowthanks

  • $30 USD Nov 13, 2012

    I have crackling sometimes & in some lines in incoming calls. Is it possible to do something for this?

  • $30 USD Nov 6, 2012

    Hi pls add my skype: khan99271 then we can talkKhan

  • $250 USD Nov 5, 2012

    Asterisk coding needed to enable unmask of caller ID over SIP trunk with supported DID service, module is very simple:1. Incoming call to DID2. caller ID stripping from PAD header3. hang up4. initiate 2 seconds call back to predefined number with caller ID that was stripped from blocked call5. hang upPlease bid only if you have knowledge in unmasking CID and have DID / Trunk providers to support both tasks.

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  • $400 USD Oct 30, 2012

    Hi,We are going to setup a new PBX based on Elastix. Our current CRM system has multiple servers and connections for services like Caller ID, Fax, Voice Recording, PBX statistics. We need to deploy the new Voip PBX and make sure that all the systems are working together with the CRM.Our PBX will have 4 ISDN BRI on an OpenVox BE400P 2 PSTN FXS and 2 PSTN FXO on an OpenVox A400P10 IP Phones, 1 Fax machine (main office)1 IP Phone, 1 Fax machine (branc office)You should provide:1. Basic PBX configuration (extensions, truncs, incoming routes, outgoing routes, IVR, queues etc)2. Configuration to send fax from email on local net3. Configuration for custom UPD broadcast for Caller ID4. Configuration to store voice recordings on a windows machine with custom file name5. Configuration of system security, fail2ban, backups etcWe will provide:1. Dial plans2. IVR flowchart3. System recordings if needed 4. Any other info needed to complete the project.

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  • $2000 USD Oct 23, 2012

    Program should catch incoming/outgoing phone call events and start recording audio streams to the web server (through restful API provided) or memory or file. Phone call should not be interfered with this new functionality. Program should not be platform-specific and should work for, at least (but not limited to) android and iOS. One possible way to do this would be make a simultaneous call to a pre-configured machine using VOIP and put the three parties in a conference. It is upto the programmer(s) to implement a better technical solution, if possible.Working on this project requires solid understanding of how mobile telephony works. Knowledge of Objective C or Android Java is not at all required, nor is any experience in creating mobile apps mandatory.Development of mobile applications (Android and iOS) and web server (with rest-style services) are done separately and should not be part of this project. Please include details of requirements from telephony service providers or others so that we can facilitate these without any delay. Please include brief description of your technical solution so we can assess the scalability and maintenance costs of the solution for our future requirement.

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  • $30 USD Oct 11, 2012

    Our main goal to minimize the BW in client side with good quality of voice .We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B.Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario:1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards.3. Number of Server B can be unlimited.4. Number of Gateways/E1 cards per server B can be unlimited5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider)A. Any mini Linux distribution exam- puppy Linux , linux mint B. Fedora desktop distributionC. Centos 5.8 or 6 7. Server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination. we will used . A. iax trunks in trunking mode.B. Open vpn static mode and dynamic mode C. Tnic static and dynamic mode 8. Asterisk web billing gui for adding gateways. Adding client , Prefix , dialing plan viewing active calls, billing cdr ,etc.we will provide you the Dedicated server asterisk and client asterisk configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW)continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may servers.continue working on project by building up WEB interface for main server adding Billing, and other options from Item 2 like adding GW, adding client, adding IAX trunks here I add some company we need similar thinghttp://rbctechbd.com/www.syncswitch.com/content/sboplease contact with us ASAP if you can do this project

    [more]
  • [Sealed] Feb 29, 2012

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

  • $30 USD Jan 6, 2012

    need to configure trunk settings in trixbox to matchhttp://open.didww.com/index.php?title=DIDWW_Configuration_Scripts#Configuration_Scripts_for_FreePBX

  • $100 USD Dec 16, 2011

    I want to set up VOIP business and need A2billing software or similar billing platform and possible hosting plan. Please give me your best price for the complete package.1) billing software2) Soft switch3) Website with payment cart and on-line registration( Can clone www.vonage.co.uk)4) PBX softwareCan use my laptop as server Spec: Intel Pentium P6100, 4GB DDR3 Memory, 500GB HDD at present have Windows 7 installed. We can use this for trial purpose, but need to move on to a larger hosted server in about 6 months time. Please give your best price in Pak Rupees and time frame

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  • £250 GBP Sep 14, 2011

    Setup Trixbox PBX with SIP trunk, CISCO IP7940 handsets, hunt Groups, DDI, Voicemail and IVR for a small call centre in Manchester

  • £250 GBP Sep 7, 2011

    Hi,using CentOS 6, Asterisk 1.8.6.0, FreePBX 2.9.0.7 Framework. There will be no direct access to our PBX for security reasons. We require a dialplan creating that will do the following:1) Dial a list of numbers from a MySQL Database.2) Once each number is dialled the call will be instantly terminated and return to MySQL the status of each number e.g: dead number, answerphone, live number. Dead numbers are numbers that are not in service. Live numbers are numbers that are active.We have over 500,000 to test this on so would want around 50 calls a second being dialled. This needs to be a quick script with the ability to easily turn up or down the dial rate. We are able to build this ourselves but due to staffing levels we simply dont have the time!Thanks

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    Tektrix does not have any open projects.
    Tektrix does not have any work in progress.
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Résumé

Experience

Director Technical

Jan 2008 - Present (6 years)

Tektrix

Tektrix an emerging Contact Center, Enterprise networking and VOIP consultancy company, based in Lahore Pakistan. We cater to every size and budget company both locally and internationally by tailoring our vast experience in Voice Over IP and Enterprise Data Networks.<br /><br />Our expertise are the Cisco IPCC, IPT and enterprise open source solutions for VOICE and DATA.

Education

MS

Pace University

2000-2002

Certifications

Cisco IPCC

Cisco

Cisco IPCC (7.0) in house training from Cisco Trainer.<br />Cisco IPT In house training from Cisco Trainers.<br />Cicso PGW/SS7 in house training from Cisco Trainer.<br />FastLane Cisco CVP3.1 Training<br />FastLane Cisco ICMSA Training