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Errol Samuels

Experienced VoIP Consultant

Username: esamuels

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Location: Huddersfield, United Kingdom

Member since: November 2008

Reputation:

5.0/5

(25 reviews)

6.6
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1 user has recommended this freelancer.

My skills:

  • $900.00 USD
    5.0
    Profile image for Seller globalrap

    globalrap

    Jun 25, 2014

    Errol is great to work with. He is very knowledgeable and I will hire him for my next projects.

    Project Description:Install, configure and document billing system Document configuration of billing system 1 hour walk through of billing system Secure server (firewall/fail2ban)
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  • $2500.00 CAD
    5.0
    Profile image for Seller it2is

    it2is

    Jun 2, 2014

    Samuels is a very smart and great freelancer to work with, always go beyond expectations; definitely we will hire him again

    Project Description:Good Day, We would like to setup a multi-tenant telephony system with the following components: Kamailio (proxy registrar), FreeSwitch (media), FusionPbx (admin & management), Postgres The freelancer...
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  • $180.00 USD
    5.0
    Profile image for Seller Christine1960

    Christine1960

    Mar 7, 2014

    Errol had to spend more time on this than quoted (not due to his fault) but finished the job without complaining or asking for more money.- Job completed 100% to spec.- Excellent experience.

    Project Description:finalise intsallation of godial on centos
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  • $3200.00 USD
    5.0
    Profile image for Seller jaugust

    jaugust

    Jan 8, 2014

    Errol is exceptional, truly a star player with a breadth of expertise. I felt like I was working with a partner from start. Errol demonstrated an ability to go the extra mile; you will want to work with him long-term. I wish I would have found him 10 years ago.<br/>

    Project Description:We are a small 6 man shop providing IT-services for our customers. We currently provide VoIP/SIP services to a few customers with on-premise Elastix PBX&#039;s. We seek to consolidate the on-premise PBX&#039;s to a virtual multi-tenant PBX...
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  • [Sealed]
    5.0
    Profile image for Seller capxange

    capxange

    Dec 31, 2013

    Errol is an exceptional developer. Highly skilled and creative. Designed, developed project beyond my expectations. Probably the best freeswitch/fusionpbx developer in the market place.

    Project Description:I currently have an asterisk pbx project loaded on AWS ec2. The current system is connected with a sip trunk that allows callers to pull up basic information on their account, located in MySQL and route calls accordingly...
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  • €3024.00 EUR
    5.0
    Profile image for Seller Menefre

    Menefre

    Dec 6, 2013

    First I want to say, that we were not looking for a short time hire but for someone we could build a partnership with and who would accompany us for the coming time. After a couple of months working with Errol I can happily say that we have a very strong, deep and professional connection and I see us working together for the next years.Errol has a deep understanding of Systems based on his tremendous experience in that field which he developed through working for many years in this industry.We had some particularly challenging aspects in our project which Errol solved because of his experience but also because he quickly adopts to new environments and learns very fast.On the social side I would describe Errol as a strong networker, a person who loves to communicate with people and always tries (and succeeds) to create a positive atmosphere. I enjoyed working with Errol and I am very much looking forward working with him in the future.<br/>

    Project Description:We are a little 4 man company providing IT-services for our customers. Lately we developed plans to provide VOIP services too, virtual PBX to be specific. Unfortunately in that field we have no experience whatsoever...
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  • $111.00 USD
    5.0
    Profile image for Seller pkjansma

    pkjansma

    Dec 2, 2013

    Will definitely hire again!

    Project Description:I need the ability to block an extension from using any trunk or outside lines. They can only call with the local designated extensions. This configuration needs to be configured in a way that an Elastix update won&#039;t crash it...
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  • $103.00 USD
    5.0
    Profile image for Seller carlosv72

    carlosv72

    Oct 31, 2013

    Excellent professional, impressive delivery time and performance, hope to work together again soon, thanks!!

    Project Description:I have freeswitch installed on a virtual server and I want to add G729 codec to the list of codecs available. There are ITU and IP Codecs, I would like IPP codec installed.
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  • €618.00 EUR
    5.0
    Profile image for Seller tillmanz

    tillmanz

    Aug 26, 2013

    Extremely professional worker! Very knowledgable!I am glad I have found him and will gladly work with him on future projects!

