Asterisk pbx projectJobs

Looking to build a setup where. 1.) UA /WEBRTC client registers to kamailio 2.) Kamailio is registering directly to PBX for all SIP. extensions. 3.) setup where Kamailio doesn't use RTP engine (not sure if i can) need to test config is almost there Need some assistance/tweaking. as i dont know.

Hi I have CCs Zycoo coocenter-S30 , need many modification for the following : 1. Change GUI for monitor user Interface ( add counter for the queue ) also Agent user (Arabic language) 2. add some math equation in reports (Arabic language) thanks

We have a basic setup ready with chan_dongle and M35 modules. We are able to run this succesfully with Asterisk. At times asteriks crashed intermitantly . Some time the gateway runs for 24 hours without issues. But some time it crashes with in few hours. We need some one with experience on Asterisk and chan_dongle who can resolve the issue for us

We looking for a VoIP / Asterisk PBX professional that is great at sharing knowledge. We working on a VoIP solution that will be integrated in our complex public transport system. Currently we have implemented a simple 1 to 1 voip call but we planning to extend the implementation. The features we want to implement are following: - Group calls (1 to many) - Custom headers for additional configurable options (mute all participants, call priorities) - Call Auto-accept - Calls history - Group calls rejoining We want to extend our VoIP / asterisk pbx knowledge and validate the approaches to some of the features described above. The responsibility of the professional will be preparation of one day virtual session (preferred Microsoft Teams) for 3-4 developers to ...

We are a wholesale distributor that needs a way to input a list of orders and automatically call the customer and play a pre recorded message.

we a Gsm Gateway software that use PCI gsm modules board as its Hardware we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module

Currently we have a gam gateway call management software , the software is working pci boards hardware . our project now is yo change the ( pci boards GSM modules) based interaction with , USB GSM Modules based interaction . The Developer modify our software to work with USB gsm modules , the developer need expertise in VoIP. Asterisk. Asterisk Modules. GSM Modules .GSM Chan , GSM Gateways, rSAP -> Remote SIM Access Profile, rSAP over TCP,,SIM Banks ,SIM Servers

Use webRTC or native language to develop VOIP dialer & call and messaging app for Android and iOS with the following features: Multi-language support: English, French, and Spanish User Registration: Language selection, OTP / OTC for user registration, etc Phone Contact Integration: Sync phone contacts each time app launches Chat with media:...French, and Spanish User Registration: Language selection, OTP / OTC for user registration, etc Phone Contact Integration: Sync phone contacts each time app launches Chat with media: text, images, emojis, audio, video, documents and other files Group Chats with media Audio Calls Video Calls Group Audio calls Profile management Notification management Message Search Option Star message Settings Asterisk Server integration Push Notification ...

i need filter on asterisk about did and clid number. We are using this project on maniplation temination voip traffic

...to answer a short survey and the call will be transfered to the survey module. The customer will be asked questions and asked to press the appropriate number key to indicate their response. Make IVR surveys form excel phone list. Receive answers from phone keys and voice recognition. Detailed requirements: * Must be delivered as a FreePBX module compatible with latest versions of FreePBX and asterisk * Must be configurable through a web page on the FreePBX web interface * Configurable features must include: number of questions, range of values for response * Results will be stored in a MySQL database with information on the call, the agent, the questions and the answers to the questions * There will be a web interface to display the result for each question across multiple age...

looking for someone to install FREEPBX 15 on Centos7

I have several FreePBX / Grandstream PBXs, hosted and on-premise, I am looking for a monitoring system that can help me with the following: 1. Automate test calls and verify two-way audio 2. Detect packet loss, jitter, or latency issues 3. Monitor SIP and RTP packets 4. Monitor SIP registration for specified extensions / trunks 5. Alerting capabilities 6. Does not overuse pbx cpu/memory

Hi Alexander, y noticed you finish a project named: Free Pbx IVR survey module. I am interested in the same project with some changes. I would need to see screen shots of the project in order to discuss with the responsable of making the software work. It is my understanding that the project uses phone keys to respond the survey, I would also need voice recognition availability to respond the survey. Please let me know if you are interested. Sincerely A. Ayala

...like to see a SIP VOIP call placed through our Asterisk server between the Entry Phone and the app. Establish a webRTC to webRTC connection having one-way video and two-way audio streaming. This call will be initiated from the entry phone to the resident’s mobile app where residents can see the visitor and be able to talk to them. Establish a webRTC to webRTC connection having two-way audio VoIP call for offsite APP communications. Establish a webRTC to SIP(PSTN) connection having two-way audio call and local VoIP connections (Guard Phone). If an alternate viable method is proposed we can consider it. The POC must include a means of adjusting resolution, i.e. VGA, and framerate, i.e. 10 frames/sec. The company accepting this project will have knowledge of c...

