Asterisk web sip jobs

Would like to ...to use a raspberry pi Zero to run Asterisk with chan_bluetooth module a a openvpn client a 3g usb dongle for internet Your task is to create a bootable image which I will plug into the rbpi and it will run the 3 things When device boots it will 1. connect to 3g internet network 2. connect to vpn 3. load asterisk with chan_bluetooth

...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud

Need sip calculator script written in php

...page is to be seen or the program is to terminate. In addition, parts that have quantity on hand values that are equal to or below the reorder point should be flagged with an asterisk. Write this program as a C++ program using structures that have bound methods, functions. Write a structure, **struct card**, that will represent a card in a standard deck

I need a new website. I need you to design and build my online store. Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, busines...Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, business loan, gold loan, car loan - investments, fixed deposits, insurance, mutual fund, SIP.

The business is a provider of cloud based communications which includes: Skype For Business Microsoft Teams Cloud Contact Centre Office 365 Sip Trunking Hosted PBX We require assistance in developing and revamping our website content, feature guides, product bundles, resources for customers (How2Guides/FAQ etc) some of which we have but need updating

I have all details .i need to configure sip and configure asterisk server . VIP experts

We are looking for an expert in Globus toolkit 6. Configuring asterisk, postfix, red media server and QGIS in Ubuntu server.

A company is running Asterisk with an onboarding system. We are looking for an engineer who can support Asterisk and Kamalieo system in the cloud. Engineer is required to set up an onboarding system for clients and support them on daily basis. Here is the following job scope. 1) To set up and maintain an onboarding system that allows a user to sign

Hello i need to bulid sip filter to filter Retail Voip traffic

We are looking to develop a mobile a...integrate specific API from our middle ware platform. Initially the app will only need to be for android devices. The mobile will need to integrate with a telephony platform for sip communication and a messenger platform. The majority of the mobile devices will not have cellular service and will be strictly wifi.

outbound calls and select options automati, will send more info. about project .

hello, we recently had some work done by a freelancer on our bespoke asterisk voicemail application. however, customers are reporting various problems: connectivity, hang-ups, can't make changes, etc, etc.. : something is definitely wrong with what was done. we need an experienced troubleshooter to take a look and make changes to a live, working system

Looking for assistance in configuring Cisco call manager for sip trunking.

hello i want to make a dynamic filter with asterisk follow this features - the calls come from customer pass through asterisk filter to check if it's an spam call before to send to gateway gsm if the call is a spam, then the call no go to the gateway but if the call is a real, then the call pass normaly

Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones

...person to do some kamailio development for us. We would like to have kamailio look up the registrar domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE) comes in it is not

I need a very simple customisation of a Android based free / open source SIP client(sCIPsimple,Linphone or anything). I want the default values of my server to be populated in place of blanks in configuration forms. So, all you need to do is replace the values and compile and give me.

I offer VOIP telephony services and development of telecommunication solutions, I need a logo and a brochure for my site currently Im using logo which is n...logo and a brochure for my site currently Im using logo which is not mine. Send sample on your propoersal based on the description of my needs This is my site [login to view URL]

Hello. We are looking VOIP Tunnel software solutions to bypass voip calls in blockage area . It is encrypts/decrypt SIP and RTP packets and send voip traffic between soft switches without blockage . We are looking experienced guy who already developed or worked on this .

We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.

Hello I need someone to help me install whitelist on my asterisk white will reject all the calls with 503 and pass only the numbers in the list

Need to configure 10 internal accounts on IP phones, connect 3 external telephone lines. Set the voice greeting and the voice menu according to the technical task. Implement a CDR report and record conversations.

Looking for a developer who is very familiar with the Twilio platform design a web (NodeJS) interface for sub-account customers that will initially provide reporting capabilities on Twilio's SIP trunking product. Using Twilio's REST API, you'll build the following: 1. User auth 2. Dashboard with usage data (much like what twilio provides now, but for

I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a

I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any

Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !

