Asterisk sip gsm jobs
I am hiring electrical engineers in following areas: 1) STM32 2) ESP32 3) GPS/GSM 4) Bluetooth / Wireless Mention you areas of expertise in your bid. I would encourage new freelancers to apply.
Greetings, We are looking to get a other few design services from you. Will mention the details below; please let me know an invoice amount for it. 1. A 10 fold (opening out) 50mm*80mm brochure kind, which will detail in t...get a other few design services from you. Will mention the details below; please let me know an invoice amount for it. 1. A 10 fold (opening out) 50mm*80mm brochure kind, which will detail in the 8 oils (will provide the content for it). The first page will be the same photo of the product, and then in next 8; 8 oils with details and at last page details of company. It will be of 30GSM. 2. A 300 GSM 50mm*80mm card which will be like a commitment for goodness card. One side will be written- A COMMITMENT FOR GOODNESS and on second side some matter which we will...
I need someone who knows about asterisk PBX for packet related issues on Asterisk
Im looking for someone can build me a GSM data receiver skimmer credit card for the pos and atm.
1. The Controller will control Streetlight 2. Measurement of Energy consumption 3. Communication through GSM. 4. Data will be sent to server for display and control the controller. 5. Over the air update firmware update.
Hi Folks, We have a standalone asterisk server recently installed on a Ubuntu box. We want to achieve the following in terms of network connection 1. Ethernet interface - This connects to telecom provider for asterisk line and should be used for everything related to SIP 2. Wireless - We want to use this for connecting to the internet Need help from a networking/asterisk developer who can help us split the traffic in the above-mentioned fashion.
...Member's Financial Standing ( Total amount contributed shown next to member's name) 4. Linked to Payment Portal (including Major Five banks in RSA) for online instant payments 5. Automatic Financial Records of all incoming and outgoing transfers with dates and time 6. Ability to automatically send monthly statement of financial activities within the company to member's emails 7. Ability to send GSM text to members every time there is a transaction taking place (Money out only) 8. Ability to upload documents for review by members with limited moderator rights. 9. Have general comment or chat section for members only 10. Have a live chat option for customer care services for public This must be able to be transferred to a host. I would like to know the cost and ho...
Hello, I am looking for a developer who can create a script to sniff phone numbers (sip/pjsip over tls). In order to avoid reading the CDR regularly, I would like a webhook that calls a url. I need date, time, calling phone number, called phone number, transaction id, call duration, status: response codes The documentation for the installation of the script. Environment : Server : PBXware (tls / ssl, pjsip/sip) Smartphone : Communicator / Glocom Linux server (sniffer) : Debian Language: python Library: scapy or other. Transport: tcp / tls
Want to able to transfer a call to another agent/queue using a prefix so we can record a note that is then played to the new agent when they pick up. FreePBX 15 Asterisk 16.13 PHP 5.6.40
A local company designed the hardware and a year later had to re designed it again and I am still waiting for them to finish the firmware the board is a 4 layer the micro controller is a ATXMEGA32A4U-AU the GSM is a MC60 QUECTEL MODULE a 4 button transmitter /recover connects on the board The GPS also check if the vehicle is moving or stationary if the vehicle is stationary a relay on the board is activated that's connected to a op-amp and then to the unit a variable resistance value will then check for changes. all this information will be send to a remote PC. An ignition supply is also required without supplying the wire, when a truck is switched off its voltage will go to 25/25 volt when it start it will go to 28v if that can be used as a switch in the unit if not i would ...
Thanks for reading, we have a problem to solve as below: 1. We are running Asterisk IP PBX soiftware on MT7628 Cpu running OpenWRT 2. For audio we are using PulseAudio and we have some problems: 2a - The echo cancel algorithm for the on-board speakerphone gives echo to the distant end, i.e. the echo cancel is not fantastic and needs looking at 2b - We are struggling to play recorded audio over our speaker 2c - The PulseAudio always requires a second reboot to work properly. If you are an engineer with solid experience in this area we would appreciate assistance on solving these items.
