Agi ivr asterisktrabajos

We want to use Asterisk PBX with Avaya AAEP. The idea is to transfer the call when it reaches the IVR to another PBX (Asterisk) installed locally. Requirements - * Integrate Asterisk PBX with AAEP for SIP transfer * Call transfer can be both bridge or blind. Both, from Avaya to Asterisk and back should be supported * During transfer metadata like caller_number, call_language, etc should be passed

its basically a voicemail ivr. plus a few features. will discus by phone. - audio playback - credit card payment

I need to build a mechanism inside of twilio that will enqueue a large amount of calls and transfer them to my PBX(3CX) once the perviously enqueued call is ended. Some on hold music and a looping voice recording would be great. There are 3-4 inbound numbers that will feed this, and 3-4 agents that will handle calls. So a user calls in to a number, hits the Twilio IVR, and sits on hold until an agent is no longer on a call. Then the call is transfered to 3cx.

Need to install Fusion PBX on a Debian or Cent os server that can handle 500 extensions and 1000 concurrent calls based on call center ring queue. Also we must be able to monitor the live calls going on.

I want to find a freelancer with these skills. C# .NET, ASP.NET Core Linux (setting firewalls for docker, deploy containers, install nuggets, add shh keys, certificates, investigate why something doesn't work, etc..) Docker - Containers APIRest SOAP Ansible - NiceToHave Good English Czech Language - NiceToHave Asterisk - NiceToHave Janus Know what means these words. Webhook, Oauth2, API, Websocket, FTP, SFTP, Minimum commercial developer experience of 4 years. I am looking for longer coworking. The project can be from 4MDs up to 60MDs. Mainly working on projects for our customers with various technologies. I prefer personal communication daily. Every project will ordered separately

I need to customize a Java open source sip auto dialer. The person needs to be an expe...customize a Java open source sip auto dialer. The person needs to be an expert with a clear knowledge about voip. the opensource with the source is at GitHub: Basically, I want the software to perform a call hold, dial on the same port another call, then upon the success of that call, bridge that call with the call in hold. The software will dial to an IVR and the purpose of this job is to be able to use the same port 6 of a maximum of 6 calls, that is allowed by the sim card into the gsm gateway. so, we need to add that feature to open source. If you feel you can do this job, please send me feedback to discuss price and timing, we need this asap, thanks!

we need asterisk developer to rebuild chan-dongle for specific usb modems our budget is 100 per modem

Develop a STT and TTS UniMRCP plugin for existing websockets based APIs. 1. Setup a testing environment using Asterisk PBX (UniMRCP Client) 2. Develop a STT plugin 3. Develop a TTS plugin

Hello, we have a production environment that has asterisk 16 on centos 7.x with FreePBX 15. we have our own developed queue call back script that returns calls if the queue is busy. we need to add some functionality and fix some bugs.

As an IVR Application Development Software Engineer, you will be developing next generation Interactive Voice Response application solutions using the latest frameworks and tools. As a member of the Engineering team, you will design, develop, and deploy application solutions for the company expanding list of clients using advanced call processing, speech and network technologies. On a daily basis, the most important responsibilities you will fulfill are to: Develop IVR applications using a variety of technologies, including Java, .NET, VXML, JavaScript, JRuby, and Groovy Experience with VXML, ASR, Call Center Telephony, and IVR development platforms, such as Cisco CVP or Genesys Deep understanding of the principles of IVR Voice User Interface design

Have a current project but want some extra eyes or even someone to take it over.

Hi Neosaransh Cloud Solutions, i m working in voip in morocco and i need a install and configuration for easy vpn issabel asterisk on my server local

Hi Satyajit K., i m working in voip in morocco and i need a install and configuration for easy vpn issabel asterisk on my server local

Hi Ricardo P., i m working in voip in morocco and i need a install and configuration for easy vpn issabel asterisk on my server local

Hi Debjeet G., i m working in voip in morocco and i need a install and configuration for easy vpn issabel asterisk on my server local

Hi Jeremie C., i m working in voip in morocco and i need a install and configuration for easy vpn issabel asterisk on my server local

Hi Prostitov A., i m working in voip in morocco and i need a install and configuration for easy vpn issabel asterisk on my server local

i need to install an easy vpn isabel asterisk

To set up installed Asterisk PBX stable 16 qnap x86 SSH small set up 4 sip 5 phones 5 softphones GUI - web admin simple voice missed number to email debatable option

Config : Route the Inbound call based on prefix to specific IVR in NAS/local Drive. Play random IVR on the spefific folder and can be many Service ( 1 folder is 1 service, can be grow with option IVR) Other things please msg me

We are looking for a custom-built SIP Softphone (PJSip Asterisk extensions) project using .Net 5 and C# to with the following features: 1) Source Code in Visual Studio project using .Net 5 for Windows Application 2) Downloadable internal directory/contacts 3) Simple UI 4) Minimizable to Tray 5) DTMF and Multi-line with hold, conference, transfer 6) Configurable ring notifications 7) Cannot use proprietary/paid license components 8) Completely configurable from web endpoint (can be XML/YAML/text for the extension, server, etc) 9) Code standardly documented in English

