Asterisk predictive sip punët

require sip trunk call to china with randomly display CLI

I need someone to recommend, install and configure OpenSource asterisk based billing system.

Create a small demo project in Delphi 10.2 Tokyo or higher that uses the BareSIP libraries () to create a basic SIP client that registers a SIP extension to a SIP server (FreePBX or 3CX for demo purposes), makes a call to an extension, sends a WAV file as G.711u audio over the call, and hangs up the call.

MIKOPBX Call routing Training We are seeking a Russian or Ukrainian freelancer with specific experience in configuring and working with MikoPBX. The freelancer will provide training. Should have previous experience in Asterisk Dialplan and configure gsm gateway for incoming and outgoing calls to setup for call center this is open source PBX need to work on it

hi we need call routing training and one login page for extentions to see thier CDR and call recordings that already available in main Admin panel we just need to show them under each user its MIKOPBX opensource based on asterisk

Hi I'm not a Cloudflare expert, I have an account with spectrum enterprise enabled (trial) and an Asterisk pbx server with a websocket webphone and stun. I need help to configure Cloudflare in order to proxy the voip calls. Max 100 euros. Please bid only if you have experience with Cloudflare and custom applications (spectrum).

...techniques. Your ultimate goal is to develop efficient self learning NLP applications. Use statistics, NLP and machine learning techniques to create scalable solutions for business problems Analyze and extract relevant information from large amounts of both structured and unstructured data to help automate and optimize key processes Design, experiment and evaluate highly innovative models for predictive learning. Candidates who complete the demo task will be chosen for continues hourly based project work. Task derails: 1) Content : ======================== We processed a claim for services rendered on December 6th, 2021 to December six, 2021. It was received on December 22nd, 2021. The amount submitted was $300.00. This claim was paid on January 5th, 2022. The amount paid ...

Requirement: to be a native speaker of the Turkmenlanguage. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Turkmen, you must also indicat...the following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Turkmen, then it is necessary to indicate “Turkmen(dialect name)”; c) if in addition to Turkmenthere is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Turkmen(dialect), spanish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

Sip panel engineer, drafting. Need stamped drawings. So the plans can be submited to manufacture and city zoning committee Structurally insulated panel design Structal insulated panel engineer or architect. Familiar with SIP panels.

Deploy a cloud voip solution in aws configure application for multi Tennant pbx deploy sbc and configure and test

The goal of this project is to create a step-by-step guide for configuring Kamailio (a free and open-source SIP server) to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown as it will be used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using Kamailio or Martini Security's offerings. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using Kamailio with STIR/SHAKEN. Martini Security offers a certificate enrollment

...using FusionPBX/FreeSwitch or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using FreeSwitch with STIR/SHAKEN (). Martini Security offers a certificate enrollment client called Olive () which can be used if FusionPBX/FreeSwitch has working STIR/SHAKEN support built-in. If it does not have native STIR/SHAKEN support, the Martini Security STIR/SHAKEN signing and verification server called Vermouth () can

The goal of this project is to create a step-by-step guide for configuring Asterisk/FreePBX to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown and used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using Asterisk/FreePBX or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using Asterisk/FreePBX with STIR/SHAKEN (https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN). Martini Security

Configurate carrier, manual and predictive dialer for France

...will provide a .pcap file captured with tcpdump. The file captures VOIP calls made on our network. You will write a command-line utility program that reads the pcap file and outputs the RDP audio streams into .au files. The audio is in 8000hz PCU format generated by SIP telephone calls. For each SIP telephone call you will provide THREE audio files. One file will contain the RDP audio for each separate side of the conversation. The third audio file will contain BOTH sides of the conversation. In addition, for each SIP conversation, you will provide a data file containing JSON text, showing the TIME and DATE of the call for the PACIFIC timezone; also TO and FROM fields showing the names of the caller and person called, if these are available in the ...

Early-stage, Midwest, startup gearing up to produce SIP wall, floor, and roof components.

Installation eines Jitsi Meet Servers auf bereitgestelltem Ubuntu Server. Zugangsdaten stehen bereit Grundkonfiguration Jitsi für Browser und App-Nutzung Nach Rücksprache: - erstellen eines SIP Accounts - Setup FreePBX mit IVR Einwahl

Telecom billing and routing software ASTPP installation and setup for community and enterprise license - Full setup with sample customer. -Gateway integration -SIP proxy. -Reseller account. -Payment gateway etc.

We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX.

We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX.

HI we are after a freelancer who can help with this. When you put our company name in to Google search bar 'Force Media'....google will give a predictive suggestion 'Force Media Reviews' This will bring up a trust pilot review, for another company '', which is not out company, however, prospective clients of our will not read in to it too much and think that the review is about us. Our company is Force Media UK Ltd. We need help in removing this. I know, loads of companies will say they can do this, but, i'm not interested in just handing over money for 'No guarantee results'. Please comment on what you can do and guarantees.

