Simple asterisk project php工作

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    10,851 simple asterisk project php 搜到的工作,价格货币为 USD

    主要指导elastix asterisk 的安装配置工作,熟悉Asterisk的模块(包括体系结构,配置文件,Log日志,History),能重新打包ISO文件

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    We are using the Asterisk PBX With a Linphone SIP client in a Linux environment operating on the Olimex A20 and PINE64 and are experiencing very high echo. We understand Linphone uses the Speex Echo Canceller. We either do not know how to adjust the echo canceller or we need to substitute it for another excellent echo canceller. We would like someone

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    ...(inclusive) and store these into an array.  Produce a chart EXACTLY like the one below that indicates how many values fell in the range 1 to 10, 11 to 20, and so on. Print one asterisk for each value entered. Notice the spacing for everything. Range # Found Chart --------- ---------- -------------------------------------------

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    ...ou seja faça ligações através de servidores PABX's. O principal ponto do projeto é que o webphone deve efetuar ligações através de vários servidores PABX do mercado, ex:. Asterisk, FreeSwitch, intelbras e etc... Caso seja necessário uso de WebRTC Servers, fica a escolha do desenvolvedor. "Podemos analisar out...

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    Configure Asterisk URA with PHP integration

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    Telefonia IP 2 天 left
    已验证

    Necesito generar una maquina virtual con la telefonia IP asterisk, elastick u otro servidor.

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    ...'moodle'): [登录来查看链接] Required fields are next, where are same fields obtained from the original report, plus (marked with asterisk*) four fields that will be calculated using the same extracted data: - Date and time - First name / SurnameSort - Email - Grade item - Original grade - Revised grade -

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    I need a person who performs Vtiger CRM Integration with Asterisk

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    ...senior React Native/Redux developer to complete the final steps in a VoIP/Text Messaging mobile app. The mobile app uses PJSip to communicate with Asterisk, and interacts with a back end API created in PHP. Ideal candidate would have knowledge of React Native, Redux, and PJSip (Optional but definitely recommended). The final stages of the app are implementing

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    Asterisk web-based GUI 1 天 left
    已验证

    need an Asterisk developer to help on configure a GUI for my new VoIP infrastructure Task - design and configure my Asterisk Do not bid if you are not an experienced Asterisk developer

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    Hi , I am looking for someone who is able to configure ZRTP on freepbx ? and would like to know if all features work with this protocol? I read some articles said that call recording is not possible with ZRTP. Does it work with all other protocols such as, TLS, UDP and TCP? Let me know if you are interested in this kind of work and let`s discuss the duration and price.

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    calls detail site - asterisk 11 小时 left
    已验证

    We want a site where we easily can see the calls that come in and what happens to them. An example could be A customer calls 70209404 (NordicCall), the customer is in the queue for 2 minutes because every agent apart from two are on DND, one agent is busy an the other agent rejects the call. So we must continuously be able to see all of the information, and there is a site with further informati...

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    ...and, if they are valid in system. If so the backend server would send call set up info (dial plan) to asterisk server (I have an asterisk Guy to work with). The app would then dial an 800 number that was returned to the app on call setup. Asterisk would match the customers CID and process the call as per the ad hock dial plan created for that call substituting

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    Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload

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    ...my budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls

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    Click to call 已经结束 left

    Hello. I need to implement a click to call system on my website. I've a list of tech su...can be billed 10€ for 10 minutes or 20€ for 20 minutes. Only after the site deductes the credits, the call will start. It needs to connect to some cheap voip server (3cx, asterisk, freepbx, etc). The support tech person cannot see the client phone number.

