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    17,653 fas call sip answer supervision travaux trouvés au tarif de EUR

    hi, I need a C++ or .... programmer that is familiar with SIP protocole and flasher player. what i need is as following. I have a SIP switch that wants to run live video to user (PC). so flash player is the choice of player in the pc. I alreay have the SIP part done. and a code drop for integration into the flash player. I need

    €744 (Avg Bid)
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    SIP/VOIP in Java S'est terminé left

    I want a VOIP /SIP Phone in [se connecter pour voir l'URL] have PC-PC and PC-Phone Calls.

    €26 - €88
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    SIP Client S'est terminé left

    a sip clinet like eyebeam [se connecter pour voir l'URL] which has got an additional chat function ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows (depending on the nature

    €88 - €438
    €88 - €438
    0 offres
    Two E-Book Covers S'est terminé left

    Create 2 E-Book Covers.? Titles Are: Enjoying Your Gluten Free Lifestyle and 'Bar Cocktails For Celiacs', Enjoying Life One Sip At A Time Each cover should be elegant in appearance Provide product in Industry Standard Format in USA.? Usually .PDF.? Provide each as a separate document.? Thank you. ## Deliverables Rent A

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    ...download the trial version so that you can see what functionality and interface looks like. Also note how you can use AnswerFiles so that you can save each set of question/answer info and easily open it on another computer. We need this new program to do everything that AnswerTool does with these upgrades: SEE ATTACHED WORD DOCUMENT FOR THE UPGRADES

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    26549 VOIP Escrow Software S'est terminé left

    I need a clone of Escrow System for VOIP Business please have a look a...[se connecter pour voir l'URL] [se connecter pour voir l'URL] we need a compelte system for running a Escrow company for VOIP Business just a hint it involves steps like billing, softswitches, sip to h323, h323 to sip, codec conversions and see on, plus live billing, brokered transactions and much m...

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    Hi: We are looking for a party who can design & implement for us a low cost voip carrier for international calls. If...party components, etc. unless all copyright ramifications are explained AND AGREED TO by the buyer on the site per the coder's Seller Legal Agreement). ## Platform voip, sip, telecom, telecommunications, low cost carrier

    €5034 (Avg Bid)
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    SIP softphone S'est terminé left

    SIP softphone that can be use make VoIP phone call from PC to PC supporting multiple session. the user agent should be able to register with and proxy server,should support NAT traversal, multiple voice codec, direct Address book import.

    €940 (Avg Bid)
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    27368 SIP softphone S'est terminé left

    SIP softphone that can be use make VoIP phone call from PC to PC supporting multiple session. the user agent should be able to register with and proxy server,should support NAT traversal, multiple voice codec, direct Address book import.

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    ...on CDYNE. The site would need to use a SQL Server 2000 db or higher (2005 is ok). <[se connecter pour voir l'URL]> You would need to be able to call that web service and make calls based off what the user wants to notify someone with. It also needs to have billing tables in the database. We are only going to accept Credit

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    ...Rent-A-Coder's web site. (as the chat is logged). This is a summary of what the questions consit of during the two days in total. There are four parts. **Part one**: answer fifteen questions in the short text regarding to basic concepts on the syntax, property or feature of XML, XML Schema/DTD, SAX/DOM, RDF, RDF Schema and OWL, etc. **Part

    €26 - €4379
    €26 - €4379
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    28362 VOIP/SIP Development S'est terminé left

    We need someone who has worked with SIP technology, specifically the Cisco Call Manager Software. Additionally we need a SIP Sofy Phone Developed in Active X for a user community [se connecter pour voir l'URL] Please respond by also sending a message to sbeck at beckventures dot com Cicso CCIE certification would be a plus

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    We need someone who has worked with SIP technology, specifically the Cisco Call Mamger Software. We have the licenses and software allready loaded on the servers and need someone to manage and make modifications.

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    ...programmer with good IQ to understand others to do small changes to an existing site. i will pay USD10 to USD25 per small task. You MUST answer each of the following :- 1) Can you work under minimal supervision? This means that you must test all the pages before asking me to test. I cannot comment on each and every tiny matters of the site. 2) Can

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    We want to set up Asterisk - VOIP (for multiple things - call center - pbx - calling card - fax - billing - ERP / CRM Sugar etc etc) along with rest of the “Open Source?? only modules. Tell us your experience and ability to undertake full or part of the project. What have you done in the past and what can you do now. This will be installed

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    You are placing bids for typical translations as per the att...during office hours and has unlimited local phone calling. Can manage and co-ordinate local installation teams, Understands process flow, follow up and can work without supervision. Adaptable to be Girl-Friday on Monday and Local Executive the next day. Bonus if you appreciate good humour.

    €26 - €88
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    29388 Small changes to site S'est terminé left

    ...following qualities as i am fed up of working with some programmers who really need a lot of supervision in their work. the programmer will be required to do some changes to the sites i have. Please answer the following :- 1) Can you work under minimal supervision? This means that you must test all the pages before asking me to test. I cannot comment on

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    Mikrotik ...Support - Queue Management - Hotspot - ABILITY TO TRACE NETWORKING ISSUES AND FIND LOST PACKETS - Proxy Setup - Security - DHCP/VPN/DNS - Security Alert and Locks - SIP Support candidate MUST be able QUICKLY read TCP dumps to determine network issues to isolate networking problems. Contact us for more detailed specifications.