Hello i have asterisk with did where 500 users call this did number we hava to create a routing logic where we will interconnect them randomly inside dynamic conferance without pin if in the conferance caller or callee press 0 we will interconnect them randomly to any other conferance without pin in case the user have no any user available we will interconnect hem to any active conferance else if there is no other user we will play MOH until the any conferance will press 0 we will set a timeout for the conferance 120 secs this is small job but i will have many others

Would like to connect Zoho CRM with local Israeli accounting software, local Payment GW as well as loal PBX via API.

Would like to connect Zoho CRM with Local israeli accounting software via API as well as Payment Gateway and local PBX provider.

Hi, I'm in need of either a Discord bot, wordpress plugin, or a webpage I can integrate into my current site that users can access and use to select and strike stages for compe...matchup. Lastly, it is very desirable if the system is graphic based. I can provide the images for each stages available, but would be required for mod to update stages available. For now, placeholder text can be used: *Stage 1, *Stage 2, *Stage 3, *Stage 4, *Stage 5, *CStage 1, *CStage 2, *CStage 3, Player 1, Player 2, Room Code, Winner Status (asterisk items will be graphical) Would finally need to know from the individual chosen for this project, which method would be best to produce this in, Discord Bot, wordpress plugin, or web page Willing to work closely to get this done. Looking for q...

asterisk web interface needed for click and call

Hello, we are a new little italian isp provider company. we are looking to build our own CLOUD PBX SYSTEM with MULTI-TENANT. i hope someone can help us! Regards

we need a developer that should have these experiences: Modules. GSM Modules .GSM Channels, GSM Gateways, rSAP -> Remote SIM Access Profile, rSAP over TCP,,SIM Banks ,SIM Servers the developer will develop the software will run GSM Gateway and SIM Bank using Asterisk and GSM Modules, Hardware and software knowledge required. payments terms . please read it , will will not release any milestone until job is done , not negotiable

we need a developer that should have these experiences: Modules. GSM Modules .GSM Channels, GSM Gateways, rSAP -> Remote SIM Access Profile, rSAP over TCP,,SIM Banks ,SIM Servers the developer will develop the software will run GSM Gateway and SIM Bank using Asterisk and GSM Modules, Hardware and software knowledge required. payments terms . please read it , will will not release any milestone until job is done , not negotiable

We are looking to support SMS on our MiRTA platform (Asterisk). We also want our clients to be able to utilize SMS using the Zoiper app (open to other options as well). MiRTA is the GUI built on top of Asterisk. We will want the SMS options and features built into the MiRTA GUI for easy deployment and provisioning (we have access to the MiRTA developer). Please provide a time estimate. The budget is suggested and can be adjusted depending on your expert opinion of time. Please provide feedback. Thank you!

For custom made Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience needed: 1) LibreOffice Calc spreadsheets 2) Software to change Calc templates to pdf without loosing calculations (Foxit Reader ?) 3) Linux 4) PDF 5) PHP We use background images to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image, the spreadsheet is invisible ! This project is not for beginners !! You must have a PC with a Linux OS (pref. Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. This project will be divided into unequal milestones. See attac...

I need guidance on how to setup a blank sim card to connect to asterisk server and allow user to set caller ID

We are looking for VOIP technology expert to prepare streaming Asterisk and Dialogflow integration. We intent to implement a voicebot that realizes outbound calls basing on PBX (Asterisk), Dialogflow, Google ASR and TTS. Streaming Asterisk and Dialogflow with ASR is the most important issue in this project.

Hacer una práctica en Linux Debian en su versión 10.1.0, Asterisk 11.25.1, softphone Zoiper o Blink en sus distribuciones Free y configurar la Seguridad en SIP dentro de un mismo escenario (Máquina Virtual), Wireshark, VirtualBox

I need to run the my voip server with Asterisk. I also want to run the x-lite softphone from the VPS as well. What I need: Set up Asterisk on my server. Configure my SIP account from my number provider to voip vps. Configure

We are looking for a skilled developer or team to develop a simple one click remote support for our team and customers with main features: - one ...scratch according to your preference as long as it includes the expected , I wish to mandate you to take care of my private server hosted at OVH in public cloud, I set up and installed the solution "GoAutodial" in v4, unfortunately I meet bugs with "Kamailio" on the telephony part in webrtc in the browser. I explain myself, everything works correctly, the SIP is well connected in asterisk and I can make automatic calls on the campaigns, unfortunately one call on two I do not hear sound for the interlocutor. I tried everything to debug it, I can't manage it. I don't think it's a big deal, I'll let y...

Hello, I want to migrate a CentOs Vps server from OVH to Digital Ocean. This little server have a automatic asterisk autocall system for security system. We want to keep all functions and add another ; - Automatic backup. - Firewall. - Check asterisk autocall system who have a problem. Thank.