I am researching community cellular networks and need an installation of OpenBTS gsm system that works with software defined radio. Should work in India. Skills Required Telecommunications Engineer, C Programming, Computer Security, Embedded Software, Engineering, Linux, MySQL, PHP, Software Architecture, Software Testing, Web Security, Python Script, MATLAB, ASTERIX Hardware, Raspberry Pi
...connected a "feature type" phone using its native dialer to our VoIP (SIP to SIP) Platform (rather than an app on a smartphone). The phone is Android OS and does have a SIP client. The phone has a messenger app for sms and the manufacturer has noted we could make adjustments to the apk as this is publicly available for developers. On our VoIP side the custom application should support the SIP SIMPLE protocol for messaging. It should send outbound SIP MESSAGE to our SIP node IP as well as receive inbound SIP MESSAGE from the SIP node. This maybe helpful … https://cs.android.com/android/platform/superproject/+/android-8.1.0_r48: and dive into the phone and messages source code. The SIP stack is in the ext...
Hello, I am running a Small call center with Freepbx 14, needs to develop a customer satisfaction survey application for incoming queue calls. Call landed to the queue > agent answer > agent completed the call > should go to satisfaction survey automatically or agent can transfer the caller to satisfaction survey. Custom IVR should be able to play to the caller, should have a dashboard to view the reports. Reports needs to be contain following Date and Time / Number / Agent's Extension / Score needs to be able to run reports monthly / weekly / daily and custom date range.
We would like someone to organize various events at our retail location on weekends.
Hi Josh, I'm a VoIP communications engineer and developer that can add a new SIP Trunk and update existing SMS API to a new API RESTFul endpoint as you want.I have 3 locations listed in my file. 2 are meant to load balance eachother. At the moment it works perfectly as the logs show the calls doing a round robin between the 2 locations. We use 2 sip providers. 1 gives us 2 trunks to load balance and the 2nd we use just for failover in case the first provider goes down or is unable to route a call for any reason. Here is my file: The config file I am using was created by someone else for a different system we use to use. # $Id: ,v 1.1 2004/08/10 16:51:36 dcm Exp $ # sample config file for dispatcher module debug=2 # debug level (cmd line:
Hi Josh V., Hi Josh,.... I am an expert VoIP developer. I will add a new SIP Trunk and update existing SMS API to a new API RESTFul endpoint.I have solution for your problem. public virtual HttpResponseMessage Get([FromUri]TwilioRequest request) { TwilioResponse tr = new TwilioResponse(); if ( == "Hello World") ("Hello back!"); else ("Text 'Hello World' for a friendly message."); return (, ); } SMS API enables you to send and receive text messages to and from users worldwide, using our REST APIs. Programmatically send and receive high volumes of SMS globally. Send SMS with low latency and high delivery rates. Receive SMS using local numbers. Scale your applications with familiar web technologies. Pay only for
Hi, can someone do the following: * Remove Logo top left of each page and replace with logo on the business card that blue loop with business name - Resource Appraisals * Change the top of the color of the bar to match...Replace fax line (800) 621-7070 with (866) 906-9326 * Edit the copyright information at the bottom of each page from 2010 to 2002 * Remove the line after All rights reserved with Resource Appraisals and the logo are trademarks of Resource Tech Appraisers Then make the document into a fillable form for Mac / PC, Internet. It will be sent to people to fill it all out and submit it to send back. Red Asterisk are required fields - All fields should support Alphanumeric characters. Under Additional equipment and Notes should support max characters for a long message...
Hell...a "feature type" phone using its native dialer to our VoIP (SIP to SIP) Platform (rather than an app on a smartphone). The phone is Android OS and does have a SIP client. The phone has a messenger app for sms and the manufacturer has noted we could make adjustments to the apk as this is publicly available for developers. On our VoIP side the custom application should support the SIP SIMPLE protocol for messaging. It should send outbound SIP MESSAGE to our SIP node IP as well as receive inbound SIP MESSAGE from the SIP node. This maybe helpful … https://cs.android.com/android/platform/superproject/+/android-8.1.0_r48: and dive into the phone and messages source code. The SIP stack is in the exter...
Hi, I have an issue with Call Deflection when a mobile calls into a DDI which is then forwarded back onto another mobile. This is getting rejected when Call Deflection is turned on. Works when off but I need it on to show the A party calling number. Need some to trace and fix my SIP Trunk parameters / settings. Using Freeswitch with the same SIP Trunk it works so it is supported by the carrier. I just don't know the 3CX system enough to debug and fix.
