Asterisk asterisk jobs

Hello, I am looking for a data scraper who can get me leads from America from Instagram for the personal trainer niche. I need 1,000 leads for now. If you do a good job, I will hire you for a lot more business. I need the following from each personal trainer / health coach’s Instagram: First name*, last name, business name, phone number* email address*, City*, State*. Anything in the asterisk is mandatory for you to fill out. If you cannot fill out, don’t add that lead, go to the next. Please let me know what your price is. Thanks,

Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and most of the software is setup. There are a few fixes to make, like Assisted Transfers to agents and PSTN lines, calls can be taken in random order, Caller must not hear himself (echo), PBX calls agent who didn't register, No-store cache control for Apache, etc. Teamviewer is required. --- Project will be divided into milestones to be discussed.

I want simple dialpad with all dial functionality in web application (html as ui) and call flow should be initiated from asterisk / pbx

Fixing and writing new asterisk codes for our servers.

Our company is in the process of moving many of our services. We would like to have a Asterisk VOIP Server installed and configured on a new AWS Instance under our AWS account. Candidate must have experience with Asterisk.

Looking for a developer with experience working with Asterisk and FreePBX. Looking for a billing solution.

New enhancements on asterisk dialplan I have 2 termination partner. Company A and B. They have their own DIDs to mask. They will provide their DIDs and we will store them in dids table to mask original calling ID. Call will come asterisk will choose one DID from did table and send the call to termination partner. it can be "company a" or "company b" but if it chooses company a DID it should send it to company a trunk. If a previous call is forward to certain DID then repeated calls will use DIDs from same company (instead of remaining limited to only that DID)

We will provide you with a clean install of Alpine Linux running on a raspberry pi. You install and configure the following software. hylafax+, iaxmodem, asterisk, freepbx We will provide you with port 22 ssh access to this device, we can also provide additional access if you require it.

JobDiva VoIP Integration Project Details, 1- Click to Dial: JobDiva supports Click to Dial using HTTP call or RESTFul API. Please check with the VOIP provider for the method that is supported and let us know the details. 2- Incoming Call Screen Pop: two methods are supported. a. Use the JobDiva API web services’ searchCandidateProfile and/or searchContact method to locate a matching candidate or contact. Pass digits only (without leading 1) as phone number parameter. b. Call a JobDiva page to get a display the search results Pass the phone digits only (without leading 1) as the ANI parameter. Check with Jobdiva to use either or www2.jobdiva.com. 3- Note for Incoming or Outgoing calls: Use the JobDiva API web services’s createCandidateNote and/or createContactNote met...

PLEASE READ THIS FIRST ! -- Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few fixes to make, like Assisted Transfers, calls can be taken in random order, Caller must not hear himself, PBX calls agent who didn't register, No-store cache control for Apache, etc. Teamviewer is required. --- Project will be divided into milestones.

Hi Asterisk O., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

i want make rules in my server 1- i want port number 1 call number 2 to i mint 2 i want when i send call for port number one when it finsh it make stop this port 2 or 3 min exe exten => _[*#0-9]!,n,Dial(Mobile/A0/${EXTEN},60,tTm,K) exten => _[*#0-9]!,n,Dial(Mobile/A1/${EXTEN},60,tTm,K) --------------- A0 port number 1 A1 port number 2 If a user make a call from a sip account, asterisk will send that call out using any one of the port from A1 to A16. If A15 was selected for that call, then after that call ends, A15 will not be used for 2 minutes for sending any other calls User Avatar and also new call from a15 should be dialed to any other 3 ports one by one for 30 seconds each so if we wait for 2 minutes and then call other port for 30 seconds, then other for 3...

VICI / PHP fullstack developer with some experience in designing and integrating front-ends. We are looking for a developer+designer with some experience with VICI (doesn’t need to know much about asterisk but some knowledge of it would be good). This position will contribute to a creating, and maintaining a custom designed interface for our VICI setup (primarily the agent screen design to start with, following up with the admin theme design and implementation), which would be used by our agents. Apart from working on VICI, the developer will also be working on some other in-house projects. Responsibilities: • Writing clean, secure and quality code that scales with concurrency. • Hands-on development, debugging and on-going maintenance of new and existing code (PHP, ...