...(for e.g. 7/1 and 30/30) Rate and payment term update by uploading rate sheet Block country codes and individual CLIs from origination IP Block country codes and individual CLIs to termination IP Show active calls Download CDRs with additional filters like date range, client, supplier, number, CLIs, number, and number group. Show stats Error logs, log for the calls not connected Addition of IVR / audio file for origination IP A client portal where the client can see the number and numbers groups allocated to their IP, stats with the filters as stated in point 14, active calls, test account, notifications, and access list. Finance module/ billing page Upload and auto access list (this would show all connected calls with the group name, test number of the number group, CL...

...SUPPORT Server Support Server Monitoring Server Optimization Server Security Server Setup * TECHNOLOGIES CLOUD PLATFORMS Digital Ocean AWS Google Cloud Azure Godaday OVH DEVELOP TOOLS Docker Kubernetes Jenkins Ansible Git Puppet CHEF Nagios Vagrant New Relic ELK * VOIP SERVICES Voice Configuration Calling cards and billing services Asterisk and PBX services VoIP Multi Tenant services Voice Broadcasting and call center services 24/7 VoIP Monitorin * INFRA SERVICES Network Support Database Support Database Admin Support Application Support System Administration and Virtualization Amazon Cloud Migration Google APPS Migration VPN Dedicated Hosting * OPEN SOURCE APPS Vi...

Hello, Few call center employees need to work from home, so VPN need to be set up to connect them from home to head office network to make & receive phone calls. Input data: - We use PRI line with 100 numbers, out of each around 5 need to work remotely. - PBX is Asterisk-based; also Yealink, windows server 2012, Call center agents use X-lite old version as free SIP-phone tool. If needed, there is DyDNS address of the head office, as head office IP address is dynamic. Server & PABX map is attached. Time to finish project : 2 days + 1 day testing. There is no VPN existing in head office currently. Call center laptops at home for connection with VPN have Windows.

I need you to a part time developer on daily basis to support my friend working in USA. need experience in Spring Framework, Unit testing, Maven Better to have knowledge on IVR domain but not mandatory

I need to build a "telephone system center based on Asterisk". (Required in version 17)

We are a UK (our own consultants/developers are based in Indi a where I have a subsidiary company) in India based open source CRM/ERP company, one of the applications that we do a lot of work with is Vtiger CRM and we have a number of customers looking at telephony integrations with FreePBX/Asterisk specifically. We also have a couple of customers who already have telephony integrations in place but these need to be replaced with integrations that don't use encrypted code due to consultants/vendors vanishing after doing the initial set up and now further customisations are not possible (ACTUALLY NON OF OUR SOLUTIONS SHOULD USE ENCRYPTED CODE) These solutions range from fairly basic click to call for outbound through to more comprehensive inbound screen popping, custom actions...

Hello Freelancer, This project is for experienced female voice artist who has own personal studio. The artist should have very attractive voice. Native English speaker will have added advantage. Off course the IVR recording should have background music. You have to record the following IVR scripts. SCRIPT 1: We have provided dedicated phone line to our employees to contact with their friends or family. Friends and family will call at the number to contact with the desired employee. When they will call they will hear a IVR. The script is given below--- Thank you for calling ALSOFT. We are certain to provide better work culture to our employees. We are an emerging and fast growing organization. ALSOFT is committed to simplify communication between emplo...

need help with my asterisk setup which we use FreePBX, and some help on my virtualizor and WHMCS setup

Needs to hire Freelancer Seeking Asterisk/Java/JavaScript/HTML developer(s) to make repairs to a web-hosted VoIP API endpoint. Our Product Overview: We have a custom VoIP solution for an open world game. Players on our servers have an x, y, z coordinate and a room r. Players to also have the option to open a web browser, navigate to a URL, and join their voice room (r) with other players. Those close to each other using x, y, z coordinates will be able to hear each other, but the volume of each fades over distance. The system should be pretty abstract. At minimum, we should be able to make calls to a Java web API (maybe running Tomcat) and be able to update player's x, y, z, and r values. We need to make sure the current product is repaired to be able to be scalable and have ...

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Searching for the valuable software house who has exp on developing iOS and Android Mobile application on SIP protocols with the help of Admin panel and Must have experience on asterisk or elastix. Deep Project details will be provided over the PM.

write quality or research content for digital media company website - write about company - write about SEO, SMO, web development, App development, PPC, Digital promotion like (Bulk SMS, Voice calls, Email, IVR, Missed Calls) - QnA about these services - Blogs on trending topics

Configuration and support for PBX Siemens HIPATH 4000 - for migration and interconnection with PBX Asterisk

I have installed OpenSIPS 3.1 server with UAC, CP and rtpproxy on Debian 9 server with 2 network interface. (Asterisk server - OpenSIPS - Service Provider) Now I am able to establish calls and also registering to my service provider. I need someone's help to fix the rtp traffic and routing.

