Asterisk dialplanpekerjaan

Saya membutuhkan yg paham dan pengalaman utk melakukan setting/configurasi asterisk saat ini sdh terpasang akan tetapi masih mengkonsumsi cpu server boros sekali rencanaakan dipakai utk call center dg jumlah agent 100 orang.

I am after a module for Asterisk or any other open source PBX that will allow me to remotely request and automated telephone call to deliver an automated pre-recorded message. For example http://192.168.1.1/call=01623661624&message=1 This will insert into a database, the number with a status of pending. It will then call the number above and play the pre-recorded message 1 (as i plan to...

I need a php to query a mysql database and originate calls for numbers (result from mysql query) using AMI . i use a SIP channel to outbound calls with only 1 simoultaneus call, the MAIN GOAL is to get calls wait until channel available or retry if CONGESTION.

Looking for someone well versed in SIP/VOIP solutions to assist us in troubleshooting a few ongoing issues with our clients. The project will be ongoing, initially I'd expect 10-15 hours, and then on an as-needed basis. When applying, please provide your specific experience with the following: - SIP/VOIP Deployments/Networks - Bicom PBXWare (Asterisk based) - Yealink VOIP Devices - Yeali...

Existing script grabs what we need. We should be able to call Polly ([log masuk untuk melihat URL]) in the script. - Let's only update if there's been an update. - Rather than send on the fly, and stream, let's download the file from amazon and convert to wav (which asterisk likes). This should cut down polly requests and time for upload/download. File should be saved to xx...

I need to redirects calls in queue but if the caller press 1 I need to exit the queue and let the caller register a message

Looking for an ongoing FreePBX Support. Multiple deployments that need to be managed. Required Skills: FreePBX / Asterisk Linux SQL SIP Dialplans

Asterisk PRO? This is a potential for a FULL TIME OPPORTUNITY. RIGHT NOW, I need you to be ready to work 48 hours straight, well you can take a few hours in between but this is URGENT. I have some world leaders who will hop on board when its ready so send me a quote for your next 7 hours and based on your WORK EFFORT we can negotiate you continued employment on a more normal schedule! Keep in min...

Would like to take my calls from Voip Server through my asterisk server and put back on Voip server by adding an external IP.

I need help since asterisk endpoint ring but do not answer the call asterisk pjsip webrtc

i need do conference call between 2 user on asterisk

I am using a customized open-source platform based on Asterisk/AGI and PHP/MySQL. I also use WHMCS for billing and collection of payments (with Stripe). With a Cron job, my Asterisk Server sends the “Billable data” (i.e. excess minutes for Package Plans) to another server where I have WHMCS. My small portfolio of customers using Voip services includes: Residential, Business (includ...

I am struggling in getting a server running Asterisk to authenticate a peer using his IP address while using Asterisk Realtime Architecture. I need to get the fields name and content needed for ps_endpoints and all the involved additional tables to allow IP authentication of a peer. It will be a vanilla Asterisk install, no FreePBX or other, just Asterisk running along a MySQL database connected u...

I am new to [log masuk untuk melihat URL] and looking to setup vidyo for video conferencing and vidyo gateway so it can be integrated with my PBX.

We are looking to control Asterisk dial plans from our third party platform. As such we will require the creation of multiple dial plan functions for the creation of hunt groups, IVR, calling plans, call assignment and routing.

We are looking for an Asterisk implementation for unmasking blocked caller id and send an e-mail to the owner of the incoming number with the number of the caller. The project should use Asterisk as a PBX and any virtual phone number provider that can forward the original caller's CID and ANI information to the PBX. We would like to use local phone numbers instead of toll-free, so, add that t...

2) Write a Python program that asks the user to enter the sales for 9 stores. After getting each of the sales, the program will display a bar graph by displaying a row of asterisks. Each asterisk should represent $ 1,000 of sales. Here is an example of the program output: Enter today’s sales for store 1: 7000 Sales Bar Chart: ******* Enter today’s sales for store 2: 3000 Sa...

if you can connect vici to bitrix for my client , give me price and time . thnx

In my programming project I build a system around an Asterisk VoIP server. My purpose is to enable streaming speech recognition once inbound call occurs, i.e. I want to run automatic voice recognition since starting of conversation two people are involved into. The ASR (automatic speech recognition) engine I have chosen to implement that is Kaldi powered by Vosk server ([log masuk untuk melihat UR...

I am looking to integrate an existing custom video-calls scheduling app with Asterisk. At meeting start time the app will send a webhook to asterisk to initiate the call. Once the call is established on asterisk the video call should start on the app (via webhook). When call is terminated on asterisk call should be terminated on the video app (via webhook). Initially Asterisk may need some calib...

I want someone to help me to configure my asterisk over GUI or command line

i need to translate response codes from asterisk termination to origination for example i am getting sip response codes 480 i need to give origination 503

My client uses a Telemed solution that uses video calling via [log masuk untuk melihat URL] integration. We need to have the audio go through a hosted PBX so the patients can use their phones instead of complicated audio input devices for these video calls. So to summarize I need an integration between [log masuk untuk melihat URL] app to my asterisk pbx so when video calls are made to the patien...