    Project Description:Objective is to configure an existing test-installation (no traffic) of Freeswitch on an Amazon EC2 instance. You need to be proficient in the Freeswitch Dialplan XML and also in Python as the server is running a Python script for DNIS manipulation based on inbound IP address and other criteria...
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  • $842.50 CAD
    5.0
    Profile image for Seller samimetro

    samimetro

    Jul 17, 2013

    excellent to deal with, easy going. very clear/detailed and a perfectionist.thank you for excellent work (again)

    Project Description:We need a Proxmox Cluster setup that would allow us to have High Availability for our KVM Virtual Machines where are running our VoIP Platform. Our VMs are running Elastix / A2billing and Bluebox and...
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    Errol Samuels has not completed any projects.
  • $440 USD In Progress

    I have a 24 port fxo card connected to pots lines.These lines do not have answer supervision.You will need to install asterisk and configure the card for outgoing callswhen there is a busy message or a busy tone give error 503when person on other end phone is turned off (usually message( give 503When call is answered by a person or a voicemail start billing in asteriskyou will be giving ssh to a centos 6.5 boxPlease do not bid if you are not confident that you can get this to work as I will give you a Negative feedback.Please do not bid if I will not be talking directly to the developer.

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  • $3298 USD In Progress

    We are looking for a person with experience and knowledge in deploying VoIP solutions.1) This project involves deploying a Multi-tenant PBX solution.2) Kamailio will Load balance traffic between the FreeSWITCH servers located in different locations.3) Managing users will be done through a central PostgreSQL Replicated DB4) Users will connect via TLS to Kamailio / SRTP for the mediaPlease feel free to contact me for further Questions.Thanks!David

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  • $789 USD 9 days ago

    Dear All,I have install Asterisk PBX and Free PBX, and configued dial plan, but some time call successful , sometimes failure, call drop very high, I need somebody know very well Asterisk and how to track the call.I have use tcpdump -i eth1 src host source ip address, to export failure call and success call to two files named Call 1.pcap, and call 2 .pcap.Call flow please refer to PPT.Who can tell me which call is the failure call? Call 1. pcap or call 2.pcap? Whose answer is correct , who win:))

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  • $1263 USD 21 days ago

    A new, small law firm with offices in Austin, TX, London, UK. and Moscow, Russia requires a new VoIP telephone system for each office. Advice regarding system type, hardware requirements, and installation is required. Consultant will coordinate with an employee of the firm in each location to direct installation.

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  • $500 USD Jun 10, 2014

    We would required skilled programmers to integrate g729 codec to pjsip stack to build SIP based Mobile client

  • $150 USD May 17, 2014

    I need an install of ASTPP with Freeswitch in an Openvz container. The OS can be either Debian 7, Ubuntu-12.04 or CentOS-6.5. What is your recommendation? This can be a single server install as it is a proof of concept. If my demo is successful, I will need help scaling to separate DB, Opensips and Freeswitch servers.I will run wholesale traffic through this switch(mostly China CC). For this single install, would you recommend installing Opensips?I will need all functionality working (including CDR reporting). I will need some documentation on how to configure correctly. All provider and customers will be authenticated by IP and Tech prefix. I offer each Customer a Standard, Silver and Premium route based on tech prefix. I blend my providers to achieve this. I would like to be able to white label at some point.I am available to answer any questions.I am available to discuss

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  • $200 USD Mar 19, 2014

    hi i would like a solution for answer supervision on my pstn lines which does not have any answer supervision or reverse polarity.

  • $150 USD Feb 26, 2014

    We are looking a opensips and freeswitch expert for a specifically project. Do you have enough time to discuss project details ?Project is like that; timely opening ports on freeswitch shall send port available and time period (sec.) to open sips and ur billing-proxy server will send all calls to open sips. Opensips will manage all thess calls as available ports on frees with also freeswitch side need some arrangements for this plan. Will been shared after agreed on other parts. Values are entried only complete offer. If have any question please do not hesitate to ask.