Can you make android and iOS app for Asterisk client? Similar as Zoiper for iPhone.

Basically converting BICC to SIP and SIP to BICC using OpenSS7 or any other proved open source library. Performance and stability is an important factor

We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [login to view URL] regards

Need a2billing expert with proven record to do a fresh complete installation of a2billing (asterisk realtime, freePBX) and API ([login to view URL])

Create and train italian acoustic model based on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should...on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should be configured for telephone (8 khz), in order to integrate it in Asterisk with mrcp server.

I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step

...someone to handle the very few tech support calls I may get while out of town. Might go the whole week with none sometimes. You need to be very knowledgeable of Freepbx and asterisk in order to support my clients. I will on occasion need projects handled such as repartitioning, adding a logical volume etc... Trust is a key because I will be giving you

I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on Android and iOS app. I would like that the cost of project be showed in separeted Android and iOS and hope that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features

I want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows and Mac. And in the future on mobiles.

...experience in this field , this is simple problem. This is my result to run my code. ***********My code************ from pycall import CallFile, Call, Application call = Call('SIP/flowroute/18882223333') action = Application('Playback', 'hello-world') c = CallFile(call, action) [login to view URL]() *******************result******************** Traceback (mo...

Hi, I need an asterisk guru to help me acheive my goal. When someone reach my voicemail, I would like my to get a notification on my pager. That mean, system would have to dial a number, wait a couple sec, and dial 123 then hang up. This need to happen everytime someone leave me a voicemail.

Hi, I am looking for Asterisk PBX expert. The person should have through knowledge of how to install configure and maintain Asterisk on linux server. We have 2 projects for which we are looking for an expert. You will be working with me directly to estimate the project and after implementation you will handover the project and explain entire project

This absolutely takes a programmer. After doing all the updates needed for WebRTC and changing OS on Server...OS on Servers, not much is working any more. So the system has to be brought back from near death. This is a system with WebRTC and 3 physically separated servers, Apache, Asterisk, MySQL, plus a commercial ZBF router. Nothing for beginners !

hello, community I am looking for an engineer in networks for making a set up of an extension in a sip server remote.

dear all , i have pcap file which it capture from wire shark for my sip switch .. i want such script which will locate pcap file , and extract audio data from it .. it is RTP ... but i do not know the encode format (g711 or g722... ) the one who can have this project should first extract audio from sample i will send to him then we can start

I need build a sip app with vpn, sip and ssl, please contact for detail

...если все правильно и звонок пошел, то ему вылетит окошко, что вам звонят. Скрипт формирует USSD команду, исходя из введенных данных, форму которой я вам дам, логиниться на SIP сервер исспользуя Логин,Пароль и адрес сервера и отправляет комманду на сервер(делает вызов), отправляя эту команду происходит звонок с подменой номера, как только трубку поднимают

...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud

Company is looking vibrant developer who can build a SIP stack on a currently developing application. This facility will run on an existing app that is only available on Andriod. SIP stack must be designed to perform on single task. 1) Press a button to dial a fixed number on a SIP network.

Hello, I have installed the SuiteCRM and Asterisk system, I need to integrate these systems (call recording, connection identification, history of connections, etc.). How much will it cost a script and configuration?

...project named "VIBER WHATSAPP WECHAT FACETIME SIP GATEWAY", that you have finished it 90% can we talk about this and maybe you can develop this for me also? Hello, We need to develop a SIP to Viber, Whatsapp, WeChat and Facetime gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber, Whatsapp, WeChat

Basic router configuration (IP Address, hostname, access credentials)  SIP Trunk configuration from Cisco Voice Routers 1 and 2 to Genesys Virtual IP  ISDN PRI ports configuration  Dial Plan configuration  Inbound and Outbound Calls Testing

we did integration between zoho crm with yeastar pbx "u100", and it was working successfully, but if we received any calls through our queue "sales queue or technical queue" it is not displayed calling id name, it displayed pbx trunk number so we would to make a custimze code which enable us to knows who is calling.