Web call asterisk website asp net jobs

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    10,860 web call asterisk website asp net jobs found, pricing in USD

    i want to install Asterisk in my company including integration

    $122 (Avg Bid)
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    Queremos conectar 2 camaras a dos softphones zoipers via asterisk sólo audio no video.

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    need help fixing a vicidial server when the agent logs in the initial call doesn't come over for 45-60 seconds well after the interface times out

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    ...that is similar. All the pricing and details can be found on the website: [login to view URL] On the flyer I need date, time, location, ticket pricing (with an asterisk that the prices are early bird pricing) I also need room to add Sponsor Logos which I will add later. The website must be listed as well. The tickets should have all that information

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    ... Profile picture upload issue 2. First Name and Last Name in 2 separate boxes should be instead of full name. 3. Fields validation issue (marked by red asterisk) 4. Target Job location (the list should include drop-down menu with Qatar cities and zones, multiple choices option should be available) 5. Job Industry, Career

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    ...Is Coding Asterisk Virtual Server. 1: Code Asterisk Server And Configure To Be Used In A Local Area Network With A. Hard Phones B. Soft phones Application Installed In Android Phones IPhone Phones N:B The Coder Will Recommend The Hardwares A. Asterisk Hardware Server

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    Hi Munir H., I want to build call center based on asterisk with some extra features let us discuss the scope and then decide the cost for now i will make it just 10 till we agree on cost

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    Need install asterisk and sip server on virtualbox for receive calls directly to computer and if 1st line is busy automatically forward to second free line with call recording and without monthly fee

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    Asterisk Queue Setup 4 days left
    VERIFIED

    I have a production asterisk installation running on my server. I have a requirement. I want to setup a queue such that Agents and end users can use queue using their mobile phones. Lets Say, their are 3 agents Agent 1: Mobile : +91-XXXXXXXXX1 Agent 2: Mobile : +91-XXXXXXXXX2 Agent 3: Mobile : +91-XXXXXXXXX3 Lets there are 5 users who will dial

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

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    ...'moodle'): [login to view URL] 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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    We're looking for a 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

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    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

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    ...rejects the call. So we must continuously be able to see all of the information, and there is a site with further information/more details 08:12:05 the call comes into NordicCall (ID.nr. XX ) via the number 70209404, the caller is the number 80808080. 08:12:05 The call to NordicCall (ID.nr. XX ) gets the welcome speak played 08:12:35 The call to NordicCall

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    ...type of call flow as native dialing on device. The customer would dial a number in the app and hit send. The app would send to the backend server (you design) the backend server would query a Microsoft SQL database and retrieve the customers office number, cellphone number and, if they are valid in system. If so the backend server would send call set up

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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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    ...with in 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

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    ...implement a click to call system on my website. I've a list of tech support persons and each one will need to have his unique click to call button. The registered client will after buying credits (already working) will choose the tech person and will ask to be called to his phone number (phone number, not webrtc). Before the call, the client will choose

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    ...Admin - login access Agent - login access Customer - login access Admin - Features Add, edit, delete Customer accounts Assign customer phone numbers (integrated with Asterisk ami to enable screenpop) Add, edit, delete Employees (no login) Add Customer business info (screenpop, location info) Add screenpop Forms Add changes reason text box.

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    ...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, we need to figure out what voip server and back end sip or trunk providers are

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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 check the server, it takes ...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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    Hello, I need someone to install a working version of GoAutoDial v4. The instructions are here: [login to view URL] 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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    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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    ...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 added) Proxmox cluster with CEPH cluster FreeNAS Website monitoring (to be added) Most host are installed and working

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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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    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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    One Server with multiple disks managed through KVM - qcow2. All OS are Ubuntu 18.04. To access the server it takes a late Teamviewer. R...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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    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 pressing some key. The info sent is normally a website URL. I have this working using vitelity sms service, but I need more robust sms handling. Mainly to handle 'stop' requests and etc. I'm using the PBX in a Flash Incredible 13.13 platform

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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 a...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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    Server with multiple disks managed through KVM. All OS are Ubuntu 18.04. To access the server it takes a late Teamviewer. Requirements: Nginx rev. Prox...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 ([login to view URL]). Normally when the code is executed, call is setup but when using same code with express, there is error

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    Hi Ambiorix R., I 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/[login to view URL] files using php from csv extracts

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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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