Xmpp asterisk jobs

We are looking for someone who can integrate Asterisk in our Laravel system. For each incoming call, a pop up should open with the details matching in our existing database. For outgoing calls we should click the phone icon and call should be redirect to the customer number. You should have prior experience in Asterisk /FreePBX integration with Zadarma or Plivo.

Dear Freelancers, We have a video bridge jitsi installed. We need to achieve the following: Need to be able to login from a raspberry pi with a calling of url. The url call will happen if someone dial an extension on an asterisk pbx. Once the extension dialled from a SIP phone asterisk call a url which makes the raspberry online in the jitsi meeting room. From this time the raspberry is opened the meeting roon and participants can log in to the room. The raspberry pi need to work with a Konftel conference system . So the Konftel will be connected with USB to the rasperry and then the raspberry need to use its camera and microphone in the meeting.

On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. this module had to be compatible with latest freepbx 16 version. - Freepbx Asterisk distribution : - superfecta module to work with : fixed price : 100 $ - ASAP within 1 day.

i have laravel app that is managing user and apis . now i want someone who help me to in aws configuration for Laravel and SIP server asterisk ( user mapping with device)

I'd like the following to happen. A user registered to asterisk will dial the extension 1001. When they dial 1001, they will enter a conference room Another user registered to asterisk will then be dialled and forcibly put into that conference I like this to be done in Node.js, using Asterisk ARI to orchestrate as much of the above as possible. I'd also like the freelancer to share with me instructions on how to deploy asterisk, their config on asterisk and the node.js code too, so that I can recreate their experience locally. Be great for this to be done within the next 24 hours!

The project is a design and implementation of a doorbell using raspberry pi4 with python using asterisk (or another open source), zoneMinder (using ip camera), with 2 android mobile phone, at deferent flour. I need only the CODE of the asterisk (or another open source like: freepbx or fusionpbx), in orderto call and listen rasberry pi4 and android mobile phone, (flour number 1 or flour number 2). I also need a intercome between the 2 flour from each mobile android phone, at deferend flour.

We are looking for an experienced VoIP developer who can design Windows and MAC desktop VoIP applications using our Hosted PBX API. The application will have to be tightly integrated with our asterisk-based PBX and our custom API. Supported functionality will include: Voice calling via SRTP Searchable Call history with access to call recordings and call notes SMS and MMS messaging Read-only access to favorites and BLF keys Read/Write access to personal contacts Visual Voicemail Do not Disturb Call Forwarding We prefer a web application running installable with an Electron wrapper on the client's workstations but are willing to entertain other options.

Hi Need to design a sip based extension portal and forwarding portal. Looking forward to hear from you

On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. - Freepbx Asterisk distribution : - superfecta module to work with : price : 50€ fixed - ASAP within 1 day.

Callback widget - Asterisk Hello! I want a price for coding Asterisk to work with a callback function. Simply explained, we will publish a form on a website where customers can request to be called within a minute. When the customer has entered his phone number, it must call one or more pre-entered numbers in the Asterisk telephone. When one of the people being called answers the phone, a voice recording should be played: "A customer wants to talk to you, click 1 to confirm that the call should be connected" When the called person has clicked on the button to confirm that they want to talk to the customer who wants to be called, Asterisk must call the customer who wanted to be called (on the phone number that the customer specified in the form). ...

I need a voice bot for accounting office in greek language. 1. The client say who agent want to connect on 20-30 key words the call go to specific agent or group

We have an open spot for a professional interested in joining an amazing project that takes responsibility of the Design and implementation of end-to-end 911 Call processing! If you are an experienced VoIP developer with strong expertise in C, C++, Asterisk, and some background with JavaScript, this is a great opportunity for you! Candidates should have the skills and experience to quickly integrate the development team and provide design direction. About us: SofTech Consulting is a visionary tech consultancy focused on transformative cloud development. Leveraging 10+ years of high-level cloud architecture experience, the latest technologies, and an extensive partner network that includes Red Hat, Azure, and other top software providers, our specialized team of advisors helps ...

A chat application is required to develop using XMPP protocol.