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  • $55 USD/hr Jan 16, 2014

    I have worked with multiple distribution of linux like centos, RedHat, Gentoo, Suse,Ubuntu etc.Specilized in Administration of linux, apache , mysql, Knowledge of Networking and Vmware , Citrix XEN

  • $1000 USD Jan 10, 2014

    Hello,Need A MANAGER VoiP where I could:* Manage my providers A-Z Termination.* Manage DID&quot;s.* Manage Customers, each with its own instance DB.* Manage PBX for these clients.....I think what we need is FreePBX + A2Billing on ASTERIX multitenant.If I answer in Spanish will be very grateful.Best regards.

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  • £50 GBP/hr Jan 9, 2014

    Assist me in setting up an in house Asterisk server for our telecoms.

  • $3000 USD Dec 17, 2013

    Project Description: Project Description: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 cards3. Number of Server B can be unlimited4. 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)6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billing cdr,etc.Specific Need: 1. Server A to convert all calls in g723.1 codec and server B to receive all calls in g723.1 2. we need some kind of bandwidth compression system( upto 60-80% than usual SIP calls in g723.1 codec), from Server A to Server B. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for terminationPls let us know for further information necessery to get the project done. This project actually similar to a provider http://syncswitch.com offering and we we want to provide the same kinda services to our termination provider or tht meant named Server B in our given specifications above. suggest us required Server configurations n OS for Server A to start the projects.Attached some screen shots of another similar project.

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  • $50 CAD/hr Dec 12, 2013

    Hi esamuels,I was just looking at your profile and found that you are the person whom i&quot;ve been looking for for the past month, I am in a startup company with a partner and we are willing to do something that we are sure to be successful with, it involves kamailio and freeswich along with redundant servers, I wish to discuss the project with you and since we are a startup, it might also interest you that we are willing to share a part of the company with you. Please contact me if you are interested and i&quot;ll fill you in onto what exactly it is we would like to do. You can check out our site as well http://intelegent.caThank you in advance,Arsen

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  • $1250 USD Dec 7, 2013

    persona con cocnocimiento en apllicaciones para asterisk con reconocimineto de voz,si usted posee esos conocimiento envieme su propuesta por favor

  • $50 USD/hr Dec 3, 2013

    Have you setup BlueBox PHP before?? You profile mentions BlueBox but i don&quot;t know if it is the same thing

  • $150 USD Aug 10, 2013

    Hi, Im Mojtaba. I am a developer of AsteriskPBX. Could you help me how can i do in this site and how can i get projects in AsteriskPBX?nThanks very much.nwith regards.Mojtaba

  • $500 USD Jul 14, 2013

    Project Description: Project Description: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 serverServer B = Asterisk Client serverExplanation 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 mintB. Fedora desktop distributionC. Centos 5.8 or 67. 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 modeC. Tnic static and dynamic mode8. 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 asteriskconfigure IAX trunking, so we can measure the BW compression making the SIP-&gt; 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&amp;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 trunkshere 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 ...also price can be negotiate by talking...

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  • $300 USD Jul 9, 2013

    Customize an asterisk and a2billing installation for pinless, online signup and payment

  • $475 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 &quot;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 &quot;Want to add to the queue &quot; or something ?Calls are recorded all time.

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    Errol Samuels does not have any open projects.
    Errol Samuels does not have any work in progress.
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Portfolio

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Résumé

Experience

Carrier Services

Jun 2006 - Present (8 years)

Carisma Telecom Inc.

Setup interconnections with other Carriers using all types of Softswitches and Gateways for example Asterisk, FreeSWITCH, Kamailio, Opensips, Nextone, Quintum, VoipSWITCH etc.

VoIP Consultant

Apr 2004 - Present (10 years)

Self-employed Freelancer

*** ASTERISK | A2BILLING ***<br />Installation, Configuration and Maintenance, Calling Card setup for pin and pinless dialing, Advanced Wholesale setup for multiple IP Address authentication, Provide advanced A2Billing Customer GUI customization, Optimize and Secure Asterisk and A2Billing Installations, A2Billing and FreePBX / Elastix realtime Integration, A2Billing Ratecard Integrations<br /><br />*** FREESWITCH | FUSIONPBX | ASTPP ***<br />Install and Configure FreeSWITCH with FusionPBX Multitenant

Education

B.Sc. (Hons) Computing

University of Bradford

2006-2009