Configuratiion Push Server to wake up our mobile SIP dialer to a pBX

press prefix 9 or 90 it should take sip line of yeastar S300 PBX and call for Outgoing

PHP based web application to manage asterisk queue and channels

I have a free PBX platform that I need to modify its press # options. It's on Asterisk VPS server. Dev will update current contents of the freePBX and point backends to url php scripts or 3rd party DID#s while preserving the current contents of the PBX. Individual developer/programmer prefered, no team or groups. (Removed by Freelancer.com admin)

I need help with Hubspot CRM PBX CTI Integration. I am looking for abilities below. Click2Dial Call Recording in CRM with each call Call logs with all details Incoming and outgoing Call pop up with details like Name , Account Names, Call Duration and call status

hi. i will need a crm system based on vtiger with integration to fusionpbx dialer. so when the lead is shown to agent using the vtiger crm the client phone number will be masked (the agent cant see the phone number) but when he press that masked number the crm will send a request to the pbx syatem to dial that number.

For custom made Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience needed: LibreOffice Calc and Linux We use background images to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image, the spreadsheet must be invisible ! The Templates are divided into 3 parts and 2 have spreadsheets with different cell addresses ! You must have a PC with a Linux OS (Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. *** See attachment.

...Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience needed: 1) LibreOffice Calc for spreadsheets 2) Software to change templates to pdf without loosing calculations (Foxit ?) 3) Asterisk PBX / FreePBX 4) Linux 5) MySQL DB 6) PDF 7) PHP 8) KVM qcow2 (for disk management) We use background images to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image, the spreadsheet is invisible ! This project is not for beginners !! You must have a PC with a Linux OS (Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. This pro...

I want to make my PBX outbound SIP calls to pick the next available SIM slot in cycling manner if the previous is busy, Setup SMS sending & receiving using SMS API, Call Credit Limit management, SMPP for SMS. Experienced freelancer with dinstar uc2000-vg can bid

i need someone to setup freebpx + asterisk on a cloud server, need asap.

We need you to install and setup an Asterisk PBX or another free PBX from scratch. You will also have to setup FOP on it, do that i can manage the PBX from there (you can refer to 3CX Management Console and Queue Manager Account functionalities to better understand which functionalities i need). - The PBX will have to record the CLI but normal extensions won’t be able to see it - Normal extensions won’t be able to dial external telephone numbers - Extensions are remote, so we will have to be able to configure those on Gigaset VOIP phones and SIP software phones as well - Extensions will usually be simple PBX extensions (i.e. 595) but they could occasionally be external numbers (in the last case scenario, we will need to keep...

Hello, I'm starting a voip services company, and I need to build an pbx server asterisk + web (back end and front end), with all the asterisk features, I await proposals.

Calling Without Internet Connection, We wanna build a technology stack for calling without internet. Using services like Jitsi which allows users to connect to conferences from regular phone lines, we are currently using services like matrix-org, riot im & jitsi. Services like asterisk-org and freepbx can used or similar open source technology can be suggested. Services like rebtel already provide this feature in their suite, we want similar solution.

We have an Askozia (Asterisk) PBX in a nursing home with only one incoming number. Currently caller hits the IVR "Enter the extension you require" and is connected with that room. We need to automate the process, such that if caller's CLI only ever calls the one target EXTENSION, then after 10 calls, IVR will ask for confirmation (from caller) to forward all future calls the same extension. Asterisk scripting can be inserted via the GUI.

I have a DX650 and i am trying to make it work on FreePBX asterisk, i need an EXPERT in Asterisk and in CISCO IP PHONES who Has already done this before with success.

We're a small business that has multiple development projects ongoing. We're an Atlanta based company and are looking for a local developer that we can work with. We're in need of a Full Stack Developer that has experience in: • LMNP (Linux, Mysql, Nginx, PHP) • Redis • Laravel • REST APIs • Vue.js • Node.js • Express • MongoDB • • Docker Pluses: • Asterisk • VoIP • SIP • Networking • Linux

Linphone for iOS and Android is up and running (able to to receive push, background calls and SMS) Our developers can provide some support for Mobile Applications and asterisk, they have limited knowledge related to push notification. - Flexisip must be installed and integrated and all modules must be in working - Flexisip must be able to communicate with Asterisk to send or receive SMS (custom work) - Flexisip must be able to send push to iOS, Android Devices Please do not bid if you don't have good understanding of the project - Flexisip Overview - Flexisip Wiki Wiki

Quiero instalar Issabel (asterisk) con la extensión FOP2, tambien el mantenimiento posterior

Hello, I want to be able to offer my subscribers voice, video, and fax. I would also like iOS push notifications for chat messages. I already have a webhook with Twilio for SMS. I have many projects that you can choose and i pay very well for good work. I’ve been told I should get pbx system but I don’t know enough about that software. So if you are available, we can discuss specific project. If you’re busy I understand.

we are in need of an expert in Linux Armbian that can compile and application in to the Armbian a good Knowledge of asterisk will be helpful