Hi Asterisk O., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

Please call me for asterisk or vicidial related projects. Also if you need leads for cold calling, I can provide you numbers based on regions and operators. All genuine numbers.(No customer details)

A module where the caller has an option to exit the queue without losing their position and a callback occur when there position is up next on the queue.

i need a guide about how to implement it: also how to send a php post to originate it in order to use in some projects

lease if you haven't skill with Asterisk and linux don't offer me i install freepbx on raspberry when i make call its ring in end point but no odio in both side see log -- Added contact 'sip:200@;rinstance=5D817960' to AOR '200' with expiration of 600 seconds == Contact 200/sip:200@;rinstance=5D817960 has been created -- Removed contact 'sip:200@;ob' from AOR '200' due to remove_existing == Contact 200/sip:200@;ob has been deleted -- Contact 200/sip:200@;rinstance=5D817960 is now Reachable. RTT: 562.179 msec == Setting global variable 'SIPDOMAIN' to '' == Using SIP RTP Audio TOS bits 184 == Using SIP RTP Audio TOS bits 184 in TCLASS field. == Using SIP RTP Audio

PLEASE READ THIS FIRST ! You must use a PC with a late Ubuntu OS. -- Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, FreePBX/Asterisk, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few upgrades and fixes to make. Teamviewer is needed. --- Project will be divided into milestones.

Hi meral, I noticed your profile and would like to offer you some work. I want to use you as a consultant as i build my first Asterisk/FreePBX project. I would like to be able to send you a question and you provide an answer. let me know if this works for you

PLEASE READ THIS FIRST ! -- Inbound call center with Asterisk and WebRTC for agent access. This is only for professionals. Some requirements: Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, FreePBX/Asterisk, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few fixes to make. --- will be divided into milestones if needed.

We have 2 sites fileservers (1 site) archive servers (1 site), email servers - exchange 2019 (2 sites) and domain servers (2 sites), Asterisk (PABX (2 sites). Domain Servers need permissions set to allow 8 staff to access securely above securely and to the extent of their permissions. We also need - - emails to be able to save to the fileserver using messagesave. - printers to email / save to fileserver. Fileserver & Email server needs to eb backedup; replicated. Domain Servers needs to be backedup in case of need to restore. ASTERIX needs to be backedup Email servers need to be replicating between sites, backed up and have antivirus loaded (we use kaseprsky). We use ACRONIS for backups. All active servers run on windos server 2019. Our Archive server is on a vmware ...

Hello There, I hope you are doing well! We are currently looking for an Asterisk Developer for Voice Operations on a project. For the project, we need multiple Experts so we are offering you an opportunity to join our project. Share, explore and learn in the project. Before starting a project we need your resume with your mentioned experience for a client and your work portfolio so we can understand your knowledge better. Kindly look forward to this we are glad to work with you. Thanks, Vaibhav Joshi

i have a sip server (asterisk) i want to be able to add user name and password so outside networks can connect and make calls

Hey there, we are looking for a new designer who can create us an US DOLLAR BILL with a Face on the bill. The details we will tell you on chat. Please start your offer with an asterisk (*) to show that you have read the ad.

I need a routing configuration for the following scenario, solution could be either in asterisk or feeswitch. 1- receive sip call from calling party 2- have two sip outbound trunks, attempt first trunk, on 183 ringing (before 2nd ring) hangup called party channel (1st outbound trunk) but keep on ringing (park) calling party channel and dial second outbound trunk. 3- bridge calling party channel to 2nd outbound trunk called party channel. Thanks, Umar

- Please update IP tables in Firwall for Asterisk PBX as recommended - start Firewall again - with existing Firewall IP table setup I have Audio issue. To be solved

We are looking for an experienced and expert Asterisk admin to configure, administer and secure a new XiVO (Asterisk) Virtual PBX. The server has been setup and and the XiVO software has been installed but we need to setup SIP Trunking (Twilio and FlowRoute), DID Routing, IVR, etc.

