Asterisk pbx pmsprojekty

I am looking to hire someone whom can prepare instructions for configuration the EdgeMarc 4806. We have many units to deploy and are looking for someone who can prepare a configuration guide. The guide must cover steps to configure the SIP-UA for the FXS ports and connect them to Hosted PBX service, as well as how to configure the T1 interface as a Network-side PRI interface for a fractional (8-channel) PRI with a phone system. The PRI should be connected to a remote SIP Trunk and include how to configure DID mapping/translations.

...billing platform. Have a redundant pbx and trunks Currently the way it’s setup is we buy a trunk for each client then pay per concurrent calls that client needs. We need a PBX that is of course Multi tenant, has amazing features, supports FoIP, ability to overstamp any caller ID we tell it too, ability to give client full admin access or restricted access and even staff access to their own extension. Currently we use Telco In A Box as our provider for trunks. We pay for caller per second. We need to have redundancy of trunks, and pbx, hosted on our own servers here in Australia. As many features as practically possible with an excellent GUI Automation is definitely very key here from customer sign up, billing and payment portal. We was looking at vital ...

Hello Guys, I have small issue the asterisk is rejecting call... due to Tel URI .... INVITE sip:+9714602XXXX@xxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP ;branch=z9hG4bK34zzdzv9zdt3bthz6bzh6d6zc;Role=3;Hpt=8e42_36;TRC=ffffffff-ffffffff Call-ID: asbcqd3893uim9dfu8m1bxokwmf3duu8tf3m@ From: <tel:0559625523;noa=national;srvattri=national>;tag=kx1t8kw1 To: <sip:046022600@xxxxxxxx;user=phone> can you help about this

I am looking to develop API for Grand stream / Open Vox IP PBX system using the Asterisk Manager Interface (AMI) or using the API document which is given by the Grand Stream (find enclosed) I need the following API Click to Call API Auto call forwarding to extension based on the mobile number Getting call logs data - CDR / CLI

we need to build a voip phone system. our own pbx and our whole system to run voip phone system to sell voip phone service,. everyone provide your quote

Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. My goal is. If a call is received within "5" second then the receiver will be considered offline but the caller will still listen the ringing tone. So the caller will think the phone is ringing when it's actually unreachable. Budget $50 Thank you.

NEED HELP TO SETUP VOIP SERVER - RUNS ASTERISK AND ALSO ONGOING SUPPORT

Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. My goal is. If a call is received within "5" second then the receiver will be considered offline but the caller will still listen the ringing tone. So the caller will think the phone is ringing when it's actually unreachable. Thank you.

Openvpn Setup and Connect the pbx

...Над чем будем работать? ✔️Есть самописная CRM написанная на php, js, bootstrap. ✔️Модульная система Smarty. ✔️Интегрированная Asterisk + OTRS. В основном используется для отдела продаж. CRM парсит много данных для обработки и подготовки баз данных для менеджеров. Поэтому опыт работы с парсингом обязателен. ✔️Также мы используем нейронные сети для анализа данных, в CRM встроен движок node js + TensorFlow + phpmlib. Если есть опыт хорошо, если нет не принципиально. ✔️Задачи, связанные с поддержкой и развитием системы. Необходимые знания: php; html; mysql; js; bootstrap; парсинг; MVC. Преимуществом будут знания: node js, TensorFlow, phpmlib, Asterisk, базовые принципы SEO. Условия: удаленная работа (частичная занятость); возможность голосового общения при ...

A dynamic regional Chicago-based Unified ...and NetFlow • Programming: PHP, Perl, Bash, Microsoft SQL, and MySQL • VoIP: SIP protocol, Asterisk, and Kamailio • Operating Systems: Windows fundamentals, Linux fundamentals, CentOS, and RPM management • Virtualization and Containerization • Good Communications Skills and Conversational English ABOUT DLS: DLS Internet is a privately owned Voice Over IP and Information Technology Services provider with primary expertise in Unified Communications and Network Operations. Our services include VoIP Telephony, Network Security, Networking, and Data Center Management. DLS operates two data centers in Chicago metropolitan area. Our software development team has been hard at work developing DLS Hosted PBX comm...

Project for Chynara to assist with projects for pms

Necesitamos un programador y diseñador de CFD para 3CX, para diferentes proyectos que tenemos en clientes de nuestra cartera que vienen trabajando con la PBX de 3CX, además una persona que pueda dar soporte a nuestros clientes experto de cualquier incidencia que tenga que ver con 3cx a nuestros clientes.

Hello, I've a few small businesses and publish posts and ads on FB and Linkedin. I need theme-oriented images for posts and ads. 1 example: we sell hotel management software (PMS) and run different promotions, like discounts, freebies, Back Friday and New Year discounts, and so on. I will need a specific image for each promotion. It's fine to have a 20% text on the image for FB promotion, in some cases, we will ask to have more texts on it. Per month I will need around 5 to 15 images (fb and linkedin). Per image, I will pay $5. If it's fine, bid pls.

...on SIP Providers and other topics. This project needs documentation, this means you will have to document all the configuration steps. I want to use AstLinux because it's stable and doesn't have a lot of moving parts. I'm interested in these two SIP providers but open to better suggestions: We can host this system on Cloud or on a PBX on prem. It's your choice. ----------------------------- Features I need for this project: - Auto-attendant based on day and night time - Redirect extensions to a different phone number, call forwarding - Request a phone call back - Background music - Conference call - Security hardening & latency **************************************** **************************************** Features

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 have multiple messages). The receiver can then press 1 to confirm (which will update the status on the database record too "Confirmed") The receiver can also press 2 (which will divert the call to another number and mark the status on the database record to "Pending") The module/development will need to be able to tell if the customer has answer or if its a ...

