Dtmf asterisk dial sip jobs

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    23,107 dtmf asterisk dial sip jobs found, pricing in USD

    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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    ...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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    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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    Description Job Details: We are looking to hire operators that will dial into an adult chat line and speak with men for 8 hours a day using a high speed internet connection, a home computer and a head set with microphone and speaker. We will provide you with detailed instructions on how to connect to the chat line. Basic computer skills are required

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    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csi...Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header Compression support! within the app.

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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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    ...sex chat! Just casual conversations, talking to strangers, smalltalk, maybe finding new friends etc. For this purpose we need YOU! If you speak Finnish. Your job would be to dial in to the system and hang around in one of the public chatrooms. Since everything is really new, and there are not many callers yet, most of the time you'll listen to radio

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    Dear all, We have a urgent Vicidial system to se...information and agent disposition data into one outlook. That means initial upload of list information must converge with output of agent disposition to create one report 6) To create sip trunk for testing 7) To create sample lead list 8) Testing and commissioning 9) Agent and supervisor training

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    I have a watch and want to put a logo in with special features the design should be made in a specific sizes for manufacture

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    Its an cafe i want a logo for it

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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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    SIP dialer for Android and IOS. Should support most Android and IOS versions including IOS version 12. Features: Runs behind NAT Support Bluetooth Echo Cancellation Push notification Call history display Account balance display Address book integration Support UDP and TCP protocols Support users to Login or Signup Supported languages: English, Arabic

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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. I would like to try to n...

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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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    ...basic IVR. Only one level. 2. Setup of 2 profiles of users. a. Will be able to make local calls only b. Will be able to make local and external calls 3. Configuration of 2 SIP and DID services 4. Configuration of Conference rooms (phone bridges) 5. Configuration of VOIP VLANat the IPPBX and Phone level (network switch is already configured) 6. Configuration

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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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    Hello. I am looking for someone that can build a system that will deliver voicemail messages to external numbers without ringing the phone...This is also knowen as VoiceCasting. It's important that you understand that the phone numbers are NOT phone numbers or extensions within the PBX. Must have vast knowledge of SIP and PBX with FreePBX platform..

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    ...in vector format. No need for hands, but we do need the clock dial (numbers) to be on a separate layer. We do not need the clock to look like a clock - meaning we do not want hands (as we have a specific hand design we will add)... effectively, we need the background art and the numbers dial. That's it. No outer frame, no hands. • 15” x 15” artwork

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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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    I want voice engineer to do IP Telephony Cisco call manager installation and expressway server ,so that sip phone register over internet.

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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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    Hello, I have an article with about 50 pages. It talks about VOIP and TOR networks. I need to rephrase this article so it can not be recognized or even a...plagiarism checkers. All the article is in English and rephrasing should be in English too, I guess a technical background is needed. Specially talking about protocols Like RTP, SIP,UDP and VoIP.

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    ...ON WHO CREATE SMILES. If you know your parties extension, you may dial it at anytime. For Sales press 1, For support press 2, for Operator press 0 or stay on the line and someone will assist you momentarily. 2. Thank you for calling. If you know your party's extension, you may dial it at anytime or stay on the line and someone will assist you momentarily

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    ...from the employee, the name and description, a way to show its current status (Offline, in use, ringing...), a way to show the current connected number and one or more icons to dial him or pickup his ringing calls. To be hired, please provide some previous works images, your current knowledge and if you like to impress myself, a rendering of a possible

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    Pbx and asterisk integration vtiger 7.1

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    We are looking for customized FreePBX applications. looking forward to an expert in Vicidial & free PBX

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    ...add more options in the future, the latter will always be with additional payments and will be discussed in the same way as this project. Future of the application: (ASAP) SIP Softphone option Option to send and receive FAX using the application, the API is the same provided. This same application will be adapted in the future using this and other

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    hi dear i want trun gsm modem with chan_dongle in asterisk now my asterisk detected 2 gsm dongle i have but its need more configure for access calls and message sims can you do it for me?

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    I need to setup some params to my Asterisk like changing headers . i require that you know Asterisk or Issabel can have worked with it engough so you have understand of trunks etc.. need support for macximum 1 hour via teamviewer

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    ...and register to IPv6 sip servers 2.) IPv6 clients should be able to use the Kamailio server as a proxy and register to IPv4 sip servers 3.) IPv4 clients should be able to use the Kamailio server as a proxy and register to IPv4 sip servers 4.) IPv4 clients should be able to use the Kamailio server as a proxy and register to IPv6 sip servers 5.) TLS must

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    ...framework in cloud (with public IP) and I want to integrate it with digital PABX and providing SIP client under Yii2 framework. The PABX has set up an SIP server which only has local/private IP address and I want to add SIP client within my web apps. The SIP client should recognize incoming call numbers (store the incoming number, extension and timestamp

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    My client have a USA registered company and client who is willing to hire a online telemarketer from the USA or Canada to dial the number and give them the piece of advise and the quotation for the content removal from the infidelity content removal website. This project specifically for the USA guy or girl who wish to make some money while giving

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    Total VICIDIAL or Asterisk installation with all required features for Blended call handling and Call broadcasting etc. Access Bigblue button or Open Meetings from Wordpress elearning software. Show installation and configuration of the complete system:

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    Dear programers, We are SME CO. that is planing to use xCALLY contact center to perform our clients contacts. Initially, we want to connect the xC...rating. [login to view URL] information 3. Survey information. Currently, we have one database for all!. xCALLY is hosted in cenOS server. It is a GUI base software that operate Asterisk for all contacts types.

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    Trophy icon E-Commerce Product Photos Ended

    ...as "old photos" and the ones I need to be edited as "New Photos". 1.) My cover photo needs to be similar to the current one with the emphasis on the date dial mechanic (like Old Cover Photo). 2.) I'd like 4 individual bag photos to be combined into one where it shows all 4 bags (like Old Photo 3). 3.) Need a photo with the bag dimensions

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    ...installed. Information for invoicing is received from our Asterisk PBX. All Operating Systems are Ubuntu 18.04 and to work on our server You must have a late Teamviewer available. Requirements: Apache with Nginx rev. Proxy and Certbots-SSL, MySQL DB, SMTP Servers, KVM, LibreOffice, PHP, HTTPS, Asterisk PBX, Ubuntu OS, etc. -- This is not for amateurs !!

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    Total VICIDIAL or Asterisk installation with all required features for Blended call handling and Call broadcasting etc. Access Bigblue button from Wordpress elearning software. Show installation and configuration of the complete system:

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    ...limits can be adjusted. We also need to know the relationship between the voltage at the opamp side of these current limit resistors, and the current limit, so that when we dial up the current limits by adjusting the trimpots we can measure what the current limits are that we are setting. Attached is a schematic of this circuit. We feed in the voltage

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    ...Management Script 18. Online Learning Management Script 19. Tour Booking Script 20. Online Property Management Script 21. Classifieds Script 22. Yellow pages clone script 23. Just Dial Script 24. Airbnb Clone 25. Flight Booking Script I need all which was built in latest frameworks with responsive design in affordable price. I do not wanted to developed

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    Create several geographic asterisk servers with one central a2billing system

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    ...All, We are need a service to setup a monitoring system preferably Nagios to monitor 4 asterisk machine and provide event logs and alert on the following status of the machine, IP phones, line and bandwidth below. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Please submit your

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