I am looking for someone to quickly put together a recording demo using either Asterisk or Freeswitch where the user is able to do the following: 1. Login into a basic site (will provide access to a DO server) 2. Have the option to start/stop a recording (simple button should be present for the user to click) 3. Once the recording is stopped it should be saved. 4. Ability to view and playback recordings with timestamps. We do not require any options to delete or edit recordings. 5. The user will login into the site on their PC and using a USB microphone will conduct the recording.
We are currently using a custom PBX solution. The PBX server is hosted in the Vultr Cloud and is based on Freeswitch with the frontend as FusionPBX. There are few issues which have come up which needs to be fixed urgently. The SIP trunk is being provided by Voxbeam 1) All incoming calls should ring the SIP softphones 2) Outgoing is not working properly for some regions. Need to diagnose that and resolve 3) If possible, need to configure an alternative SIP trunk in case Voxbeam is failing. The process should be automatic.
We are a telecom, marketing and analytics company, we’re looking for a PHP/Laravel full-stack web developer with experience in VoIP/SIP & Twilio or similar Telecom company and having knowledge of telecom and how telecom systems work especially FreeSWITCH & OpenSIPS. You must have the following skills to qualify for the job: PHP Laravel Javascript/JQuery SSH + Ubuntu Command line MySQL GIT - GitHub Twilio Telecom knowledge of VoIP, SMS /MMS, Call center style call queues, call conference rooms FreeSWITCH OpenSIPs AWS React These skills are optional: Facebook APIs We will offer long term work every week but only if you prove your skills in the first week or two. We are only looking for serious developers and to prove that please fill the attached document, u...
I need to create a B2BUA but for SIP to Microsoft Teams. I know direct routing will do most of what I need but Video is my goal. I need to be able to take SIP Video and send it to Teams Video phones. Early video would also be required. Willing to license whatever I need from Microsoft. PLEASE DONT BID IF YOU HAVE NEVER DONE THIS TYPE OF WORK BEFORE.
...integrated as required: Admin Area: Create, Suspend / Un-suspend, & terminate PortaOne account. Send Welcome Email with SIP account details. See and adjust client credit balance. Create products with SIP configuration options (such as PortaOne client call rate card, account type, DID monthly/annual charges, subscriptions, and credit limits) Create products for fixed subscriptions / Add credits Allow enable/disable of international dialing and call forwarding. Client Area: Show PortaOne Credit Balance In real-time Show SIP Account Details Topup dropdown with fixed amounts for easy use and fast checkout, top-up must create invoice and allow WHMCS payment gateways. Change SIP Secret / password Call Rates (call plan) Call History - XDR (Includes search, Do...
...as required: Admin Area: Create, Suspend / Un-suspend, & terminate PortaOne account. Send Welcome Email with SIP account details. See and adjust client credit balance. Create products with SIP configuration options (such as PortaOne client call rate card, account type, DID monthly/annual charges, subscriptions, and credit limits) Create products for fixed subscriptions / Add credits Allow enable/disable of international dialing and call forwarding. Client Area: Show PortaOne Credit Balance In real-time Show SIP Account Details Topup dropdown with fixed amounts for easy use and fast checkout, top-up must create invoice and allow WHMCS payment gateways. Change SIP Secret / password Call Rates (call plan) Call History - XDR (Includes searc...
I am looking for a script from where sip trunk gets connected and make calls through different numbers to one number. Its basically known as TDos. Any one who know about this work and can do it , PM
Hello…Good day, This project is to build a web application (based on DLMS Protocol by using the Gurux library) and a gas meter as an iot device must be connected to our web application by GSM Simcard through the Internet connection…(The programming language should be c# in the asp.net core3), and this web should receive data from the device and shows it to the user. This device also has an installer software(GUM Communicator) that the user uses it to install the device ... Our main goal is to build the mentioned web application. (There is also a sample that can be used to build a web application. You can get ideas from it). also the Job description is written in a pdf attached file completely again. The device does not connect directly to the database, the status of t...
Hello, I need to develop an AGI script to integrate a dynamic IVR with google's STT. Asterisk versions: 13, 16 and 18.
