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    8,611 free sip flash media server робіт знайдено, ціни вказані в USD

    Що потрібно… «Розвернути» на сервері ASTERISK Налаштувати транк sip gsm +transfer Запис розмов та фіксацію номерів трансферних Адмін панель Панель доступу user Надаштування GSM шлюзів

    $576 (Avg Bid)
    $576 Сер. заявка
    5 заявки
    Goautodial specific functions 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...any number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load balancing on 3 trunks (sip provider). When dialing on lists, send one call to one trunk, next to the second trunk and next to third trunk and so on... 4) Failover : after dialing a number on a trunk, if

    $262 (Avg Bid)
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    ...to launch our new product to the world Naturally flavoured Coffee beans. Imagine its early morning your somewhere between being asleep and being awake and you have that first sip of the freshest, best quality, real Arabica coffee with a hint of rich Caribbean vanilla rum and tropical spices or smooth almond marzipan. Ahhh absolute haven, aromatherapy

    $250 (Avg Bid)
    Гарантований Прихований Топ-конкурс
    $250
    5 робіт
    Digital logo 5 дні(-в) left

    I'd like to create a logo for my website and work as an author and speaker on personal development. My initial idea is a play on my initials (SIP) and I've attached a sketch of versions that I have used. Would like someone to play around with that and come up with something professional/digital.

    $62 (Avg Bid)
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    56 заявки

    We have setup bitrix24 onpremise and we have asterisk-vici installed. We want to use SIP connector. Any one with experience in setting up onpremise Bitrix24 and telephony with SIP connector can respond.

    $578 (Avg Bid)
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    3 заявки

    ...have a SIP client on my network, its a Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX server has

    $143 (Avg Bid)
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    ...kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC.mydomain.com-onnet_1-b2b_1' with owner SIP/telsam_ip-00003486 in place (Method: BYE). Rescheduling destruction for 10000 ms. if you interested in I can share you more information. and budget is negotiable fixed or hourly

    $100 (Avg Bid)
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    OpenSIPS Expert Needed 4 дні(-в) left

    We need an expert in OpenSIPS that can change the SIP response of 200 ok to something else or delay it full details will be given later.

    $325 (Avg Bid)
    $325 Сер. заявка
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    Hospitality Mobile App 4 дні(-в) left

    ...allow only Checked In guests to sign in using surname and Room number. once signed in also allow guest to connect to the Hotel WIFI also within the APP we will need a build SIP/VOIP on the App which will interface with hotel PABX and the app will become the guest extension as long as his signed in and for the duration of his stay. This app will need

    $2471 (Avg Bid)
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    WebRTC / SIP expert 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    The client is using astrix server. They will give us a vpn connection to their network And we need to setup WebRTC client to talk to astrix server. We need to use open source functionality working which it can't currently get a function to work within a type. [увійдіть, щоб побачити URL]

    $20 / hr (Avg Bid)
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    VOIP SIP CALLING SYSTEM 3 дні(-в) left

    Want to start a VOIP/calling card solution ,need to install the necessary software (asterix,freepbx,a2billing etc).Advice the best solution

    $274 (Avg Bid)
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    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

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    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [увійдіть, щоб побачити URL], [увійдіть, щоб побачити URL], Debian ...Plans, PBX Extensions, I/O Trunks, Voicemail, DISA, IVR & local number display on the go, Call Forwarding, Call waiting, Linphone setup and instructions, call recording, setup sip trunks. for 4 - 6 lines. PHP or LUA scripting, VOIP, interconnect

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    Trophy icon Logo design for "Livingston Depot" 1 день left

    "Livingston Depot" is the Name we need a logo for.. - "Sip & Shop Station" is the Tag Line This is a boutique cafe that is located in Livingston, Tennessee. This shop offers local artists, hand crafted items for sale, with a small cafe offering beverages, and lunch items. The keyword inspirations are elegance, down home, boutique, station, depot

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    Asterisk Expert 1 день left

    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support

    $17 / hr (Avg Bid)
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    3CX Gateway for Skype 20 годин(-и) left

    I need a call with multi skype user via SIP. and i saw 3cx gateway for skype program. this product is no longer available for download nor maintained. [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] The 3cx gateway for skype is running only windows 7. and needed skype old version

    $29 (Avg Bid)
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    must be connect multi Skype user to a SIP-PBX. and call via skype to pstn or mobile number. multi skype user can running on asterisk machine. extensions are calling via each skype-account

    $30 (Avg Bid)
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    1 заявки

    We need an implementation of the services mentioned in the subject. Thanks.

