Asterisk softphone dialer jobber
I want an calling application people can call via internet to mobile(sim) with this method :- > METHOD + VOIP Server like Asterisk / FreePBX + Protocol like SIP/PJSIP/WebRTC + Library like liblinphone for Android or iOS. + Making outgoing call using this above library functions like , () + A little programming to change number every time. ONLY BID IF YOU KNOW THIS METHOD AND TYPE "READ" IN YOUR PROPOSAL IN TOP. OTHERWISE I WILL IGNORE YOU
I want an calling application people can call via internet to mobile(sim) with this method :- > METHOD + VOIP Server like Asterisk / FreePBX + Protocol like SIP/PJSIP/WebRTC + Library like liblinphone for Android or iOS. + Making outgoing call using this above library functions like , () + A little programming to change number every time. ONLY BID IF YOU KNOW THIS METHOD AND TYPE "READ" IN YOUR PROPOSAL IN TOP. OTHERWISE I WILL IGNORE YOU
Looking for someone with experience in AI Voice Bots and FreeSwitch or Asterisk. Must have experience with STT or Transcribe.
Job Description Should have worked on: 1. IP based Call Center. 2. Dial Plan in Asterisk. 3. Good understanding of VXML and IVR development. 4. Good understanding of VoIP, SIP, SS7. 5. Prior working on voice cards, media gateway, WebRTC etc. 6. Expert in Freepbx 7. Understanding of Telephony, CCXML,VXML, json - Scripting skills (Perl/Shell etc.) - Understanding of Linux/Unix commands/distributions (RHEL,CentOS,Fedora) 8. Expert in Cisco voip along will get priority
We are currently hiring Telecallers or a team who has past experience working in a call center. You will be login on our dialer to take calls and set up appointments. You would be operating during business hours in CST (Central Standard Time, US). Applicants who are fluent in English and have neutral accent will be considered. This is a full-time job whereas currently an agent is making around $1000 USD a week on this project. Only apply if you are able to devote 8 hours a day with us.
Hello, We have a server with 3 main parts: 1) PBX server (Asterisk + local Interface) with yealink voice gateway 2) CCTV recording for 50 cameras with Raid storage 3) MS Server 2012 with MS SQL Server with attendance system software, Recently we had attack by Ransomware so we had to restore MS Server 2012, and we were advised to install a cloud backup. Veeam seems us best option in terms of price-value. Project is to install & set up the back up of all data (except CCTV records) into Veeam to be able to restore server if any crashes or ransomware or other issues will happen. The Machine where Veam access will be installed: "IT laptop" with 2.0 Ghz, 12 GB RAM, HDD SSD type with 300 GB free space.
We have configured PJSIP to talk to Oprerator Airtel India. Registration is done. - But the second calls leg are getting disconnected after 30 seconds and but the first call leg continues without disconnecting. We want someone to check the configurations using anydesk and make this SIP incoming and outgoing working. There is no issue with operator as its working with chan sip.
Telemarketers needed to call business owners seeking financing. Businesses in USA. Calls must be made during USA daytime hours. Telemarketers worldwide. 15 second script. If they are interested input their name number and email into our site contact form. Deals close in 2-3 days. Telemarketers are paid $100 per clo...25% to 50% Script: " Hi this is (your name) calling from Discount Merchant Advance we have the lowest merchant advance rates guaranteed we guarantee you the lowest merxhant advace rates. How much can you use right now?" If they give a number input their info into the system and say " An account executive will follow up with you mmediately. Thank you goodbye. " That's it. Easy sale. Dialer: Dialpad 100 Leads provided. More lea...
We are currently hiring Telecallers or a team who has past experience working in a call center. You will be login on our dialer to take calls and set up appointments. You would be operating during business hours in CST (Central Standard Time, US). Applicants who are fluent in English and have neutral accent will be considered. This is a full-time job whereas currently an agent is making around $1000 USD a week on this project. Only apply if you are able to devote 8 hours a day with us.
into this image. here you can see a tab named "Aadhar" this has a help tip saying this tab will accept 1) either a 12 digit full numeric values or 2) a 12 digit full numeric values followed by an asterisk * . But, In this 2 set only one part works, i.e the 2nd part only works. you try to include the 1st part to the code of the master branch and let me know i will check the same in the UI and we can be sure on this.. Deadline is urgent, 2 days.
im looking for an asterisk developer to work full time must have extensive experience with asterisks system and voip protocol
Download free trial of this Modify this to work with FreeSwitch/FusionPBX and SuiteCRM. Will have a development server setup soon for you to use soon. Need it done in 4 weeks, place your bid and let me know how confident you can do this.