A list of apartments and their owners on a screen. By pressing a phone button, you will call that person. If the person wants to open the door, he presses *9 or something like that. All on a web page. There are JavaScript sip clients. The phone part is by asterisk installed on the same raspberry pie using google voice.

Hi, I need to build a sytem that can detect IVR when you call, record a part of the ivr and detect which messages is, with previus record files into the server. The target is to to avoid IVR balance, IVR of blocked sims and detect which record has been detected. We need the build in Python, asterisk.

...are in need of an e-flyer or e-brochure that is easy to email to potential customers. This marketing material should look modern and sleek, incorporate our logo, describe the program, and list pricing (note: the hourly rate for the aircraft should have an asterisk to indicate the rates are subject to

Onyx Flight is an aircraft charter operation that offers a program called the "Onyx Flight Membership" program. We are in need of an e-flyer or e-brochure that is easy to email to potential customers. This marketing material should look modern and sleek, incorporate our logo, describe the program, and list pricing (note: the hourly rate for the aircraft should have an asterisk to indicate the rates are subject to your membership agreement and adjusted yearly). Yearly Membership Fees are listed in the attached description document, but for the hourly usage rates we need to add light jet rates ($3700 an hour) and mid-size jet rates ($4800 an hour) in addition to the King Air 200 rate on the document. I've also attached a little pdf I made just to have some easy marketin...

I need a Goautodial V4 or V3 based dialer with an ivr. with hosting

I will provide Arabic voice over recording services for IVR systems, TV commercials, Radio Commercials, motion graphic videos, Tutorials, documentary, educational courses and mobile apps with high quality, fast delivery and competitive price.

We have a social media app using open source linphone SDK for android and ios SDK for voice. Need voip developer to enable video in existing system and autoscalable solution. Using asterisk on backend rabit MQ

1. Set up IVR Flow based on the requirement. 2. Integrate the IVR with two external systems for validation and retrieve data based on the inputs provided • Flow will need to recognize caller’s input using DTMF and LEX Voice bot • Basic Flow o Validate the caller using TAX ID based on information available SYSTEM 1 o Validate ID (SSN) using SYSTEM 2 o Validate DOB based on the information available in SYSTEM 1 o Once the information is verified, look at a specific table and read out the information using TTS o Must be able to send information through fax to caller when needed

Hello I have a centos8 running kamailio asterisk and a2b, the service has chat with file transfers and of course voice and video calls, I have built it with several developers and I have also android custom built clients and apis, Im looking for someone to start installing the required above explained prefferably on debian 10 but 9 would do its a cooperative work between you and me to copy my current service from centos to debian and make the clients work, we will deliver a fresh stock debian 9 or 10 server with its public ip and no firewall

Desenvolver uma extensão que ao ser acionada ligue para uma número e faça a conferência com o canal que acionou a extensão.

...trunks_rules.rule_name > cdr.prov_rule_name trunks_rules_rule_cost > cdr.prov_rule_cost > cdr.prov_rule_first > cdr.prov_interv > cdr_prov_name Relationships between tables attached in image Copies of db table in [2021-01-12 14:00:45] WARNING[6348][C-00000290] res_agi.c: DeadAGI has been deprecated, please use AGI in all cases! [2021-01-12 14:00:46] ERROR[7046] cdr_mysql.c: Failed to insert into database: (1442) Can't update table 'cdr' in stored function/trigger because it is already used by statement which invoked this stored function/trigger. [2021-01-12 14:00:46] ERROR[7046] cdr_mysql.c: Failed to insert into database: (1442) Can't update table 'cdr' in stored function/trigger because it is already used

i need the numbers in twilio to be sip accounts and to create an ivr for incoming calls.

Hello, I have installed an Openvox UC 501 Asterisk Based IPPBX at my home. I have a BSNL FTTH Connection with a Landline Number. The landline is currently working through a POTS port on my Syrotech ONT Modem. I want the Landline (VoIP) to work as an SIP trunk on my IPPBX. I live in Raipur, Chhattisgarh, India. Thanks.

Asterisk Version - 13.35.0 Vicidial Trunk Version - 2.14b0.5 DAHDI Tools Version - 2.1.0.2 We have set up VICIdial successfully and it's running as per requirement, We are facing a problem in the callback. A scenario which running: The agent gets a call and talks to the customer after finish the call if a callback required he will schedule a call, call triggered when times matched with the set time and allocated to the free user. that's fine If any Agent is available then VICIdial initiates a call, Call ringing, When a call picked by the customer then VICIdial assign a call to the agent meanwhile if the agent got another call(Inbound call) then call hangup, in such case we don't any information about callback instead of will check in database. Scenario Required: I...

I have a FreePBX instance and need assistance cleaning up the configuration, tightening security, and troubleshooting SIP connections. Please do not bid on this project unless you have in-depth FreePBX experience.