    $1495 (Avg Bid)
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    ejabberd configuration Закінчився left

    I need help you configure my ejabberd server with sylkserver for translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

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    We have a multi-tenant PBX system and need a simple SIP phone provisioning tool written in Javascript Node.JS, using Angular or React, running on a MEAN/MERN stack. This will not be integrated with the PBX, however our B/OSS system will add, suspend, and delete tenants and extensions to this new system you will create via a REST API. A commercial

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    VoIP server for Termination Закінчився left

    Setup a VoIP server for termination the requirements are: 1. SIP connections to gsm gateways through VPN is required 2. Billing software 3. Load balance traffic to gateways 4. setup connections with 3 clients

    $238 (Avg Bid)
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    ...Financial Planning ;Invest online;Testimonials ;Blogs; Careers ;Contact us With A live Chat options & watsapp messenger Few Algorithms need to be used –like calculator for – sip;retirement;marriage;child education etc...

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    I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call

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    I have already built app on play store called "Loan Emi Calculator". Please review the app befo...Calculator". Please review the app before you bid, here is a store link.. [увійдіть, щоб побачити URL] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

    $158 (Avg Bid)
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    I have an existing Shoretel system with a PRI switch. Using the SETU VTEP device I want to replace my existing PRIs with SIP trunking by using the SETU VTEP as a gateway. I need help programming the device to integrate with my existing Shoretel system.

    $200 (Avg Bid)
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    Asterisk Voice IVR setup Закінчився left

    1) Set up Asterisk. You will get a machine with Linux OS of your choice. 2) Install UniMRCP and UniMRCP Kaldi Plugin. Configure SIP line for inbound calls. We will provide the URI to configure in the plugin. 3) Configure an Inbound call plan that does the following for inbound calls: (a) Prompt the caller to punch in a 4 digit code. (b) Post

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    convert cisco ip phone to sip for working in yeastar s100

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    Need to Configure 2TATA Sip Trunk line on my freepbx via remote desktop like TeamViewer. FreePBX hardware is assembled and installed freepbx.

    $18 / hr (Avg Bid)
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    Sip client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in code, over GPIO inputs trigger sip client start call. (Gpio inputs must configurable in code) Sip client must have

    $245 (Avg Bid)
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    Hi, I'm looking for someone to connect a twilio number that we currently have to a sip endpoint thats already configured for voice. We are using the X-lite Voip client and would like to get SMS functionality enabled on it.

    $39 (Avg Bid)
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    SIP CLIENT FOR RASPBERRY PI3 this software to be developed for Linux using Python. SIP client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in the code, over GPIO inputs we

    $373 (Avg Bid)
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    5 заявки

    Looking for someone knowledgeable on freepbX and mobile application development to create a VOIP application for customers service . The reference app is named as "smart line" You can check the app's features . I need the exactly same features with some new designs .

    $6 - $11 / hr
    Прихований
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    ...the other teeth drift and block out the permanent teeth. So want to increase awareness. Want parents to know what not to do. Need a catchy animation to launch as mass media. Need people to see the horrific effects of decay - shock. Want message to cut across. Want people to know that ECC can be totally eradicated. Animation I feel would be

    $14 (Avg Bid)
    Гарантований
    $14
    1 робіт
    Set up SylkServer Закінчився left

    Set up sylkserver to translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

    $21 / hr (Avg Bid)
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    Add SIP Trunk feature to frontend Закінчився left

    Add the ability to enter sip trunk credentials to asterisk

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    1 заявки

    I am looking for a senior architect/developer to design and develop gsm telecom solutions based on ss7 or diameter or sip communication protocols. The scope will be discussed later, the pay will be very good