I am running a VoIP Business and I need someone who is familiar with asterisk systems. If you can work on vicidial and magnus billing and belong from India (West bengal)please contact me on (Removed by Freelancer.com admin)
We are looking to develop a Freeswitch dialer for Voice mail drops
Holai Ambiorix R., vi tu perfil y note que trabajas con Goautodial por lo que te solicito ayuda con una situación que tengo con mi instalacion. Instale Goautodial 4 en asterisk 13.17 en un servidor con IP publica, configure una troncal, las llamadas manuales salen pero no hay sonido en ningún sentido, tampoco suena el audio de bienvenida, "You are the only person in this conference". Agradecería mucho tu colaboración para resolver este tema. podemos discutir detalles por chat.
Hello, I need a documentation to build a secure and reliable cloud AstLinux PBX. I'm looking for an expert in Asterisk AstLinux. The instructions must be clear. We cannot use any web GUI such as FreePBX. We have to use the minimal AstLinux distribution from: I can get it up and running, upload the ISO image to a cloud VM, open all the TCP and UDP ports, and give you full access to the server. We like to use , but if you have a better suggestion we will consider it. ##################### ###### SECTION 1 ###### Usage example of IVR menu: # WORKING HOURS: # Monday to Friday 8AM to 5PM # Auto attendant: Press 0 for Reception, press 1 to speak with Bob, press 2 to speak with Alice, 3 to speak with Jon, and 4 to get our address. Extension: 0
I’m needing a asterisk developer to assist and help maintain the system someone with extensive experience is needed
I am looking for a solution for the inbound cluster, I will try to explain my scenario to my best, need experts to please help me out. I have a 5 server cluster on Vicibox v9, server 1 is DB, server 2 is WEB, and the rest of the 3 servers are telephony. I have approx 250 people to login into this cluster to receive calls only. Question No 1: Can this cluster setup be able to handle 250 users' web requests? Question No 2: When I create a phone ext for each user I have to define on which server this phone ext should register and get calls from, is there a way that I create all phone extensions on one server to get all users to register phones on this server but calls should be load balanced on other servers as round-robin.
Hi. I would like my vicidial auto dialer configured to have multiple press 1 campaigns running
For troubleshooting issues with random call drops on incoming calls, we are looking for an engineer with a good understanding of SIP protocol and experience with Asterisk system. To further describe our problem, external calls are landed in our telecom providers PBX and forwarded into our Asterisk (v15). Calls drop suddenly, and our telco provider has informed us BYE-requests are generated from our Asterisk. Call drops do not happen if we make calls to Asterisk from the local server. We need help to find the root cause of the issue and to correct it.
We had a developer create a CDR report for a customer on an Asterisk based platform. The report is generated daily on the PBX. The customer would like the report email to an email distribution list daily at a specific time. Looking for a script to be created that will grab the new file from the specified folder and email it to the customers email distro as an attachment daily at 9 AM EST.
Looking simple app in React native. Where i can receive video call from Asterisk sip and send back DTMF code. Only incoming video call required # DTMF code can be sent by app support android and ios.
Requirement : Asterisk - php I have an Asterisk box running Incredible PBX and phparia. There is a php web page written (See attached png) that has inputs boxes for 2 phone numbers - a connect button and a disconnect button. Project is to add php code to web page that will dial and connect to Phone 1 : on answer will dial and connect to Phone 2 : then bridge the call. A disconnect button will end the call. There are 2 sets of inputs so 2 different calls can be placed at the same time. Done One of Many Small Projects
Asterisk santrali c# dili kullanarak ve kaynak kodlarıyla beraber bir yönetim programı oluşturmak istiyoruz. Daha önce asterisk santrali üzerinde çalışmalar yapmış, konuya hakim kişiler ö santrali c# dili ile yapılacak görseller projemize eklenmiştir. Buradan inceleyebilirsiniz. Bu tarz bir yönetim paneli talep ediyoruz. Bu projede bizlere yardımcı olacak ve proje ile ilgilenen kişileri teklifleriyle birlikte bekliyoruz!
Looking for a freelancer to make a basic dialer mobile app that will update call logs to the server user wise. Please note the below points. 1. The user will dial manually. 2. Auto dialling with excel data upload is not needed. 3. Updating of remarks in dialer also is not needed. 4. The quote should have two options - with and without call recording uploading. 5. For incoming calls, the user can select whether it is a personal call or an official call. 6. The app is required for Android only. 7. Need to know if freelancer can integrate the app with Suitecrm
...for various clients each month. Agent requirements: 1) Strong command of the English language, without an accent. 2) Ability to learn and adapt to new ways of working. 3) Laptop/PC with headphones, microphone and a good internet connection (Not Wifi). 4) Working from Monday to Friday, European office hours between 10:00 - 17:00 EEST. 5) Being able to do cold calling. 6) Responsible for his/her dialer (Example Skype) to make calls around Europe (We pay for the subscriptions monthly). 7) Ability to make at least 12-15 calls/Hour. We offer: 1) Fixed income for each appointment set, with biweekly or monthly payments. We pay 20 Euros for every appointment set. 2) Possibility to take on new roles in the company later on Team Leader and Project Manager. 3) Training and support prov...
