need to set up a softphone that can be on a continuous loop to dial out on incoming data(telephone #s). I have incoming data that includes a phone # within the data. I have it set with my twilo account to call out on the phone # instant but i need a softphone installed so it can call out on the phone # automatically.
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
...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 in and will
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
...as a "Design Tracker" similar to an order tracker for pizza or sandwiches. I want the graphic to be a half circle dial like a speedometer. There would be 3 stages, therefore 3 graphics. Stage 1: Order received Stage 2: Custom Floor Plan Finished Stage 3: Design & Budget Ready for Review Heres our website address for an idea of our style and colors:
Language Solutions Experts ,we are mobile solution/service providers based out of India working with almost all the big mobile operators...into telecom VAS domain. We do business with almost all the major telecom operators in African continent, Middle east and South-East Asia. We would like to engage with you for IVR/Phone Messages prompt recording .
...connection of your internet • Ensure that you have entered the correct email address If after executing these steps your problem is not resolved then get hold of your smartphone and dial the ATT Email Customer Support Number to avail help from professionals. Read More: [login to view URL]
...(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 --------- ---------- -------------------------------------------
... when paired. The alarm and vibration should be programmed to cease immediately once the predetermined movement ceases, or if manually terminated by the user with the watch dial/phone. Additionally, the application will record a log of the start and end date/times of these movement episodes, the heart rate during the movement*, the GPS location* ,
I am a abroad study consultant and looking for male/female candidates who can work either from my office or their homes. They will have to dial the calls on the data which I will provide and will have to provide me the confirmed leads. The area of calling would be specific as I only deal in MBBS courses in Russia, Ukraine and Kyrgyzstan. Also, there
Hello there, we are going to create a new brand for our company and we need to update records for our IVR system. You can refresh the script and correct it if you feel that this would sound better. Down below you can find lines that have to be recorded: -Thanks for contacting ProDesk company! -To purchase our Quality Control Solutions, select 2 -Thanks
...'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 -
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
play m3u8 stream inside stream display a button call ( png ) when "click" button call dial 0123456789 without stop playing stream please don't waste my time. if "Bid ok then no ask for more money after" or "want deliver money before start to work" or "be sure you re able to do the apk function call speaker on" or "need 45 days" etc ....
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.
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...
...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 info (dial plan) to asterisk server (I have an
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
...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
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.
...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.
...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,
...Percent of wins based on whether user's reaction time was better or worse (if opponent is supplied); e) Average “1st” and “MOV”; f) Average/bell curve of E.T. +/- compared to DIAL; g) Average/bell curve of individual timers, like 60, 33, ⅛, 1000, MPH, etc; h) Average times (reaction and E.T. mostly) as it relates to hours of the day; i) Based on left
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.
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 .
I would like that when a customers calls the customer service number ( 0203/0106 ), They are greeted firstly and data in collected via ke...keypad/voice recognition would be used to determine who customers are before being transferred to an advisor. So I would need data collected before he call is transferred. Need IVR to be implemented And dtmf
...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
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
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.
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
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.
...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
Asterisk devlopment for voice broadcasting and vicidial