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    Write An Erotic Short Story3 Закінчився left

    ...writer to write an original story featuring a seductive buxom woman who's big unwieldy breasts contain a milk that will wickedly shrink the size of any man who drinks it. One sip is enough to make him a midget and too much will send him down to bug size or smaller. Find a way to work this weird concept into a sexy short story with a full plot and I'll

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    Trophy icon Need a logo for a company beverage Закінчився left

    Hello, I need a logo for a beverage company. The name is "Soufrière Hills" The tag line is “every sip makes you family and comforts your soul.” Please use the following hex codes only: #549127 #10375d #edc857 Use all three or two colors in any combination. Please be as creative as possible. This logo is for a beverage that is from the island of

    $30 (Avg Bid)
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    ...60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of the 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 B 2. Server...

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    sip to tapi tsp file Закінчився left

    ...simple sip to tapi tsp app with simple features like sip registration like an sip client no voice or codec are needed the incoming call must only transmitted to tapi tsp driver with caller in number the out bound calls will be managed with action url so from any application how supports tapi functionality can dial out by using the tapi via sip to his

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    Hello, we are selling fasteners in germany. We give you a list of 1...to 3pm GMT. Also part time is possible. You need to speak very good and clear english. Please bid for 1000 Calls/contacts. I will provide a phone account to use. Skype or a SIP Account, so that you dont have costs. When you are good, this can be be a project for much more contacts.

    $590 (Avg Bid)
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    SIP intercom broadcast Закінчився left

    SIP intercom broadcast software View online status View the status of the call Setting up a voice recording Call management Single-key talks, multi-party meetings Surveillance, monitoring, shouting and deterrence Arbitrary Paging and Interphone Recording Talk queuing, strong

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    Project for mula9211 Закінчився left

    ...service to companies and resellers. It is based using our own heavily customised linux distro. We also want to experiment with jingle (xmpp client) and SIP integration using a combi of sylk server and opensips. IMO a real challenge for experienced engineers. I can be reached by mail at max@axeos.com. We can start with some small projects to get to

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    sip trunk configuration with 3cx Закінчився left

    i have installed 3cx pbx and every thing is good then i install sip trunk and configure the did/dod put when call my office number sometimes it ring and sometimes the number are busy need help

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    Project for Anton F. Закінчився left

    Hi Dear, We have dial plan for inblond calls . Mean calls which land to server its route to IVR or IP only DID which are uploded into portal . mean did numbers which are in system only that can route with ivr or ip . Now i want make change without disturbing other dialplan just only one prefix i route to call center or IVR so all calls of that

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    install asterisk on Puppy Linux Закінчився left

    you have to install asterisk 1.8 on Puppy Linux make sure asterisk work with sip & iax both also wireguard & TeamViewer you can download OS from here [увійдіть, щоб побачити URL] also if you want you can use Ubuntu or Centos you need to make iso and iso size must be under 500MB i can pay up to 100USD do not ask for more

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    ...will reside on Tizen the function of the app is a bridge between sip client and Phone Native dialer The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Tizen. Must run on background

    $1267 (Avg Bid)
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    SIP Softphone for Windows and Mobiles Закінчився left

    We are new small VOIP S...VOIP Services provider and We need a SIP Voip Softphone App for mobiles mainly, eventually for windows too. The App would be similar to Zoiper, Bria, SIPGo, SessionTalk, etc. but customized with our Logo, our proxy domain and most important the capability to use all video and audio codecs including the now free licensed G729.

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    ...develop applications so that the total solution helps the customer to a higher level of service to their customers. Teleoffice has developed a structure with database, web server and REST api that can be accessed via apps and web interfaces. To this environment, we now want to find a resource to extend our development team with the following qualifications:

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    Setup correctly Jitsi platform Закінчився left

    ...can setup correctly Jitsi videobridge with Need closed rooms, only accept rooms offered in XMPP XMPP prosody server to be able to handle room authentication + concurrent usage per room previously setup Jigasi dialin number (At the moment SIp registers and enters room but no voice never) Online translation and online transciption Customize mobile app

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