Asterisk web sip työt

Hi I have a fusion pbx, I need a dialing plan that I will contact with a did number and every call will reach the microchip on my computer. I also need help setting up a micro sip If fusion pbx is not familiar to you. And you have an offer of another platform that you can set up and quickly. So it doesn't matter to me - the main thing is the result as above

... you can use this app job is for configuration ----- 2. you will place the app in an android 10 non rooted using ADB in a folder /system/app to make it system app ----3. you will give it set up an app a accessibility services ( ----- 4. configure app with permission CAPTURE_AUDIO_OUTPUT --- 5. connect the sip client to a sip server ( i will give you user-- password- ip ) ---6. configure the audio uplink and down link all the codes and methods are in the doc 7. in order to test it we will have to make the android where the app was configured to make a call locally and since the sip client in the app is connected to our sip server we will be able to hear and talk via the sip to the person that the ca...

...project is to create a SIP to WhatsApp gateway that will allow voice calls to be passed from SIP to WhatsApp, ensuring that the call reaches its intended recipient. The development platform or operating system is not a concern, as the project can be completed using either the Linux/Windows WhatsApp executables or the Android/Windows Phone mobile versions of the application, regardless of the version number. The gateway must return appropriate call error codes to the SIP backend, such as CALL SUCCESS, BUSY, UNAVAILABLE, etc. To determine the success of the project, we will choose the solution that consistently triggers successful SIP/WhatsApp calls. Step-by-Step Process: Incoming calls will be directed to the WhatsApp gateway. The gateway will then convert...

Please bid only if you can do it and only write your exact bid for the project. I want a simple light java gui or a web app that will show the status of the asterisk channels. It’s gonna be like a table showing in the first column the channel name and second column the status (up-down-free-dialing-ringing-talking…) Also next to each channel we want a button that will block or unblock a channel. The person doing this must have great knowledge in Asterisk AMI. The gui or the web app can show channels from multiple asterisk servers and add them in 1 table for simple monitoring.

Hi, This is intended to be an integration between Vanilla Asterisk and ZOHO CRM.

This main.g...blob/master/examples/playfile/ This i need to send receive audio data to websocket and websocket data pass back to asterisk. Now code is :- asterisk sending audio stream data to go program and go convert wav file in send back to asterisk before change the chunk size 320 and 16-bit, 8kHz, mono PCM (little-endian). Need to change:- create a new websocket client and client audio strem data sent to asterisk via go. asterisk side receive audio strem data send to websocket client side . AudioSocket is a simple TCP-based protocol for sending and receiving realtime audio streams. There exists a protocol definition (below), a Go library, and Asterisk application and channel interfaces.

need to install asterisk 11 with unixODBC with php and mysql i can do php and mysql part you job will be install asterisk on centos 8 make sure our server have to work asterisk with database after install i need complete doc i will award and pay only after result dont ask me before to pay if you can please bid

I need someone to build hardware prototype GPS device using module NRF9160 SIP (Nordic Semiconductor). It needs to send data to server using internet from SIM card, also battery level and accelometer data from sensor. For more information please contact me.

We want To create a chatbot for taking bo...incoming calls and enable users to make bookings over the phone We want to use AI NLP Here's how it can work: Customer calls the Twilio SIP number. Twilio SIP receives the call and forwards it to your conversational AI application built with Wit.ai. The conversational AI application processes the customer's voice input and generates a response. The conversational AI application sends the customer's booking request to your back end for processing. Your back end processes the booking request and returns a response to the conversational AI application. The conversational AI application sends the response back to the customer via the Twilio SIP call. ——————/——&mda...

Setup docker/ kubernetes based call center / contact center Inbound and outbound calling Experience in setting to similar contact/call centers, using WebRTC Experience of administering SIP / VOIP

With the current logo... * add an eagle to the logo * Have the eagle match the brand colors * add the word "Academy" to the bottom of "Impact and Prosper"

Hi I need a sip trunk for calls acd USA. I have almost no asr - but all the acd is legal and approved by the customers. If you are dealing with the subject and know of such a supplier or are one yourself, contact me. If you have no experience in this matter then please do not contact me

Hi I need a sip trunk for calls acd USA. I have almost no asr - but all the acd is legal and approved by the customers. If you are dealing with the subject and know of such a supplier or are one yourself, contact me. If you have no experience in this matter then please do not contact me

My interest is to configure voipbuster by sip trunk in issabel, in such a way that I can send the callerid from issabel

We are able to register successfully in but we are unable to get the data dynamically through our frontend using our AGI. Our Devlopment language is PHP and we using Asterisk as a soft s

Setup Vicidial Full configure with Telnyx or Pilvo VICIDIAL full setup with SIP trunk - we provide server.

I need a solution to resell voip. my provider gives me sip trunk then I want to resell ports to my customers that uses asterisk PBXs

Hi everyone, I'm looking for experienced VoIP Engineer to give us helping hand on real-time speech recognition system we are building. Our stack: Debian, Asterisk v.16-19, Kaldi + Vosk, Python/Javascript (node.js) The main problem is that we are unable to pick a real-time stream of RTP which is a plain UDP, transform it to Websocket data and send to the Kaldi server to recognize. We would prefer to get it done without Kamailio and RTPengine for now, just plain Asterisk possibilities like UnicastRTP, ARI etc. So we are seeking for experienced Asterisk engineer who can give us a valuable hint, share experience and/or write some code for us. Thank you in advance.