We are currently hiring Telecallers or a team who has past experience working in a call center. You will be login on our dialer to take calls and set up appointments. You would be operating during business hours in CST (Central Standard Time, US). Applicants who are fluent in English and have neutral accent will be considered. This is a full-time job whereas currently an agent is making around $1000 USD a week on this project. Only apply if you are able to devote 8 hours a day with us.
...(male/female/stores/admin) into dashboard while creating the create new user form and to show on New Dashboard. **Configure to send reports by Email to supervisor,agent email ids, every day and email notifications to agent on status changes. **Require to add Reason for Break to be available on status of agent and in dashboard. ** Require to auto add users into queue as soon as they login. - REQUIRE ASTERISK IP PBX SKILLS ** Require to ensure Agent Paused Counter Threshold limit means- Shows the number of agents paused/ready available on visible queues module, with the option to prevent further pause actions when a certain threshold is reached. **Rest configuration on styling and fields additional to show data during development if customer needed, only minor not major. ...
This is our regular requirement
...(male/female/stores/admin) into dashboard while creating the create new user form and to show on New Dashboard. **Configure to send reports by Email to supervisor,agent email ids, every day and email notifications to agent on status changes. **Require to add Reason for Break to be available on status of agent and in dashboard. ** Require to auto add users into queue as soon as they login. - REQUIRE ASTERISK IP PBX SKILLS ** Require to ensure Agent Paused Counter Threshold limit means- Shows the number of agents paused/ready available on visible queues module, with the option to prevent further pause actions when a certain threshold is reached. **Rest configuration on styling and fields additional to show data during development if customer needed, only minor not major. ...
It's a US-based campaign. we need reps in any area that speaks English well. with a 5$ to 7$ price range hourly rate system/dialer and leads is ours.
Easy task for an Excel Pro. The sheet (attached) needs to be populated with SEQUENTIAL dates and times according to the pattern shown, from July 1, 2021 to July 1, 2022. To prove you are paying attention, please include an asterisk at the end of your proposal (as I have done ). *
I need someone with experience in Asterisk, Zabbix and Grafane to generate a batch file to capture how many SIP channels are in use on an Asterisk. This information goes to a Zabbix server and will be presented on a Grafana server. All servers are up and running. I need batch and the professional to install and configure the servers to present the information captured by the batch.
I would Like To make a VOIP Billing For VOIP Calling
we are hiring callcenter & freelancer if you are callcenter owner or freelancer looking for genuine campaign to start with experienced without any upfront and with Data Dialer and VOIP share your company profile or resume us process (night shift) per sale 40$ no fix Salary do tele sales and earn as much as you can
The app substitutes the first 6 digits of a number and converts the 6 digits prefix into "+ country code from data base + prefix from data base + last four digits of original number" The app gives you the option to save the converted number as a contact or dial it using default dialing app. Users also have the option to update the database.
we are hiring callcenter not individual if you are callcenter owner looking for genuine campaign to start with fresh or experience without any upfront and with Data Dialer and VOIP share your company profile us process (night shift) per sale 40$ and more
PROJETO LIBERAÇÃO DE EXAMES. Desenvolver um programa que seja capaz de entrar no sistema web, abrir um cadastro de um cliente e verificar o status do exame do cliente. O programa deve ter uma interface web simples com login de acesso, com as opções de cadastro de código de clientes e cadastro de código de exames e opção de agendamento da liberação com os campos dia data e hora...vai ter uma opção para o responsável pela liberação fazer esse procedimento diretamente pelo celular teclando o numero 1. Esse dado vai ser armazenado em um banco de dado, que será consultado pelo programa. Se durante a consulta tiver 1 o programa entra no sistema e libera o exame. OBS.: Para desenvolve...
We are a PBX telephone company looking to grow our client base. We believe in hiring a Telemarketing who can communicate with our leads and share valuable information about our products and services. Once leads are interested, you are required to set up an appointment to meet with our meeting. We paid hourly, and for every le...client base. We believe in hiring a Telemarketing who can communicate with our leads and share valuable information about our products and services. Once leads are interested, you are required to set up an appointment to meet with our meeting. We paid hourly, and for every lead that becomes our client, you get a commission from us. We need someone that speaks good English. We will provide you the dialer.
