VoIP Jobs

I am looking to set up a Twilio Ringless Voicemail System with VoxDesk for 1-3 users. This system should be able to drop voicemails without ringing the line.. I realize you've probably never used VoxDesk so we're looking for someone who can test it out. It is a no-code platform but you need to be familiar with Twilio. We will also want our service to forward to a different telephone system. We have a $150 budget for this. Thanks

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.

voice broadcasting, automations call , IVR set up software with help of SIM card IVR device

we need an expert in flutter to develop flutter app that will combine 2 plugins #1 flutter gsm dialer # =sip client the app will rout the flutter gsm dialer mic audio to the flutter sip client speaker and rout the flutter gsm dialer speaker audio to the flutter sip client mic To route the microphone audio in a Flutter GSM dialer with a Flutter SIP client speaker, you can use the Flutter AudioRecorder and AudioPlayer plugins to capture and play audio, respectively. You will also need to use a SIP client library to establish a SIP session and communicate with the SIP server. Here are the general steps you can follow: Add the flutter_sound and flutter_sip plugins to your Flutter project. The flutter_sound plugin provides audio recording and playba...

I need an expert in installing FreePBX system on our server. I have specific requirements and I want these requirements to be fulfilled before setup. If you think you are the right fit, please let me know – I am looking for someone to start this project as soon as possible. Thank you for your time!

Hi, I have goautodial 4 setup on server with ssl/tls everything is working great except for the carrier dial plan that does not make calls out. All i need is assistance with the dialplan to make sure that the calls can be made from the webrtc phone built into goautodial 4. The software is setup already the solutions looks like it works fine. I assume the problem is with the dial plan. I need someone that can go through the system to make sure calls can be made out using the web phone built in.

We are looking to build a new email access platform that allow our customers to get access to their emails over a phone call and they will be able to fetch new emails, read emails over a phone call by using the system text to speech and they can reply, forward, delete or mark spam by using speech to text and also writing their emails reply by using dictation via a phone call and then the dictation will be converted to text by using speech to text. Our system will be based on an email server which will provide the pop3 email access to the phone system in order to fetch the new emails and allow the customer to control their emails by reply, forward, delete or mark as spam by selecting options on the phone keypad or by voice commands. Each customer will have an email account on the email se...

the project have 2 api First api --- route audio from whatsapp mic to sip client speaker and from whatsapp speaker to sip client mic second api this api will reside in android and will be connected a to remote server that will be able to instruct the api in the android to initiate call in whatsapp and to monitor the call stat like ring dial . call connected and end call and to be able to terminate the call both api will work on android devices with android version 7 and up

we are looking for a exceptional good expert about SIP protocol and SIP states. Your task will be to help us to identify things on SIP to be able to discuss with our developers (with low expertise on SIP). The developers have to implement some features, but do not understand the SIP protocoll well to find the correct paths. Your task will be to help in the discussions about low level SIP featuers like: - how to list all registered SIP devices (softphones, smartphones-apps, desktop-apps, physical phones, ...) - how to identify how many onging calls are running in parallel? - how long is each call onging? - who (device) has taken the call, what time ended the call, ... - and many more Your task will be to consult only, except you are a developer too. If this is the case, you are welcome t...

Hello, We have server on which is installed FreePBX. We want to buy Yeastar TG100 GSM and Cisco SPA502G. We need to configure this 2 devices so when call is made to GSM SIM cart that is in Yeastar the call to be trough VOIP to Cisco

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.

An Android tablet will be placed at the door. The tablet will log in to a local SIP server, set in the settings page. During idle state, it will show a static image or video of the user's choice, set in the setting's page. When the user taps on the screen, he will be presented with 2 choices, "Guest" or "Delivery" Tapping on 1 of the choice will initiate a SIP video call to the preset extension number. The extension number will be set in the settings page. Video of the conversation will be recorded. Video will only be from tablet to server, there will not be any video from server to the tablet. There will only be audio from server to tablet. After the call ends, it will return to idle screen. 1 setting field to define where the videos will be saved, opt...

Hello i need someone to install and configure vicidialer or any other webhosted dialer webrtc need to know the cheap hosting/vps/server so i can buy and let him setup ... Regards

I need some DID experts with a long experience who can set up the whole process which is needed to create DID numbers together with the Telecom company. Please do not respond if you do not have that experience.

Looking for someone who can configure auto dialer for freepbx We already have it working but now it dials the customer and when that is picked up, it dials the agents. It should start dialing the agent the moment it starts ringing on the other end

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.

We are setting up a FusionPBX with a peering trunk. But we are having registration failed with error 403. Would be great if you can help us with clearing the error to register the sip peering as well as setting up an IVR. Please do not bid if you are not a professional of VoIP.

I'm looking for a developer to create an AI based VOIP system. The main focus of this project is to implement natural language processing as a feature of the VOIP system. I'm looking for a full feature set, with all the bells and whistles that come with the AI feature. Additionally, there will be some existing systems that need to be integrated into the new VOIP system, we want to offer this service as an API, so experience in integrating existing systems is highly desired. The idea behind is people can automate their inbound call centers and our AI bot will take all calls Example system is