Preciso de um desenvolvedor REACT/ e NODE.JS com bastante habilidade de agilidade. Além disto, preciso de um profissional que já tenha construído sistemas com protocolo SIP. ATENÇÃO: APENAS DESENVOLVEDORES QUE FALEM LINGUA PORTUGUESA.

I have backend and Janus Webrtc gateway with SIP plugin. On HTTP request to backend API it handles Janus commands and returns session_id, handle_id and token. I need html and javascript to continue from where to initiate new call via SIP plugin or answer incoming call (Janus running in HTTP mode). I will pay 50 USD.

We are looking for someone who can install on our server and make it have astersk setup and working. Just need audio call option

I am in need of a grant to assist in the growth of my business. I am a small business who started in 2019. We offer paint and sip, crafts and other types of art. We have classes in our main location and travel to resorts, wineries, VRBO and other places. We have parties for bachelorettes, birthday’s, and any type of event. We need additional funds to expand the business in additional areas and locations, hire more instructors and add to the opportunities I have written a few grants but have been unable to secure one. I have my business plan and information ready, I just need assistance on putting that into a grant proposal.

Project : Dongle Panel Function : web-base hosted VoIP to GSM gateway is accessible from anywhere on any mobile device, it helps to terminate/originate SIP calls traffic using 3G Modems/Dongles and an active USB hub connected with your Linux PC. With the features of exquisite enclosure and high performance, specially designed to maximize cost . Dongle : meaning Huawei usb modem with sim card attached to Freepbx or issable End-User • Same Panel (As Attached Pictures) • Add Call Feature via panel • For all functions I supposed to choose which dongle i need to use . Admin-User • Control All users (Add,delete,rename,change password,list of all users) • Payment method : PayPal & Stripe (2 options will be monthly

potřebuji do svých stránek udělat sip klienta. Používám laravel, php a javascript. Takže nejlépe v JS. Potřebuji to promtně turbo rychle

...and to preserve the heritage that has been passed down for generations. At Qahwa, we believe in quality over quantity. We source only the finest Yemen Mocca coffee beans and roast them to perfection, capturing the essence of Yemen in every cup. Whether enjoyed in our coffee shops or at home, we strive to provide an exceptional coffee experience that honors the rich tradition of Yemen. With every sip of Qahwa coffee, we hope to transport you to the mountains of Yemen, where the aroma of coffee fills the air and the beauty of its heritage inspires us all. Join us on our journey to preserve and share the legacy of Yemeni coffee, one cup at a time...

Diseño y desarrollo de una aplicación Aplicación móvil OTT Windows – Linux – Android (todas la posibles versiones) -IOS, iphone, - IPN(las diferentes versiones) con las siguientes funcionalidades: - Cliente SIP para VoIP, con la opción de elegir entre servidores SIP - Cliente de chat (REST API), el chat debe ser capaz de enviar y recibir Audios, texto, emojis, imágenes, vídeos, ubicación y archivos - Soporte de videoconferencia - Notificaciones PUSH - Soporte de cliente VPN

Hi I have an ongoing problem from few hours. My server appears to be performing a portscan toward 'random' ip addresses and udp ports. The process that is sending those packets is Asterisk which was already at version 16.29 and I've upgraded it to the last version, 16.30 but the problem persists. I can't provide ssh access to the server but I can provide whatever packets captures you prefer. I'd like to know the source of the problem, if it's some virus in the office of the client or some external attack. Max 100 euros.

Hello I will need a mobile app for android to transfer calls from asterisk to skype (viber) and from skyp(viber) to asterisk...alone audio calls. I will wait your answer.

Hi, I have several asterisk 11 servers. I need: 1. Export inbound and trunks 2. Merge duplicate data. 3. Import data to asterisk 16/20 server. Notes: Code will be reviewed and approved. Code will be tested on tests servers. Developer will not have access to production servers. DM me for any further questions) Thank you.

Looking for a sip softphone for windows desktop with g729 codec ,call transfer and conference option. Should work behind nat. Stun and rtp ports should be auto filled and static. We will build with basic options now and will upgrade more , so bid with your best budget.

Hello Freelancers, I’m looking to implant live DTMF logging into my agents panel using asterisk. The DTMF will be read from asterisk server and will be shown in my agents screen in real time.

We need a Linux expert to install faxing in asterisk..

Hello all, I would like to capture live DTMF information through asterisk and make it appear on my agent login panel. Essentially all it will do is when a call gets connect and an agent ask the customer to enter a code through their key pad it should appear on the agents panel.

We have an extension which is registered to the asterisk server.. but when making a call the Asterisk returns 401. The task is to find out: 1: why this is happening 2: what we need to do to fix it 3: Make test call to prove fix We will only pay for a fix..

