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
...design that is similar. All the pricing and details can be found on the website: [登录来查看链接] 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
Necesito configurar una centralita Asterisk para 5 puestos de trabajo y darle acceso al exterior a través de la red de fibra contratada, en Madrid. Busco un profesional en instalación de este tipo de centralitas en oficinas y que pueda realizar el trabajo en Madrid.
our company is looking for experienced cold callers who can generate leads. This job requires fluent English skill and at least two years similar experience. work time: 4-8 hours per day. between 8am and 1pm EST. please send your resume to be considered.
... 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
...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
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
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
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
Friendly Cold-Caller and virtual Assistant with North American Accent We are a young, dynamic and eco-friendly sports company with an exciting product. Your Task: -Call potential B2B clients and use your friendly charm to foster interest in our sports-product -If a prospect is interested, send a standard email and schedule a call or meeting with our
Friendly Cold-Caller and virtual Assistant with UK English Accent We are a young, dynamic and eco-friendly sports company with an exciting product. Your Task: -Call potential B2B clients and use your friendly charm to foster interest in our sports-product -If a prospect is interested, send a standard email and schedule a call or meeting with our executive
It is a simple work.I wants to hire someone could help me to call the people in USA. I have the people's name and company name, need you to look for the person's correct email and cell number from website or call his company to ask for contact information. If you have HR or Headhunter's experience will be preferable. If not,it is also ok.
...PERMISSIONS that would allow Caller to Grant a permission to a user Check if a user has a particular permission Caller is an external system that uses our module (particularly it calls our API). Permission is a user right to take an action or access a resource. Permissions for resource access can be READ or WRITE. For example, Caller can grant a permission
...(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 --------- ---------- -------------------------------------------
This is an ongoing project as I have thousands of contacts to reach out to. Will need a highly skilled cold caller with English native accent to call gatekeepers of companies and ask for the direct number of the decision maker. If possible, talk to decision maker and get permission to send email or text with sales materials. I will provide scripting
...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...
This is an ongoing project as I have tens of thousands of contacts to reach out to. Will need a highly skilled cold caller with English native accent to call gatekeepers of SMB companies and ask for the direct number of the decision maker. Please note that every non-relevant contact number provided (e.g. incorrect) should be replaced free of charge
...do are using USB caller devices to show caller ID on the screen, at the moment we are using This product: [登录来查看链接] Which has drivers to install virtual com port for this device. There is another USB device, which is USB HID, can you be able to create virtual port to transfer caller id info same as device
...clue what we are doing. Right now, we are able to call each other via sipjs / webrtc, but on answering a call it breaks, because the OK on the invite is never send to the caller. But apart from that, we need a lot more help to get everything setup and when everything works as expected, we would introduce the feature to our customers, so we probably
...'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 -
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
...do are using USB caller devices to show caller ID on the screen, at the moment we are using This product: [登录来查看链接] Which has drivers to install virtual com port for this device. There is another usb device, which is USB HID, can you be able to create virtual port to transfer caller id info same as device
- Task: Caller will be provided list of numbers and should make about 150 outbound calls. A script will be provided but you will have to update it with notes. - Qualifications: I need someone that speaks great English, has organization skills, experience with VOIP (SIP) or has a similar program already so that we can check. Duties: Negotiations
I am kiran patel working in a bank, i am interested in developing mobile apps which are of great value to society and easy to use. Below are the re...person must be converted to text on the go, so that it can be read by the user and he can respond accordingly by speaking. 3. This app must also records call with both the caller and receiver voice.
I am kiran patel working in a bank, i am interested in developing mobile apps which are of great value to society and easy to use. Below are the requi...person must be converted to text on the go, so that it can be read by the user and he can respond accordingly by speaking. 3. This app must also records call with both the caller and receiver voice.
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.
...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 (ID.nr. XX ) comes into the queue 500 08:13:05
...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
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,
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.
I am looking for provider who provide me sip gateway for Indian operator but with open caller id and API. My concern is to send broadcast pre recorded voice.