Hello Developers i need a python developer to make a simple python app to send authorization request to a list of jabber users . example : jabber_authorization_sender@domain .tld:jabber_sender_password_here:jabber_user_to_receive_authorization_request@domain .tld

...structuring: ___ DB structuring & Maintenance: ___ API (creation): ______ API (integration): _____ Hostgator Server Set Up: ___ Hostgator Server Security: ____ Hostgator Server Maintenance: ____ Linode Server Set Up: _____ Linode Server Security: _____ Linode Server Maintenance: _____ Managing Linode access with SSH keys NOT with passwords: _____ Laravel:_____ Vue JS:____ React:____ Github: _____ Asterisk: ______ Trouble Shooting: ____ PCI Compliances______ Please add anything else you would like us to know about your skills and be honest so that we know what projects are best for you as we have many + What is the best per hour rate you can give: ...

The project is to create a service running on Debian operating system. The service should connect Asterisk PBX, open a socket on port 7891, and send the output of the following Asterisk events: - Inbound Calls: Call waiting in the queue, Call Ringing, Call Answered, Call Hangup, - Outbound Calls: Call Waiting, Call Answered, Call Hangup. The output of each event should be in one line only (Not sending few outputs for the same event). The project includes also a client that should run on a pc, connect the remote server on port 7891 and receive the server outputs.

Expert needed in Asterisk Issabel PBX hosted on AWS as part time

Need someone who has expert in Asterisk Issabel PBX server on AWS. We would need as a part time everyday few hours.

I have asterisk, I am looking for someone who can configure Flexisip and on ubuntu 22.04 for push notification with mobile linphone

Look forward , expert connect my pc and install Asterisk installation, VOIP configuration in Ubuntu 20 server. pls bid if u r available instant start. Thank you

We need the following project to be done. You should havve experience with SIP and Asterisk. The setup is planned an debian Linux. - Setup Asterisk Server on VM - Connect to 3 SIP Trunks - Setup a rules set that connects a database and forwards rhe calls based on the databse info - Setup a director for incomming calls - like > Welcome Text - based on incomming phone number redirect to certain department (call center style) - Add API for CRM > Funktions: > Start Call > Drop Call > Forward Call > Forward Call accompanied (menas speaking to the person before) > Callbox for CRM Users > Call Log to Database - short documentation and all API JSON Calls wit Syntax description We will also listen to alternative setups if you have suggestions. Importnat...

...to build themes on Joomla 4) ############################### ############################### 2) THE TASK You will be updating a Joomla 3 website to Joomla 4. Joomla 3 website is here su***pfo***rt [DOT]com And the cloned Joomla 4 website which you will be debugging to make it work like the Joomla 3 version is below. Here su***pfo***rt [DOT]com/sup***fort***demo/ Obviously delete the 3 asterisk to write the URL inside the browser and see the websites. ############################### ############################### 3) PROVIDE REPORTS AND HOW TOs IMPORTANT: I will give you access only to the cloned demo Joomla 4 site and not the live website Joomla 3 Every change you make has to be recorded meaning you will write the solution on how you fixed the issue. EXAMPLE If you ...

I am working to develop Asterisk Server. My Asterisk server has to integrate Azure Speech to Text API and OpenAI chatbot + Azure Text to Speech API. So If I call the special number, I can talk to AI I need Asterisk python AGI code.

Hi, we are looking for freelancer who can set up asterisk that have mass call feature or power dialer and the payment is based on project not hour

i need some python scripts for my freepbx asterisk voip system

I am trying to explore the possibilities of installing a SIP server or SIP GSM Gateway into an android phone. Using a remote sip client, the sip server would be able to authenticate remote sip client using username and passwor...Gateway into an android phone. Using a remote sip client, the sip server would be able to authenticate remote sip client using username and password. Once authenticated, when sip client request a call out. Sip server will trigger a call out using the android’s SIM card. Or Android SIP server will report itself to a VoIP switch like asterisk. SIP client will also be connected to asterisk. When there is a request to call out from SIP client via asterisk. Asterisk will treat sip server as a trunk and thus making an outbound c...

I am trying to explore the possibilities of installing a SIP server or SIP GSM Gateway into an android phone. Using a remote sip client, the sip server would be able to authenticate remote sip client using username and passwor...Gateway into an android phone. Using a remote sip client, the sip server would be able to authenticate remote sip client using username and password. Once authenticated, when sip client request a call out. Sip server will trigger a call out using the android’s SIM card. Or Android SIP server will report itself to a VoIP switch like asterisk. SIP client will also be connected to asterisk. When there is a request to call out from SIP client via asterisk. Asterisk will treat sip server as a trunk and thus making an outbound c...