I have small issue the asterisk is rejecting call... due to Tel URI .... INVITE sip:+9714602XXXX@xxxxxxxxx SIP/2.0 Via: SIP/2.0/UDP ;branch=z9hG4bK34zzdzv9zdt3bthz6bzh6d6zc;Role=3;Hpt=8e42_36;TRC=ffffffff-ffffffff Call-ID: asbcqd3893uim9dfu8m1bxokwmf3duu8tf3m@ From: <tel:0559625523;noa=national;srvattri=national>;tag=kx1t8kw1 To: <sip:046022600@xxxxxxxx;user=phone> can you help about this

I would like a logo with my name Lori Kranenburg. The first and last names can be in different fonts but they need to be easy to read. I want the colours to be purple: HEX: #700460 RGB: 112|4|96 CMYK: 60|100|30|20 PMS: 2356 C and blue: HEX: #022B7A RGB: 2|43|122 CMYK: 100|92|23|12 PMS: 2147 C I would like the logo to include the image of a butterfly. Preferred dimensions are 200px by 200px for a square logo OR 600px by 200px for a rectangular logo

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 - Yealink Device Management Portal

Existing script grabs what we need. We should be able to call Polly () 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 xxx so they can be referenced in asterisk dial plan. - Time should also be parsed to use exisint asterisk sound files. SHould parameterize the language in case different languages are setup in the future.

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

Need someone to configure nec gateway with one line and 4 nec telephones

Custom developed / Mobile calling app similar to Zoiper , bria or x lite , or ICX that can handle IP pbx and also client connection via sip

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 mind if you take less for your first job and more like a paid interview and skill set assessment. My point is that we don't want to risk high amounts to test your skills. If you can prove yourself to be a RESILIANT team member then you could score a DREAM JOB with us.

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 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 using ODBC.

I am new to and looking to setup vidyo for video conferencing and vidyo gateway so it can be integrated with my PBX.

I would like to have the Wazo PBX installed in a Vultur Cloud Instance.

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.

I need you to design to integrate our PBX with opera PMS using Char so we can use the call accounting feature.

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 to the equation. *** IMPORTANT NOTE: Before commenting, please make sure you are able to write a short plan that you'll follow, and be provider-specific so that we can pre-check the provider of your choice. Another important thing we require is that you comment about which method you'll prefer (ANI-based unmask, or other method(s)). Please, comment only if y...

We are looking for a vtiger specialist. It is about the following tasks to be solved: 1. An exemplary vtiger 7.3 version hosted on our server can be functionally connected to our PBX (Voip Provider) so that we can use all telephony features there. Astericx is not necessary. It's just about the application. More information from our VOIP Provider (Cloud PBX) see link: 2. Task: The vtiger Crm has to be partially modified. E.g. the several pictures should be included in the products without any problems. The comments should also be able to be exported when exporting to an Excel file. Likewise, the setting of various cron jobs. Please indicate how many hours you need for task 1 and task 2 and give an hourly rate at. We are also interested in references from what

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 Sales Bar Chart: *** …. Enter today’s sales for store 9: 12000 Sales Bar Chart: ************

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

This is what is needed: - Create package to be used by synology package manager to run on latest DM (see wiki for more info: ) - Package includes: + latest installation of 3CX + 3CX should have access to local directories on NAS, so it can save recorded voice calls. Should be fine job for someone who: - is skilled linux system engineer - knows about synology NAS system and spk packages etcetera Regards, kevin

...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 (). As it needs some integration into Asterisk software, I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed

I would like to have the Wazo PBX installed in a Vultur Cloud Instance.

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 calibration on voice travel time so it matches the video travel time. The objective of this integration is to allow users to use their phone as the audio source for the video calls, if you know of a better way to achieve this I'm open to hear.

heavy duty door contest. You submitted a design with 3 boxes and a Rhinosaurus and "Formidable". I would like to hire you to create some additional versions. 1) change "security" to "DoorSafe. 2) for both version #1 and the current version, show using a blue (PMS 300) and a silver-gray coloring. Thus, you will have 4 total versions of the Rhino + 3 boxes.

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 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 app to my asterisk pbx so when video calls are made to the patient they also ring on his phone.

...devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI 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 vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to develop a power dialer, Predictive dialer, voice broadcasting, IVR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3. Good knowledge of Asterisk AMI 4. Familiar with sip trunking. 5. Familiar with email kernel Thanks....

...devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI 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 vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to develop a power dialer, Predictive dialer, voice broadcasting, IVR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3. Good knowledge of Asterisk AMI 4. Familiar with sip trunking. 5. Familiar with email kernel Thanks....

...devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI 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 vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to develop a power dialer, Predictive dialer, voice broadcasting, IVR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3. Good knowledge of Asterisk AMI 4. Familiar with sip trunking. 5. Familiar with email kernel Thanks....

...devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI 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 vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to develop a power dialer, Predictive dialer, voice broadcasting, IVR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3. Good knowledge of Asterisk AMI 4. Familiar with sip trunking. 5. Familiar with email kernel Thanks....

...devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI 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 vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to develop a power dialer, Predictive dialer, voice broadcasting, IVR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3. Good knowledge of Asterisk AMI 4. Familiar with sip trunking. 5. Familiar with email kernel Thanks....