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    laravel expert 已经结束 left

    ...expert. Project: Telephone Answering Service Software Project Type: Software + API Integration Software Name: Switchboard Timeline: 2-3 weeks Phase: #1 User Types: Admin - login access Agent - login access Customer - login access Admin - Features Add, edit, delete Customer accounts Assign customer phone numbers (integrated with Asterisk ami

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    VOIP product 已经结束 left

    ...Sentel to see if you can determine the technology they are using for their back end provider (SIP or otherwise) and server details.(ie. Centos with WHM and cpanel running Asterisk) I already have some hosting options in mind and I prefer Centos with WHM and cpanel, running various services to accomodate the VOIP server and the website. Basically,

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    Integrate Issabel-Asterisk with Twillio Elastic Sip Trunking to records Calls from Twillio (through Flex)

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    One Server with multiple disks managed through KVM - qcow2. All disk OS are Ubuntu 18.04. Security is important. To chec...Security is important. To check the server, it takes a late Teamviewer. Requirements: Nginx rev. Proxy, Apache2, Certbot - SSL, SMTP server, HTTPS, PHP, MySQL, KVM - qcow2, LibreOffice, Ubuntu, Asterisk PBX for invoice info., etc.

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    Install GoAutoDial V4 已经结束 left

    Hello, I need someone to install a working version of GoAutoDial v4. The instructions are here: [登录来查看链接]://[登录来查看链接] I will provide the VPS required and SSH access. Requires: PHP, MYSQL, COMPOSER, ASTERISK knowledge.

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    Asterisk PBX application to perform the following. To setup a SIP call between two servers using a specific codec. Have one side playback audio files. These servers will be connected over a WANem emulated link to induce packet loss/jitter/delay etc and produce degraded audio. Have the other record the degraded audio and store it .

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    Sistema IVR 已经结束 left

    ...del lenguaje natural. Este sistema permitirá concertar citas con distintos establecimientos de forma rápida, fiable y segura. Necesitamos apoyo en todo lo relacionado con Asterisk....

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    a2billing asterisk expert 已经结束 left

    A2billing corrupted no sync with asterisk , unable to make outgoing calls although customer has credit

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    Migrate Queuemetrics 1.5.0 and asterisk 1.6 to lastest Queuemetrics version and asterisk (per best recommendations from Loway)

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    Need help with enabling join Q on a queuemetric 1.5.0 and asterisk 1.6. The idea is that agents can select the queue from queuemetrics agent area.

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    Zabbix fine tuning 已经结束 left

    ...working. Needs fine tuning. Host include: Windows 7 through server 2016 with active directory. Printers Xerox, Canon, HP Peplink routers, Balance and SOHO series FreePBX (asterisk running PJSIP) Ethernet switches Nortel/Avaya, Arista, 3com Microsoft SQL 2016 Standard with AG (to be added) Multitech rCell 100 Cellular router HP Servers with ILO (to be

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    Looking out for someone who has good hands on FusionPBX, we are looking out to move over to Fusion (Freeswitch) from (Asterisk). We are in Melbourne, Australia.

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    Asterisk Dialer 已经结束 left

    VTiger Installation , Click to call & Configuring of Cepstral TTS, configure scripts, Integrating payment gateway

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    i have asterisk Voip FreePBX installed on my server at home and i want to let it show the caller id on all of my tv i have chromecast on all of my tv and 2 of them are smart samsung tv tell me if you have solution for this

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    I have a number of phpagi scripts that work on my older system, but now that i'm moving to a new asterisk system, they fail to run. They all suffer I'm sure from the same problem, whatever it is. Old system uses PHP 5.3.10 And new system uses 5.3.3. I'm not sure what else you need to know, but ask away.

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    Project for Sergey S. 已经结束 left

    Добрый день ! Интересует консультация,или помощь в настройке уже работающей Asterisk, подробности могу отправить в скайпе или по почте. Заранее благодарен.

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    Project for Prostitov A. 已经结束 left

    Добрый день ! Интересует консультация,или помощь в настройке уже работающей Asterisk, подробности могу отправить в скайпе или по почте. Заранее благодарен.