3 server System, for an inbound call center, with Apacbe, Asterisk, WebRTC, MySQL, on Ubuntu OS. First thing to be fixed, is for agents to take calls that are in a queue, on our Operating Panel. This is for FreePBX / Asterisk specialists ! Some requirements: Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, FreePBX/Asterisk, SSL, Ubuntu, etc. -- Project has a few other things to fix and will be divided into milestones.

Hello, About 3 years ago you made an Asterisk/ Freepbx application for me. Now I want to make some corrections and extensions. Are you free and interested in the order? Best regards Florian Vollmer

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Need help in asterisk pjsip , Experience needed 1. Asterisk PJSIP Registration 2. kamailio 3. Rtpengine 4. NAT

i have created the extensions 10, 11 and 12, i can call between this phones using softphone but now i want to create a php file wich can be executed in order to call to the number "12" (for example). this is a part of circuits projects

PFSense Firewall Expert to configure and manage multiple locations via VPN Network & Server Admin for Linux and Windows Servers VMware ESXi VCenter Virtual Server Installs and Management Zimbra Email Server System and Security Management FreePBX Asterisk VOIP System Management Website Server setup and management, SSL, Speed Optimization Automated Monitoring for Servers Systems and Websites via PRTG, Zabbix, Observium etc This is an Ongoing Opportunity for Work Billed Hourly or per Task

...of Vicidial and it should work with Asterisk 1.6 or 1.8 or both. (or higher) All Vicidial UI and branding has to be replaced with our custom brand name, and interface should be changes sufficiently to look different. Logo branding removed Marketing automation Lead analytics dashboard Analytics dashboards SMS +++ To Email and Phone WhatsApp through the system Inbound Call Tracking Outbound Call Tracking Call Analytics Live chat Further the software needs to have some 30 additional reports/metrics displayed in addition to the standard reports available on vicidial. The code needs to be encrypted and should be packaged as an ISO.******* and also wget command format.********* The final mod vicidial should be able to provide per users/agent licensing Asterisk and Vicidial ex...

VOIP Reseller Setup: Able to do installation and configuration Asterisk, A2billing, and openSIPs on my server.

I give here an order for the export of the entire addresses of Twixtel 60 May 2019. The addresses are all visible, however, only the addresses can be exported that do not contain an asterisk *. The star stands for Robinson list. There can be a maximum of 9,999 addresses per export in csv. To be exported. I want all addresses (4,400,000) in csv. Have files. An easy game for every programmer.

This is a system for an inbound call center, with Apacbe, Asterisk, MySQL, WebRTC and Ubuntu OS. A few important things remain to be fixed. This is for specialists ! Some requirements: Java Script, , Bootstrap, CSS, HTML, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, FreePBX/Asterisk, SSL, Ubuntu, etc. -- Project will be divided into milestones.

I need 5 java and Asterisk Developer from US

I need a developer that can build a plugin to make any SIP sdk useable in Cordova 5.2 (or greater than) on Android and iOS. Javascript interface any SIP sdk in Cordova to realize the following app: - register with existing sip server Asterisk - make and receive calls thank you in advance for your response, Elena

Project Scooping My initial Scooping. We are a community based ISP, in order to broaden and add value to our user experience, we are looking to develop VOIP client app in both iOS and Android platform, that our subscribers can use to use to make Voip Calls. The mobile apps to be developed over Asterisk open source Private Branch Exchange platform. We have already developed the server side technology with the Asterisk platform, and will need someone to help would develop the necessary APIs to be required for the mobile apps. The mobile apps need to communicate with the server using the SIP and IAX protocols. Developer will be responsible and would be to develop fully functional native mobile apps on both iOS and Android platforms. Developer would develop the UI/UX, pages and...