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    Asterisk PBX Configuration S'est terminé left

    ...Channel Bank Configuration 3) SIP FXO Gateways - Welltech 4) Ability to read logs to determine problems 5) implement controlled dial plans Candidate MUST know HOW to configure Asterisk as a PREMISE based PBX. Minimum system configuration will be as follows: 24 FXS (Channel Bank) 4 or more FXO (Digium FXO or SIP FXO Gateway) We will consider

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    ...some more questions to answer and I'm very very busy. I need a native english speaker to answer the questions remained(attached, about 5-7 questions) and review my answers and make suggestions(I will send my answers after bid acceptance, in total there are about 10-13 questions). It'd be great if you could send a sample answer along with your bid

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    ...looking for someone that can almost do it all. Here's the basics of what I need: - Support: email and ticket system based support for a few sites - Sales: again, 2-3 sites to answer pre-sales questions - Marketing: someone who can write some articles, post articles, write press releases, post press releases, get active in forums, write some FAQ's and tutorials

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    ...(voicemail, caller ID, call forwarding, 3 way calling, etc.) via a web-based GUI. Basically all the features that are available with most ITSP, should be available to our customers, and it is your job to set it up correctly. All possible compression codecs should be installed as well. All outgoing calls should be via one of our SIP / IAX providers and

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    31151 SER Billing S'est terminé left

    ...provide callshops solutions for callshops all over the world. Now we want to expand the business and need a programmer to make 2 GPL softwares work together The Softwares are the Sip Express Rourter from [se connecter pour voir l'URL] and Starshop. Starshop is today built to work with Asterisk from Digium but we want it changed to work with SER instead. The documentation on the

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    SIP Softphone Functions: Make & receive calls, hold, transfer, do not disturb, auto answer, record calls, support of STUN server, registrar, proxy, should support up to 10 SIP servers / SIP lines, phone & line level meter, echo cancellation, touch-tones, adaptive jitter buffer, automatic gain control (if possible), skip silence. Codecs at least: G.711

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    H323 Dialer for pc2phone S'est terminé left

    ...g723.1 and g729 codec support 2. Balance display from a php page where user is passed as a parameter . ex: [[se connecter pour voir l'URL]][1] 3. Rates and Call log page which will open up a webpage with a userid. 4. Multiple instances , the user will be able to run 4 or more instances of the same program , it will be used to handle

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    ...First Part: The flash will ask user for basic SIP username, password, stun/nat and codec to use (The SIP address server is hardcoded and shall be changeable easily in the .fla or instruction on how to change it). Once connected, the user can dial the number he would like to connect to. Hangup, redial and call button should be in too. Second Part:

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    ...First Part: The flash will ask user for basic SIP username, password, stun/nat and codec to use (The SIP address server is hardcoded and shall be changeable easily in the .fla or instruction on how to change it). Once connected, the user can dial the number he would like to connect to. Hangup, redial and call button should be in too. Second Part:

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    ...First Part: The flash will ask user for basic SIP username, password, stun/nat and codec to use (The SIP address server is hardcoded and shall be changeable easily in the .fla or instruction on how to change it). Once connected, the user can dial the number he would like to connect to. Hangup, redial and call button should be in too. Second Part:

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    SIP Flash application S'est terminé left

    ...First Part: The flash will ask user for basic SIP username, password, stun/nat and codec to use (The SIP address server is hardcoded and shall be changeable easily in the .fla or instruction on how to change it). Once connected, the user can dial the number he would like to connect to. Hangup, redial and call button should be in too. Second Part:

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    €88 - €263
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    VOIP Sip DLL S'est terminé left

    Require a Non Microsoft RTC based SIP Phone with as many options of the following as possible Telephony Features ---------------------- Record Audio Mix Acoustic Echo Cancellation [AEC] Message Waiting Indicator Audio Tuning Wizard Touch-tones [RFC 2833, in-band, out of band & INFO DTMF] Automatic Gain Control Audio Concealment

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    I will create a feedback form in html. I need some one to program in php to be able to collect the info from the form and 6 images that may be uploaded with the form to an email address or to an admin panel if that is necessary due to file size of images. Images probably regular 1024 - 1280. This form will have required fields. I also need a zip code finder. I will provide an excel sheet with...

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    ...(Masters degree) - Experience in web analytics - Some accounting/actuarial knowledge - Programming and database whiz - Can work from home, independently and without supervision - highly organized and proactive - ambitious You are an extremely fast learner and have a high IQ. You have a history of starting long-term projects and completing

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    ...is intended to serve as a QoS and traffic shaping layer between a SIP proxy registrar and a satellite SIP proxy forwarder installation. Each server will use the open protocol MPLS stack provided by James Leu and Ramon Casellas on Sourgeforge. A DiffServ component should ‘gold’ route SIP traffic to its intended network path. Specific Goals: 1. The

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    ...is intended to serve as a QoS and traffic shaping layer between a SIP proxy registrar and a satellite SIP proxy forwarder installation. Each server will use the open protocol MPLS stack provided by James Leu and Ramon Casellas on Sourgeforge. A DiffServ component should ‘gold’ route SIP traffic to its intended network path. Coder Competencies 1.