Hello! I have installed a Cloud FreePBX server [Fully configured] SIP with TLS + SRTP But the user's location is in a country that has SIP Traffic restrictions (Blockage) Everything is working and the calls go through 2 SIP Extensions but there is something wrong with the voice! The voice is like shaking! Note 1: If I use a VPN to avoid the SIP traffic restrictions, the voice is so clear and there is no such problem in the call Note 2: The funny is once I try to dial the other extension of *43 for the Echo test and our voice is still shaking If I pressed any num key twice or three times, our voice becomes so clear! I think may one of the following ideas are resolving the problem: 1- Install & Configure a Cloud SIP Proxy Server like Kamailio, OpenSIPS or any other SIP Proxy Ser...
Hi guys, we run fusionPBX on AWS with an end user via grandstream ht801 ATA. She is US based and complaining about excessive Post Dial Delay (PDD) - meaning she dials the number she wants to connect to...then hears silence for 4, 5, 7, sometimes 10 seconds, then hears the ringback tone begin. We use BulkVS as our main provider for outgoing, however also use Alcazar and can switch to others. I have a VoIP phone on my desk connected to the same and am experiencing the same issue. we need to: 1. be able to measure the PDD 2. understand what is causing it to be so excessive 3. correct it - forcing a ringback to 'cover' the gap is acceptable if not other solution You can work with our SysOp, Harold, and he'll provide access and whichever information you need. thx Len
I need to build a proof of concept python multithreaded application that will connect to one or multiple Asterisk servers using AMI protocol (preferably using a library) and trigger a callback function when receiving events. At the same time, one or multiple clients can connect to this python application using a predefined port (like 5039) and using the AMI protocol receive the events from the asterisk servers and send commands to one or multiple clients. Each connection to the Asterisk servers needs to be a thread and each connection from the clients needs to be a thread. Data must be passed between the threads. When the connection to the asterisk manager dies or is interrupted, it needs to be reconnected. When the connection from the client dies, the thread ca...
We are currently hiring Telecallers or a team who has past experience working in a call center. You will be login on our dialer to take calls and set up appointments. You would be operating during business hours in CST (Central Standard Time, US). Applicants who are fluent in English and have neutral accent will be considered. This is a full-time job whereas currently an agent is making around $1000 USD a week on this project. Only apply if you are able to devote 8 hours a day with us.
...send me any private messages. Please send your bid titled 'I Can Create An MQTT Broker Integrated With Postgre sql' We have a project that is an Emergency help call system for the elderly. It combines unique hardware developed by us that is both an IP Phone and an MQTT Client. Our central database on which the Asterisk PBX operates is in Postgre sql. We want to create an MQTT Broker on our server setup that automatically creates Clients based on the same login credentials as the Asterisk PBX, i.e the MQTT Broker will read the Postgre sql databse and automatically add clients and connections whenever a new Client/customer is added to the IP PBX. You will need specific experience in MQTT Brokers and database integration. Note - our prefference is to host the MQ...
... We are looking for an agile developer with skills in multiple technologies including Node.js and angular, ideally some php as well with the ability to adapt to different technologies by self learning. We are looking for attention to detail, communication and reliability. Mandatory expertise Nodejs Angular React NestJs Websocket Mongo OPTIONAL Sails Ionic Openvidu/mediasoup Asterisk Chatbot php symfony !NOT NEGOCIABLE! You are expected to work in GMT+2 timezone. We pay 2'500CHF per month (can be reviewed depending on seniority) and offer 20 days of paid holidays per year. TO APPLY 1. Let us know your experience in years and short description of your work performed on all mandatory elements and ideally on optional elements. 2. Send your CV 3...
I need a PHP script that will send by email a text file under the control of cron The file to send is located in the folder /var/lib/asterisk/logs_dts/MMMYYYY/ WHERE: MMM = month name upper case letters such as JUN. YYYY = 4 digit year MM = month number such as 06, 01 DD = day two digits such as 22, 01 YY = last two digits of current year Example: For today Jun 22 2021 it would appear as logs_dts/JUN2021/ The text file is not very large. When the script runs, it will locate the correct file and email it to my customer and send a copy to my email address. I think I covered everything but if you have questions please let me know. Thanks Bart
Develop an IP PBX (Telephone Server) with Asterisk based on Yocto Linux that will run in a virtualized environment on a rugged box pc, e.g. duagon BC51M. The telephone server processes function calls based on rules that are defined within the scope of the numbering plan/concept. The system is expandable and compatible at the vehicle coupling with vehicles that support the corresponding numbering plan/concept. For the signaling between SIP clients (telephones) and the SIP server (telephone server) as well as between the SIP servers, the function number plan is taken into account. The function numbering plan is managed as part of the phone server configuration. The telephone server provides an Administrative Webportal that can be used for SW/Config Provisioning, configure/manage dialp...
Web soft phone code needed. Advise (a recommended opensource or free) or develop and provide code for web softphone the provided code will be added to a Laravel app We have a FreeSWITCH pbx that will handle the SIP/calls The soft phone will need: dial pad, auto pickup call feature type a number to call optional additions: answer/decline call button transfer call feature
