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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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  • $1000 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 USD
    5.0
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    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 USD
    5.0
    Profile image for Seller pirat2012

    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 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 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 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 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 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 USD
    5.0
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    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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  • $50 USD
    4.0
    Profile image for Seller damianfernandez

    damianfernandez

    Sep 20, 2011

    Great freelancer! Completed project ontime. Skilled with Asterisk.

    Project Description:Looking to develop a simple asterisk system that will be for customer service. users will connect, provide credentials, and access system menu. Users will be able to update their record status using their acct number...
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    Tektrix has not completed any projects.
  • $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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  • $3000 USD In Progress

    Hi Tektrix,I attach the project as discussed before.I would like you to tell me the following before start:- how will the language translations will work exactly- Which kind of GUI I will have and what will I do with it- which kind of log I will have- the exact server I should rent on godaddy.comI"ll wait your answer to start working soon.Best regards,Steph

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  • $41 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 freelancer.com to communicate.

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  • $1333 USD Yesterday

    We are looking for Freeswitch development. We have an existing IVR application which we want to convert to Freeswitch. Please share your expertise in it and sample application to evaluate your experience in this domain.

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  • ₹33333 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...

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  • £111 GBP 3 days ago

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

  • $1333 USD 9 days ago

    Configure an freeswitch and elastix servers as follows:Freeswitch server as follows:•Configure sangoma Quad T1 board:oFour ISDN-PRI, protocol 5ess, ESF, B8ZS.oTwo will receive inbound traffic with numbers to be dialed out on the other two American telephone Format with 9 for outside line (9-XXX-XXX-XXXX)oThe other two ISDN-PRI are connected to the Central Office and receive I outbound and inbound traffic.oPRIs pass ANI and DNIS information to and from all four PRIs•Configure server to record all calls that go in or out of the server. Recordings of call, if possible, will have file names that contains number dialed outbound/Number called Inbound and stored in directories by date. So directories names will have date MMDDYYYY, and file names will have 2225552222-3800. Dialed number and dialing extensions for outbound calls.Elastix server•Configure as a Fax server (can be a different instance)•Entities in the systemoCustomeroDIDoDID ProvideroCarrier•Fax ServicesoFax to EmailEach customer will be assigned one or more DIDs.Whenever Fax will come to any DID, system will lookup for the customer to whom the DID is assigned. Once customer is found, system will lookup for email id of the customer.System will receive fax in tiff format and convert the tiff file(s) to pdf. The pdf file(s) will be sent as attachment to the email address of the customer. A global cover page will be added as body of email. Password protect pdf files. Password will be set per customer. So customer needs to use the password set for his account to open the pdf files.System will log CDR for the received fax.oEmail to FaxAny customer can send fax by sending email to a number of a fax to the domain of the fax server. Like 201551212@ the domain name of the fax server. .Email will come to the email server configured on the same server as the fax server.System will authenticate the sender using the from email address and proceed further if the email address is of any customer.System will accept just pdf/tiff attachments to be sent as fax. System will convert pdf to tiff and send it as fax to provided faxnumber. Either fax sending process fails or succeeds, system will send job confirmation email to the sender with the relevant status. ( a global format for the job confirmation email will be created.)Customer"s Primary DID number will be set as callerid when sending Fax to given Fax number.Decrypt password protected pdf files and send as fax. Password will be set per customer. So customer needs to use the password set for his account to open the pdf files.System will log CDR for sent fax.Predefined number of retries for failed fax. Notification email will be sent for each attempt.•CDR LoggingoCDR Fields ( Please confirm if any change in cdr fields)Date-TimeCustomerCallerIDDialed NumberStatusDurationTotal Page NumberSent/Received Page NumberoCDR Logging will be done in flat files (csv format) as well as in database.oSystem will be configured to archive csv files on daily basis @ 12 midnight GMT.•Admin PortaloManage CustomersoManage DIDsoManage Customer - DID assignmentoCDR Report•Customer PortaloEdit Customer ProfileoCDR ReportoList of assigned DID•Audit logoSystem will be configured to write audit logs for fax2email and email2fax services into flat log files. System will be configured to rotate the logs on daily basis and keep logs for predefined number of days.o Allow multiple simultaneous faxes to be received for the same number (not sound busy tone)Assume that the server will be installed and at base line installation with latest level of elastix and free switch. Access via the internet to both server. Documentation of configuration and changes is required.

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  • £277 GBP 11 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

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  • $2500 USD 13 days ago

    I need to integrate Fusion PBX with ASTPP or A2Billing.Only bid if YOU HAVE DONE this before....Please bid, and be ready to show me that you have already integrated. If there is no demo of an integrated system as required above, then DO NOT bid.Mo

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  • $30 USD 20 days ago

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

  • $1666 USD 22 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

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  • $555 USD 23 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

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  • $277 USD 24 days ago

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

  • $2777 USD 29 days ago

    Voice Chat with the following features;1. Subscriber registration2. Search for Friend 3. Direct call with a service code4. Subscriber status online/offline5. Leaving message to chat users 6. Listening/Answering left messages 7. Chat user rating 8. Web based maintenance, monitoring, reporting (time based, user based, etc.)

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  • $55 USD/hr 29 days ago

    Need resource to add to current level of .net platrform working with Freeswitch softswitch.IVR functionality.

  • $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

    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