I have an Alpine docker image with Hylafaxplus and iaxmodem and asterisk. my phone provider is twilio. I try to send faxes, but there are a lot of errors. here are some errors. Unspecified failure to train with receiver No response to PPS repeated 3 times. No receiver protocol (T.30 T1 timeout) I imagine that there is a problem in the configuration, can you help me? thank you

What is expected • Availability outside normal business hours on demand. • Ability to create and maintain system documentation (policies, diagrams, etc.) • Strong knowledge of Windows Servers, Unix, and network/web/core subcomponents. • AWS, GCP backup, recovery and health monitoring practices. • Experience with PBX systems (FreeSwitch, FreePBX, Asterisk, etc.) What is good to have • Experience in managing Database servers (MsSQL, PostgreSQL, etc.) • Knowledge of scripting languages (PowerShell, bash, etc.) • Understanding of TLS/SSL and certificates chain use/distribution What is not required • Customer support • QA (Testing, bug tracking, etc.) • DevOps • User training Expected employment type: • Full-time ...

Can not send “@” in sms by Dongle Usb Example: I send: test @ I get: test ¡ What im missing in dialplan ? [textmessage] exten => 111,1,NoOp(SMS receiving dialplan invoked) exten => 111,n,NoOp(To ${MESSAGE(to)}) exten => 111,n,NoOp(From ${MESSAGE(from)}) exten => 111,n,NoOp(Body ${MESSAGE(body)}) exten => 111,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)}) exten => 111,n,MessageSend(${ACTUALTO},${MESSAGE(from)}) exten => 111,n,NoOp(Send status is ${MESSAGE_SEND_STATUS}) exten => 111,n,GotoIf($[“${MESSAGE_SEND_STATUS}” != “SUCCESS”]? sendfailedmsg) exten => 111,n,Hangup()

I need to Configure SIP trunk between the billing platform MagnusBilling and the MultiTenant PBX. I need to set it up so that all tenant PBX's are able to call out from the PBX and get billing individually by the MagnusBilling platform.

Here are goals for the modification, 1. Modification for both MacOS and Windows 11 platform. 2. Provide installation packages for both MacOS (.pkg with cert) and Windows 11 (x86 - msi/exe). 3. Rebrand Blink to our product brand (2 brands). 4. Modify login to use Web Services, this web services will be provided by us. a. Web services will providing SIP credential and phonebook. b. Web services will be using encrypted messages. 5. Removing unnecessary GUI and functions. 6. Synchronize contact/phonebook with cloud Web Services. 7. Change the color scheme of the app according to our specification. 8. You will need to handover the codes. 9. You need to prove the codes are working and can be compiled using virtual machines setup by us. 10. We wi...

We require someone who can integrate php agi in asterisk box to replicate the similar commands to control dialplan as given under below link here, To enable our customers to run their own IVR using API and SDK Architecture using Cent OS, LAMP framework, Asterisk and PHP-AGI User Accounts connects using API and run their own IVR business logic Our VoIP Arch -> Connected to multiple sip endpoints for each users on their account. I need some one who can help me setup an asterisk on linux machine probably in some better datacenter such as aws or azure that could run behind an proxy public ip for client server connection. And than use php agi to develop an sdk that could send and receive rest api and xml commands for controlling ivr dialplans

Need create notification for ippbx support ios and android (React Native) want to make notification to support app wake When the screen is turned off or the screen is locked when someone calls through the sip application, both ios and android

Hi We are trying to find out the possibility of developing a middleware (B) for our system. We have (A) a Voip Switch (Originator) (C) A VoIP provider (Terminator) (B) will be sitting in the center and 'listens' to each Ring Back Tone (RBT) when a call is established and 'ringing'. A--B--C Originator -- Middleware -- Terminator RTB frequecy will be based on standards according to Internation Telecommunication Union (ITU). B will reject calls when RTB frequency are not met.

Need some assistance to getting our app to work Incoming and Outgoing calls. demo like ctxSip

I have developed an automated system to make calls in asterisk, this is with a , the problem is the calls do not go out and I was investigating, the reason why the calls do not go out is because they do not include exten => _91XXXXXXXXX,1, AGI(agi://). the problem is that I don't know how to include it in the

* Modern Looking Softphone App. Flutter language to be cross platform, SIP coding, should include: - User Registration form; - Keypad to enter phone number; - Access to phonebook; - Separate page to send and receive SMS/MMS; - Show Balance; - Recharge Balance; - Group call; - Low Battery consumption for background activity of app; - Add users through phone number; - Indicator if other user is online/offline; - Ringtone selection; - Voicemail; - Call waiting; - Users can add multiple phone numbers after purchase; *Multi-Page App includes the following pages: - Softphone (keypad page); - Users page (where users can see other online/offline added users/add users, chat through SMS/MMS, or create groups to make group calls); - Wallet page (shows balance, users...

our company has an asterisk phone pbx connected to local carrier and additional voip carriers. we are migrating all our systems (accounting, sales, etc) under Odoo. We are looking for someone to configure Odoo VOIP module and connect to carriers.

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