Asterisk wallboardJobs

I'm lookin for of a skilled engineer to manage and deploy a SIP & Kamailio on docker compose that also requires some admin services. I need these services done on a part-time basis, but the engineer must stay available and be ready to help should any issues arise. This means it is essential that the engin...environment (not final, but wished): - Ubuntu 22.04 - dockerized - in a later task, to enable kubernetes for Kamailio and HA Good entry points: Budget? will not be disclosed. So place your best hourly rate!

I'm looking for help resolving an issue I am having with Asterisk on a Linux server. I'm unsure of what caused it, but it needs to be fixed urgently. The ideal freelancer for this project should have expertise in System Administration for Linux platforms. I'm open to suggestions on how to diagnose and remedy the issue, so if you think you can help, don't hesitate to reach out and make your case. We are getting a lot of chan_sip.c:4413 __sip_autodestruct and have tried restarting the server once but without any positive results.

...routing. We do have an existing infrastructure which needs to be connected to, so knowledge of this process is needed. We are also interested in advanced features such as call quality monitoring, packet capture, security measurements etc, analytics etc. There is a billing system which is a sort of SBC itself, but very small and only intended to bill call termination. Also, we have a bunch of Asterisk servers spread all over the internet acting as PBXs and sort of B2BUA proxies/gateways for the rest of infrastructure. The idea of this project is to build a geo-redundant system having multiple PoPs worldwide to hide our VoIP core and become invulnerable to L4 UDP/TCP DDoS attacks by hiding key elements of the real infrastructure. This system should be implemented on either Kama...

I'm looking for experienced mentoring services to help setup and integrate an existing SBC system with Asterisk. I need assistance for 1-3 people with setting up the system, configuring SIP trunking, and ensuring that the integration with Asterisk is working properly. With the right combination of assistance, knowledge, and experience, I'm confident that this project will be a success.

I'm in need of a skilled engineer to manage and deploy a SIP & Kamailio & Asterisk solution on-premise that also requires some admin services. I need these services done on a part-time basis, but the engineer must stay available and be ready to help should any issues arise. This means it is essential that the engineer is experienced with the SIP & Kamailio & Asterisk solution and be comfortable dealing with any issues that may arise. The engineer must also have experience in deployment on docker & docker-compose and later on administration of this system. This position will require some extra work every week, but I'm ready to discuss the amount of work needed with the chosen expert in order to get the best possible results. We usually di...

Hello! I am looking for a freelancer to help me set up a VoIP system using Asterisk and a Skype trunk specifically for external calls. This setup is for my personal use, not for an existing or new business venture. The setup should be able to handle a high volume of external calls with clear voice and quality. I expect a complete solution in terms of setup and installation of all the VoIP components to ensure that I am connected and ready to place external calls. I am essentially looking for someone who can configure Asterisk and the Skype trunk with good quality, so that I can get started making external calls. Thank you!

Need to integrate Asterisk PBX into Zendesk. Using Asterisk AMI and POST API into Zendesk. When a call comes in on a specific ques ticket must be created in Zendesk using caller ID as a requester (if the customer number is not in Zendesk).

I have issues with the production server so I am looking someone to fix it asap. I have 2 different applications assigned to the the same carrier and carrier transfers calls simultaneously to both the ip, but right now as the call reaches fusion its disconnecting the call not allowing it to get transferred to asterisk server.

I have a linux server running asterisk running Chan_sip which only support UDP, and can not be upgraded. The asterisk Server communicates to Phones and SIP TRUNK Providers on UDP port 5060 using the LAN Public IP X.X.X.X interface. I need to have some new phones connect to the Server over TCP Port 5062 I need to have some new phones connect to the Server over TLS Port 5089 We will need to have kamalio or Opensips centos 7 to run on the same server as asterisk, and should allow the following: 1) The Proxy should communicate with the asterisk server on UDP SIP port 5060. 2) Listen for incoming SIP TCP traffic on 5062 on the LAN IP X.X.X.X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the interface. 3) ALL T...

This is for a Asterisk and Zoho CRM integration. We need to receive and make calls on Zoho. Also be able to record, transfer, pop up a window with caller contact on Zoho.

API SDK module required with Asterisk using PHP and MySQLwith subscription and credit billing model for users to enable them to add their own SIP trunk and route their own sip accounts and use API through any of the SIP trunk they have to process their IVR instructions on given webhooks. Basic MySQL structure to start with. a) users table [api_key (unique), loginid (unique), pwd, trunk_id , user_type (admin/user) ] b) trunks table [trunk_id, username, password, ip/address , port, protocol, transport_type, did_numbers, status] and other fields and table that is required. Experience Required : Asterisk AGI PHP MySQL Reference API SDK documentation Calls will be made through the voip server using the rest API tht needs to be developed in PHP below is the sample how we will

Hi Pablo B., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I'm installing Kommo CRM in my company and we currently use voip offered by local telephone company. I'd like to configure our voip numbers on asterisk and then Integrate with kommo CRM. Can you do it?

I have 2 asterisk servers A and C A is running on Centos 7 and has a static IP C is running on Freepbx (raspberry Pi 4) and has no access from the outside. We also have B which is running OpenVPN and has a static IP current configuration is A -> B > C -> outside Trunk We need the following to be done 1. A SIP remote extension to log in to A 2. A passes the registeration via B (OpenVPN) and register as an IAX2 extension in C 3. SIP remote extension made the call, A will translate the call from SIP to IAX2 Another thing, you would need to access using Anydesk via my PC to access both A and C due to VPN. Thanks

I need someone who experience in Linux and script development knowledge for Asterisk /freepbx patching. I will integrate existing asterisk PBX to Kommo CRM. The Asterisk/freepbx V18 PBX and kommo are ready, but don't know how to run the script development as below link:

I am looking for a freelancer who can help me build a cloud PBX telephony software that is supported by Twilio or Asterisk, using the SIP protocol. This telephony software will not need to be integrated with an existing management platform.

We are having major issues with our vicidial. Channels not connecting and calls being burned and reps sitting waiting. We are losing money and employees daily and need someone to help us get dialer running smooth asap.

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.

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)

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'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.

Se requiere free lancer para desarrollo de funcionalidades y aplicaciones en Asterisk

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 ...

...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

Quero construir um sistema web para gerar relatórios e tarifar ligações dos bilhetes gerados pelo asterisk.

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...