I need a cloud FreePBX system to manage my sip numbers. In the meantime, I need to use database to store all of the CDR (including incoming and outcoming, etc.). Moreover, the instant incoming calls need to show to a webpage on my front-end app, and the response from the app (such as answer, hang up, hold on) will send back to the Asterisk.

Buscamos un perfil que cree contenido referente al contact center, debe tener conocimiento de dicho sector, customer experience, employee experience, software de contact center, trunk sip, centralitas, departamentos de contact center, etc. Que sea de idioma español nativo. Conocimientos del contact center. Queremos que este con nosotros mensualmente, hay mucho que redactar al mes, artículos, cursos, grabar podcast, pdf de contenido para su descarga. Tenemos 7 lineas de negocios y cada una de ellas es de una rama de contact center.

Hi I am looking for a development specialist for Asterisk PBX (I work with vitalpbx based on Asterisk) for the purpose of 3 developments 1. Press 2 call 2. Connection to the CRM system 3. Connection to the ZOHO system Regarding section 1 Sends the system API:

Hi I am looking for a development specialist for Asterisk PBX (I work with vitalpbx based on Asterisk) for the purpose of 3 developments 1. Press 2 call 2. Connection to the CRM system 3. Connection to the ZOHO system Regarding section 1 Sends the system API:

...receive speech packets and data. 3. We need to make certain enhancements to the interface and communication between the UniMRCP plugin and the tone / speech detection library we own. 4. This project will require you to work about 12-16 hours a week for 6 months. 5. Skills: (a) Pro in C/C++ (b) Familiar with UniMRCP plugin development architecture. (c) Telecommunications Engineering, Asterisk, Socket Programming, System Programming IVR Software, gRPC, Websockets. 6. Deliverables (a) Fully-functional plugin in executable form along with libs. (b) Product must be containerized. (c) Complete source code of interface. (d) No GPL code to be used (e) Documentation: (a) User Documentation or README file to setup/run, (b) Design and source code documentation. ##Yo...

Hi, I want to start saving data from different sensor types into a Neo4j Graph DB and try visualizing the relationship between the data to be able to work out a kind of Predictive Maintenance in a near future. I need someone with expertise on Neo4j to help me along implementing this project. I will be later developing a .Net WebApp consuming this database.

We experiencing the issues with audio in IVRs and MOH under high load(300-400 concurrent calls). They are breaking up and very choppy. The CPU load with 400 calls is around 50-60%. There is no issues with about 150 calls. The server details are below: CPU: Dual Xeon Silver 4309Y (16x2.8GHz) RAM: 32 GB SSD: 1TB(RAID 1) Asterisk: 16.21.1 Front End: Issabel It should be enough resources to handle this load. We need someone who can analyze the issue and provide the solution for it.

...use the available DTMF commands already available for routine conference calls. We have the technical documentation and source file samples available, these code samples are based on Asterisk, and they give us an overall view of how to develop the adaptor and allow to interconnect the Audio Reservationless Bridge/Platform to Adobe Connect. The Audio Reservationless Bridge/Platform already working and has been tested. This API/Interface/Adapter will be considered IP and owned by us. See the attached drawing for additional details. We are seeking an experienced developer intimately familiar with Asterisk. You must beable to start on the project ASAP. Thanks Tom! ...

Hi, I want to start saving data from different sensor types into a Neo4j Graph DB and try visualizing the relationship between the data to be able to work out a kind of Predictive Maintenance in a near future. I need someone with expertise on Neo4j to help me along implementing this project. I will be later developing a .Net WebApp consuming this database.

...include integrations with external APIs and the ability to create and sell products through a WordPress module. In addition, the software should include stand-alone modules for live chat and SIP communication, a learning management system, and appointments. There are several additional features and functionalities that are outlined in the attached document. It is important that developers carefully read and understand the entire document before making a bid on this project. The document includes details on the Master SaaS, Invoicing, Safe Mail, Modules, Stand-Alone Modules, Templates, Accounting, Chat and SIP Client, and Onboarding Wizard, as well as using external APIs and storage. Developers should have experience in building software solutions with similar features an...

hello! i’m looking for a drawing outline for a canvas. this is for a paint & sip event so i just want a black and white outline. i’d like something like someone wearing jordan 1 shoes and a little bit of their legs with a sock on it with a K logo which is attached in the attachments

hello! i’m looking for a drawing outline for a canvas. this is for a paint & sip event so i just want a black and white outline. i’d like something like someone wearing jordan 1 shoes and a little bit of their legs with a sock on it with a K logo which is attached in the attachments

hello! i’m looking for a drawing outline for a canvas. this is for a paint & sip event so i just want a black and white outline. i’d like something like someone wearing jordan 1 shoes and a little bit of their legs with a sock on it with a K logo which is attached in the attachments

I need someone to setup a simple javascript control of SIP (phone calls) using I have a SIP trunk with Telenyx.