Install asterisk on server ubuntu 20.4 and connect to account SIP

voip configuration to make calls using asterisk and google voice

Well established online "advice site"- needs to complete our Asterisk PBX system build: Minimum of 3 years previous experience preferred. Proven Experience with Asterisk 13 or later, Apache, Linux, PHP, (Laravel) and MySQL required. Knowledge in TCP, SIP, RTP, UDP protocols. Experience and advance usage of FreePBX or Elastix. - Dial-plan programming in AGI, AMI, REST / ARI ( in PHP ) - IVR programming in AGI, AMI, REST / ARI ( in PHP ) - SIP trunk & Gateway integration with third party VoIP providers Experience in REST API development and websockets. Knowledge in & paypal payment gateway Knowledge in Git. Strong troubleshooting skills a must

The main part of a project is a creation of software which will help to manage GoIP equipment (Hybertone manufacturer - gateways and sim banks). It should be probably based on a softswitch (Asterisk is NOT OK here). All the features which will be included should be available in the interface - Some basic features are: 1:Simulation of Human behavior -Generation of the flow of incoming calls -The list of 'preferred' numbers. -Imitation of SIM card movement around the city. -Generation of the flow of incoming SMS and USSD requests. -Implementation of daily and weekly cycles of human activity. -Time slots for activation and time slots between calls. 2: SIM cards Operations -The IMEI substitution using an actual base of devices. -Black lists and white lists of phone numbers t...

I need to install and configure asterisk connector on my suitecrm server. the link to asterisk connector is

This project is for my favorit efrelancer.

...just saying "Israel is bad" (example below). If you're interested in the job, please contact me with a link to your existing work and a short description of your point of view regarding the Palestinian cause. Example: first pic herzl and weizmann weizmann: so which land do you think we should colonize? [asterisk] herzl: im not sure yet, maybe argentine or palestine [asterisk] 2nd pic weizmann: let's go for palestine, since god promised it to us herzl: you believe in the torah? [asterisk] weizmann: i dont, but if we can get rabbis to say that, we can rally more people around it 3rd pic killing, destruction, ethnic cleansing in palestine, with herzl and weizmann toasting to it in the background footer link to herzl's "argentine or p...

We need to connect our CRM with our phone system in place. We need integration between Zoho CRM and WAZO (Asterisk 15) Features: - Click to call from Zoho Interface: callback user extention + keep time duration - Pop up when incoming call from WAZO and open customer record or if customer is not registred, open record creation and keep call duration It worked with ZOHO Phone bridge before annd it worked proprely but ZOHO CRM stopped the service few years ago. Thank you

Asterisk switch to WhatsApp and Telegram voice termination, including routing of calls real-time and database management. Must be able to speak English and communicate effectively.

Hi; I am a private IT enthusiast, who is interested in learning more about IT. I have recently come aware about Asterisk, Kamailio and Opensips. I would like to build a lab using virtual machines to see, how all of this works. What I am looking for is a kind of tutor, who could guide me through the steps to build this lab and get it working, while explaining to me tha essential things that I need to know. I am not a professional, so I do not need much details, even though sometimes I would have to get some. If anyone is interested in helping me building this lab once with Kamailio/Issabel and Kamailio/FreePBX and also using Opensips - connect it to Jitsi or MS Teams, do not hesitate to contact me and tell me, how much would it cost. Thank you.

Sip Refer To from remote server to asterisk tought kamailio Tengo problemas para manejar un Refer To remoto desde un proveedor hasta asterisk. La respuesta del servidor siempre es 481 Call / Transaction Does ...... .

Asterisk PBX and VoIP expert required. More details in chat.

We need an Asterisk 11 dialplan context that: 1. Executes when dialing any number from the extension 1005 and also from extensions 1070 through 1078 and 1081 through 1089 2. Blocks the call and plays the "number-not-in-service" playback file if the number dialed by the extensions above is equal to the result of the query "SELECT mobile FROM vtiger_contactdetails WHERE firstname REGEXP '^[0-9]+$' and mobile = ${EXTEN: -8}". The query is done to a localhost MySQL database

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

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

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.

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.