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    33074 sub domain script S'est terminé left

    ...there. These sites will only include ones with redirection service. (Would like large list of preloaded sites if possible, I have some. 3. To run automatically without supervision. 4. It enters the pre-determined information and clicks to the next page until registration is complete. 5. Software continues until all sub-domain names are registered

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    ...product at the repair center. 3. if a costumer ordre a product we want to do a invoice and a packing note directly from the prog. 4. a sip phone client to make recieve and transfer voip calls. 5. a mail client to answer req. by mail where we can reuse old answers. and the short explanation: i want our repair, service, hotline, helpdesk, and sales to

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    ...features our products has to offer. **_FORUM_**??" maybe a discussion forum to talk about products, drivers etc. From the site we want a phone where costumers can call us direct thru our sip voip phone ## Deliverables Rent A Coder requirements notice: As originally posted, this bid request does not have complete details. Should a dispute arise and

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    This project is very small scale and requires much more explanations than actual coding effort. It is for proficient sip and asterisk developer willing to spend a few hours hacking the asterisk sources. Please read the attached ms-word file for full requirements/specs/description Here is only a brief summary: Note ??" Although the description is long

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    Voip SIP Fax S'est terminé left

    ...account, and they gave me some strings to put into the SIP software to make it work. the SIP software is x-lite ([se connecter pour voir l'URL]) what is missing to this software is the FAX sendig support. so i make 2 different bid requests: 1) DO YOU KNOW a software that let me send faxes using my own voip SIP account? if yes please tell me, and bid for $5. It should

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    ...Asterisk configuration. This may include answering questions, referencing examples, providing example code, or logging in to our phone server and making changes under the supervision of a team member. We also will be creating custom built asterisk scheduling tools and will be looking for help and programming work with this as well. This job will most

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    ...features our products has to offer. **_FORUM_**??" maybe a discussion forum to talk about products, drivers etc. From the site we want a phone where costumers can call us direct thru our sip voip phone ## Deliverables Rent A Coder requirements notice: As originally posted, this bid request does not have complete details. Should a dispute arise and

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    Question/answer tutorial S'est terminé left

    We need a demo Question and answer tutorial. There will also be a graphical represenation of the question aimed at giving the user a hint to answer to the question correctly. There will be a feed back to the user whether the answer is right or wrong. For this Demo, there will be only 3 questions and 3 possible answers for each question (2

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    ...read this project thoroughly, does not have yahoo IM, and does not have a website (portfolio) themselves. The coder who accepts this project and does it well, without supervision, will recieve more work from me on a long term/continuing basis WITH bonuses. Good luck to all and Happy bidding! ## Deliverables 1) Install Hamweather to existing

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    €88 - €175
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    ...to complete the projects. You will work an 8 hour day, five days per week. The other four hours per day you may schedule at your convenience. We will provide you with a VOIP/SIP telephone at our expense, which requires that you have Windows XP as your operating system. You should be proficient in Microsoft Word, Excel, PowerPoint, Quark Express (or some

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    ...Description is below. I WILL NOT accept ANY bids who have no ratings or have not read this project request thoroughly. The coder who does this project well and without supervision will recieve a bonus, as well as, long term/continuing work from me. Thank you all and happy bidding! ## Deliverables 1) Install *Easy PHPCalendar* to existing site

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    Asterisk/Digium Converter S'est terminé left

    Searching for a Linux server administrator experienced with the open source PBX software Asterisk by Digium. You will need to install and configure Asteri...experienced with the open source PBX software Asterisk by Digium. You will need to install and configure Asterisk on the server for us remotely. To do conversion analog line into SIP Trunks.

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    ...can do it – Intimate knowledge of SIP stack and C/C++ and previous experience of creating customized SIP applications or developing asterisk patches for very specific purposes. What needs to be done - Alter the behavior of SIP stack. How – Produce a patch against asterisk sources or develop a stand alone SIP router with dedicated function.

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    ...Carrier 'SIP' asterisk 1.2.4 'SIP' sipura-spa3000 'FXO'PSTN OR Carrier 'SIP' asterisk 1.2.4 'SIP' grandstream ht488 'FXO'PSTN OR Carrier 'SIP' asterisk 1.2.4 'SIP' [Any other single SIP FXO port] 'FXO'PSTN Problem: FAS on asterisk ' spa30...

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    ...application for making outbound telephone calls from a single computer using multiple VoIP SIP (RFC 3261) telephone lines. ## Deliverables I am looking for a basic contact manager (Name, Phone Number, Address, City, State, ,email, Notes section, TimeZone, Call Outcome) with the ability import data from .csv files. It would also be great if the contact

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