...leases, to identify key terms and highlight any potential issues or areas of concern. The tool will use machine learning models trained on a large dataset of legal documents to accurately identify relevant terms and issues. This functionality will be helpful for individuals or organizations who want to ensure that their legal documents are accurate and complete. Predictive analysis: The platform will include an AI-powered predictive analysis tool that can analyze past legal cases and decisions to make predictions about the likely outcomes of future cases. The tool will use machine learning models trained on a large dataset of legal cases and decisions to make predictions about the likelihood of certain outcomes. This functionality will be useful for individuals or organ...

Hello, goal: make BLF (Busy Lamp Field / Presence) with VitalPBX (Asterisk) work for Linphone. Let's discuss via chat. Thanks, Christian

i want a website for investors it should contain sip calculator, online investment and some thoughts

Im working on a predictive maintenance project. I have sensordata (about 0,6 gb) from liquid gas pumps. I need to extract knoweddge about the state of the pump. i dont know whats possible based on the data i have. (Maybe the remaining useful lifetime of the pump, or a chance that the pump will fail in a certain week would be helpful for maintenance). In detail: Visualize the pressure/temperature/electricity of each pump over the time entire period in fancy graphs. Also aggregation of maybe 1d/1week/1month/1year. Figure out which model works best for my case, then Build ML-model split in train/test. Then visualize comparison between actual data and forecast/prediction. Then plot the feature importance. Maybe shapbeeplot? The data will be provided when the deal is settled.

sip softphone development with source code and CRM integration

This project is about customising Vicidial & Asterisk applications. + Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required

Identify the relation between the values of k and x,y. Write a formula for the transition from x,y to k. Summarize the whole work!...points with the first 50 k and the last 50 k with their coordinates, which these points produce. The main areas that fall into the work are mathematics, geometry, cryptography, statistics, programming and information protection! The reason for this is that the ECDSA function is written according to the latest developments Modeling an algorithm. Please, only high-class experts in identifying predictive functions with powers for the test and more training epochs. 4.5 billions row The file itself: a little preface.

I have the list with a multiple choice test. The problem is that the right answer is always placed first among the three possible answers. Furthermore, the answers are numbered and the right answer has an asterisk, as well as always being the first one, the right answer. I would like that the right answer be randomized among the three of every single group, to eliminate the numbering 1, 2, 3 and remove the asterisk. You can give me the new file in pdf, word. Google sheets or excel. You should not touch the questions. I attach the file. Thanks.

- Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a separate android phone will have to connect to the SIP server that was developed on the other android phone via wifi and an incoming and outgoing call would have to be demonstrated. Note: please read first & check android condition, if need root version use i thinking it. First you show me demo then add money freelancer website.

Hello, I would like you Caller Id asterisk sip that can be used anywhere in the world. Thank you

We are looking for an appointment setter for our ISO (independent sales organization). Our pitch is extremely simple and easy to adopt. We pay on a pay per lead basis. We provide the dialer (predictive), the VoIP and the data. Furthermore, we are looking for people with experience in the telemarketing industry and in particular appointment setting for home improvement companies.

We are looking for an asterisk programmer to create a module to manage automatic outgoing calls campaigns using TEXT to SPEECH automation in ISSABEL or FreePBX. The manager user must upload a CSV file that constains all the information of each customer to be contacted and when the call is answer the system has to reproduce the audio file in spanish with the specific information for this customer, it is not reproduce the same audio to all users. The module must let the user to generate all the reports to know the results of the campaign (Contacted, not contacted and time of audio reproduction, etc)

We are currently working on VOIP project, the GUI part of our project is completed. We want DID Number, Ring Group, Call Queue are automatically implement from our GUI to our asterisk backend.

...marketing mix, capital budgeting, · Portfolio optimization · Decision Making under Uncertainty · Simulation · Introduction to @Risk · Types of problems: inventory management, capital investment analysis · Market share estimation, sensitivity analysis Decision making under uncertainty. · Descriptive Analytics/Visual Intelligence · Predictive Analytics · Proactive Retention of Retail Customers through Sales Prediction using Linear Model, · etc. · Credit Risk Modelling using Logistic Models · Demand Forecasting (Time Series Models) and Inventory Optimization · Prescriptive Analy...

I use 3CX startup. I want to add a sip trunk for calls in Belgium. I have problems with provisioning my Yealink W76P (W70B & W56H) and use it as a router phone.