We are looking for an experienced call center developer to build from the scratch. It needs to be for multiple users and ability to use as a predictive dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is t...

We are looking for an experienced call center developer to build from the scratch. It needs to be for multiple users and ability to use as a predictive dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is t...

We are looking for an experienced call center developer to build from the scratch. It needs to be for multiple users and ability to use as a predictive dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is t...

We are looking for an experienced call center developer to build from the scratch. It needs to be for multiple users and ability to use as a predictive dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is t...

We are looking for an experienced call center developer to build from the scratch. It needs to be for multiple users and ability to use as a predictive dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is t...

We are looking for an experienced call center developer to build from the scratch. It needs to be for multiple users and ability to use as a predictive dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is t...

We are looking for an experienced call center developer to build from the scratch. It needs to be for multiple users and ability to use as a predictive dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is t...

i need to translate response codes from asterisk termination to origination for example i am getting sip response codes 480 i need to give origination 503

I have FusionPBX which works on Freeswitch. I have discovered a bug in the system. 1) When normal external call is made from an extension number in fusionpbx, the sip header contains the extension number and DID number. This is fine. 2) When call forwarding is activated the extension number and DID number are no longer in the sip header. This means that the call cannot be authenticated at exten...

i need to translate response codes from asterisk termination to origination for example i am getting sip response codes 480 i need to give origination 503

I want someone to setup cloud pbx Asterisk 16 Issabel

I m working in voip by system asterisk based on chandongle and i need a program that i can change IMEI of dongles

I use open source VoIP software asterisk and it reads plain text configuration files. I would like it to read configuration files that have been encrypted with password using gpg or 7z. Sources of asterisk software here: [log masuk untuk melihat URL]

Hi we just create a new cloud server with Ubunto 18.04 version. We need to install DESKTOP GUI. After that we need to install WAMP one we have the webserrver we need 3 app Magento Open Source 2.4.2 [log masuk untuk melihat URL] Asterisk 18.2.0 [log masuk untuk melihat URL] Dolibarr 13.0.0 [log masuk untuk melihat URL] Once installed we ll need our tasks on this project .

This feature allows a caller holding in a queue to press '1' and enter a phone number to be called back at when their slot in line comes up next. We are looking to augment our team with a software developer who can begin with this project and possibly move on to work on other projects.

Plataforma telefónica en Asterisk que corra en Linux en donde se realicen llamadas de salida, ingresen llamadas y se tenga un marcador predictivo. Las marcaciones de salida deben poder realizarse desde una aplicación web.

Permanent fix of One way or no way audio issue on freepbx/asterisk (more of an RTP issue)

The current issabel [[log masuk untuk melihat URL]] distribution when installed with Asterisk 16 has support for multiple parking lots. There is an issue where the calls parked are always placed in the default lot, and never in the additional lots. [log masuk untuk melihat URL] We need someone to determine a resolution to this problem, and create documentation for how to fix it. We will then use ...

Hi I'm using JGRASP Overview In this exercise you are going to recreate the classic game of hangman. Your program will randomly pick from a pool of words for the user who will guess letters in order to figure out the word. The user will have a limited number of wrong guesses to complete the puzzle or lose the round. Though if the user answers before running out of wrong answers, they win...

I'm using JGRASP Overview In this exercise you are going to recreate the classic game of hangman. Your program will randomly pick from a pool of words for the user who will guess letters in order to figure out the word. The user will have a limited number of wrong guesses to complete the puzzle or lose the round. Though if the user answers before running out of wrong answers, they win. ...

Provide integration system in .perl for a vicidial server. Only experts in vicidial and asterisk nedded.

Hello, Our project is similiar to all those automated outbound dialers, except we need some customizations due to we are only looking to send missed calls (outbound calls to be disconnected after 1-9 second of rings or single ring) - we do not mind if you work with any existing platform or configure any other available software likes IVM, Asterisk etc to the task as long it can fullfil the follow...

Hello, Our project is similiar to all those automated outbound dialers, except we need some customizations due to we are only looking to send missed calls (outbound calls to be disconnected after 1-9 second of rings or single ring) - we do not mind if you work with any existing platform or configure any other available software likes IVM, Asterisk etc to the task as long it can fullfil the follow...

I need a shellscript that runs the following 7 commands and gives me output like here AABB 1 output AABB 2 output AABB 3 output AABB 4 output AABB 5 output AABB 6 output AABB 7 output There must be TAB spaces between AABB x and output cat /var/log/asterisk/full.1 | grep "CALLERID(name)=" | grep "AABB1-" | wc -l cat /var/log/as...

Necesito crear video y chat con webrtc para que se comuniquen entre navegadores pero también pueda hacer llamadas solo voz hacia una central asterisk pero necesito que me expliquen cómo funciona y la documentación

Hello, I am looking for an expert who can help me with sending VOIP pushnotifications for iOS (iPhone) devices using Asterisk / FreePBX environment. I am using Linphone SIP mobile client and I need you to help me configure VOIP Push service for the incoming calls. If you have experience in what I am talking here, then please bid on the project. Bid on this project only if you have experien...

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