I am looking to 1. Get asteriskpbx and goautodial on a server and connect it to a 16 port GSM gateway. 2. Setup a voice blaster and configure settings for sms /email on call dispositions. (AWS SES and SMS Gateways will be provided) 3. Setup appointment/visit scheduler for reminder calls/sms to be sent before the appointment date & time 4. Livechat integration with website

We are looking for a Call Filter solution for Asterisk .Call Filter will Whitelist and Blacklist numbers using rules.

Hello - Ahmed referred me to you. I have an issue with WebRTC working with a FreePBX/Asterisk PBX server. I use sipML5 and everything connects fine. The problem is there is no audio when OUTSIDE the network. If I am connected to the VPN, everything works how it's supposed to. As soon as the VPN gets disconnected, the audio goes away on both ends. If I connect my PC without a router and give it a static public ip with firewall turned off, I am also able to get the audio to work even without VPN. As soon as I put my PC back behind router with a private IP it then stops working again. RTP trace shows packets getting routed the same as when I use a PC softphone. The softphone works just fine with audio whereas the webrtc client has no audio besides the two cases I described above. ...

Necessito de um orçamento para software para Chat de Voz em ASP.NET C# com as seguintes funcionalidades: - Mensagem de Bem-Vindo - Mensagem da Administração - Salas Gerais - Salas Privadas - Salas Numeradas - Sala da Monitoria - Pin - Torpedo - Lista de Participantes On-Line - Murais de Mensagens - Caixa Postal - Menu de Opções Básicas - Menu de Torpedos - Menu de Opções Avançadas - Menu Opções de Pin e Caixa Postal - Menu Trocar Senhas ou Gravação do Nickname do Pin - Menu de Caixa Postal - Menu Lista de Participantes On-Line - Recurso Salas Numeradas - Recurso Sala da Monitoria - Menu de Murais - Menu de Opções do Mural - Recurso Regras do Serviço - Env...

This is a system for an inbound call center, with Apacbe, Asterisk, MySQL, WebRTC and Ubuntu OS. Most is functioning but a few important things remain to be fixed. This is nothing for freelancers who must google to find answers ! This is for advanced specialists ! Some requirements: Java Script, , Bootstrap, CSS, HTML, PHP, Web Dev., FreePBX/Asterisk, SSL, Ubuntu, etc. -- Project will be divided into 3 milestones.

We are looking for someone that can create a web app that can connect our Asterisk system with our web app through Asterisk AMI. We need this app to be able to: * Our calls typically enter a queue before being transferred to an available agent * Provide our web app with the caller id number, caller id name, start time, end time, time in queue, call time(length), extension (connected or attempting to connect), DID * Agent status - on call or not, waiting for a call, online, offline, ringing, call duration, total calls * When the agent answers the phone - provide the DID to the agent's web app through so that it will populate the account information on their screen * Verify whether the agent is logged in, logged out, working, or on a break to determine to send them the c...

hello greetings I have an Elastix 2.5 server and I want to integrate the Twilio API call () to send SMS and Whatsapp to my on-call staff when any calls come in for that team. The REST API call. I have knowledge of ASTERISK and PROGRAMMING, and will do the implementation on the server, then I need the scripts ready and a configuration and deployment manual.

Need technical expertise to perform the following projects 1. Install vtiger CRM on a cloud server 2. Integrate vtiger CRM with Asterisk PBX using two cloud servers the functionality for records and all fields in Asterisk PBX to be exported into vtiger CRM 4. Create the functionality for "Click to Call" while in the vtiger CRM

We would like to have a industry standard VoIP website for our business. Please develop a website with the best Graphics, Design and Mobile compatibility. Company Name: ALLIANCE GLOBAL C...with the best Graphics, Design and Mobile compatibility. Company Name: ALLIANCE GLOBAL COMMUNICATIONS LLC HD logo ( Multiple options to select to Finalise the logo) Pages are Home Products Support Service Plans Contact Us About us Request Quote Chat Copy rights- You have to write best content ever. Brochure design Flyer FAQ You can find examples from on-line also go through the general asterisk PBX features and how it works. Make sure Google analytic and Generic Search Engine Optimization. Make sure the best search keywords to be available in the Google. DO NOT COPY CONTENT. DO NOT CO...