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    Project for Igor G. 已经结束 left

    Добрый день ! Интересует консультация,или помощь в настройке уже работающей Asterisk, подробности могу отправить в скайпе или по почте. Заранее благодарен.

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    Template System 已经结束 left

    One Server with multiple disks managed through KVM - qcow2. All OS are Ubuntu 18.04. To access the server it takes a late...OS are Ubuntu 18.04. To access the server it takes a late Teamviewer. Requirements: Nginx rev. Proxy, Apache2, Certbot - SSL, SMTP servers, KVM - qcow2, MySQL, PHP, LibreOffice, Ubuntu, Asterisk PBX for invoice info., HTTPS, etc.

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    Asterisk devlopment for voice broadcasting and vicidial

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    Memindahkan database dan script yang telah terinstall di server centos 7 yaitu menggunakan asterisk. Software telah tercustom akan dipindahkan ke server baru. Diharapakan yang telah memahami asterisk dan freepbx

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    I want to add PAI to all outgoing calls from an asterisk to a provider. Calls are from other pbx and are received without PAI. Caller CID from uri must be send as PAI. The system is Elastix 2.5

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    I've created a dialplan that allows callers to receive information by sms when they call one of my numbers and press...using the PBX in a Flash Incredible 13.13 platform. I would like to try to nexmo first, then later add other providers. I'm sure there is more needed from me to do this project, but if you have done this before, you know the answers

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    Problemas con Asterisk 已经结束 left

    No puedo conectar mis gateway GSM con asterisk y tampoco puede escuchar ni tampoco mi audio en las llamdas, necesito un experto en Asterisk.

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    We are looking full stack developers to build a telephony system based on engine Asterisk or Freeswitch. We need to build the whole system including a client area and manager interface. Experience on VoIP/Telephony proyect is a must. Please contact us for more details.

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    Need an Asterisk/FreePBX guru to show me how to setup xlite and Microsip sip phones to register with FreePBX extension.

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    i need someone to create and IVR for me using Asterisk that will interact with mysql database for callers details and readout details form database using TTS (Hindi and English) there should be option to call http apis and also the call flow should be easy to modify i have previously done it using PHP api but never got time to finish this (i don't

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    Total installation with all required features for Blended call handling. Need features like IVR, Voicemail, Call transfer, Call broadcasting etc

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    I need a IVR system which build on Rasberrypi. System should be able to decode DTMF from customer and respond accordingly (system should be able to modify the respond). System should be capable to call some Http APIs . There will 3 level menus. Normal market available sim will be attached to the SIM Module and were we expecting to receive call.

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    Document Server 已经结束 left

    Server with multiple disks managed through KVM. All OS are Ubuntu 18.04. To access the server it takes a late Teamviewer. Requirements: Ngi...KVM. All OS are Ubuntu 18.04. To access the server it takes a late Teamviewer. Requirements: Nginx rev. Proxy, Apache, Certbot - SSL, SMTP servers, KVM, MySQL, PHP, LibreOffice, Ubuntu, Asterisk PBX, HTTPS, etc.

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    There is node js code which is calling ari client library ([登录来查看链接]). Normally when the code is executed, call is setup but when using same code with express, there is error

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    Project for Ambiorix R. 已经结束 left

    ...noticed your profile. I have a project to develop an asterisk dialplan. Native asterisk, not freepbx etc. Phase 1 is to write the dialplan and get working. Dialplan not that complicated. Macro to route inbound calls to dids in gateway. Phase 2 of the project will be to create the dialplan/[登录来查看链接] files using php from csv extracts from database

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    We have a new digium g100 gateway but can't config fine on issabel/asterisk pbx. Need support to finish/working PBX (inbound/outbound) on E1. The digium detect fine the Asterisk but not make calls/inbounds (trunk/routes). We think is tunning in around 2 hours or less. The support work on Teamviewer.

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    Pbx and asterisk integration vtiger 7.1

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