I need a design of PCB with STM 32 MCU DTMF audio response using 2g or 4g module
Hi, we are developing mobile application for SIP calling in React Native and SIP js and desktop application through Electron JS. We need help in following parts. 1. To setup proxy server to send notification when call is incoming for mobile app to wake up and register. But the call to be connected to main SIP server instead through the proxy server. 2. To implement the call conference in both mobile and desktop applications.
We got an issue with our PBX. Our SIP trunk provider indicates that our router blocks some traffic from his IP. We need to fix this. The router works well for many months without any issues. hardware: Mikrotik V6.48 RB3011UiAS PBX grandstream UCM6204 , up to date I join a pcap file with the issue. See example at line 15912, 15917 and many other lines. The PBX work well during about 18 minutes after this the issue happen and we need to reboot the PBX.
I want a Softphone WEB with integration of Janus Gateway + Api Gateway. Request: Contacts - Add, Delete, Update, Click call, Click send Message, Alert new Mensage, Resume. Call - Make Call, Mute, Transfer, On Hold, Conference, Show Participants Call, Dial Pad - Use Gateway Janus Sip Legacy. Enable video call using Janus Video. Chat - Send Message, Received message, List menssage - Integration Backend Services - API Gateway The application need to use Ionic 3+ and Angular 6+.
This project is the installation and configuration of fusion PBX on Debian 10. You will be provided access to a VM. What must be delivered Full functional system integrated to 1 sip trunk, VoIP phone and softphones A fully secured system that is only accessible to admin and users registered in PBX Full documentation of how to in Freepbx eg: add new PBX user, configure softphone, setting to be used for softphone, VoIP phone config and settings Auto-attendant with MOH for inbound Leave a VM by pressing a number VM must be sent as an email to admin and PBX registered users. New users can only be added by the admin Outbound calls are restricted to users registered in the PBX. Call recording when a call is answered. You must have experience of doing so as well. The development must ...
i need to integrate 8 port dinstar with my own CRM
Hello, Me and my friend we have created a SIP video intercom, we would like to start the production and sale of devices, but to get customers we need free applications and a server that will allow easy configuration. We are currently using the Russian application, but it is difficult to configure due to the need for port forwarding in the router. Not every client has the option of port forwarding, so we want to implement a free server in our database + free applications for our clients. The Linhome system is the perfect solution for this task, we have been testing for some time and it works stably with our intercom. I hope our requirements are not too high: A.) APP. We need versions for iOS, Android and Huawei are needed, and in them: 1. Adding our logo 2. Change of the graphic des...
Hello, My freebpx is not connecting to asterisk and i need a expert asap to help me with it, will be via anydesk,
I am looking for an expert in ESP32 and GSM Module. We need someone to write C/C++ code to connect to AWS IoT core. I have written all details of my project in a file and have attached the file below. Thoroughly read the file and let me know if you can do this project.
We are looking for someone who can setup an asterisk server for us. It must be setup with correct security. Furthermore, you must show how we setup new agents and DID numbers with queues and opening hours. We do not need an interface; it can be directly in the conf files. A ”phone” must also be made, like this image, where it is possible to Answer calls Transfer calls – both hard transfer and soft transfer (speak first) Put the agent on DND Pause the call Make outgoing calls Furthermore. the phone must be built into 2 different system, it is not something you must do, but you must ensure that it is possible, either via an API or similar. It is naturally important that if an agent is assigned to a queue and a if a call comes into this queue, then it must ALWAY...
Hello, we already have asterisk and freepbx installed and we need someone to help us configuring extensions and test calling.
I already have a ready-made programme developed, I want a person to Installation of this programme into a fresh server, Few changes and also a long term maintenance contract.
Hey, I am Muhammad Javed from Pakistan, I have a lot of projects but don't have professionals to manage them. Basically I am developing my own team of engineers. I am finding experts in following areas: -->> Multisim / PSPICE -->> Arduino / PIC Microcontroller -->> ESP32 / STM32 -->> GSM / GPRS -->> Internet of things IOT I am looking for long term work relation, I would encourage new freelancers to apply.
I need someone to develop a custom Asterisk - FreePBX module/application, this must have; - I want to make a keypad recording and audio player on call, for example (I call someone and want to play an audio stored on database and if user inputs #123 on keypad I will get that output)
We have an esp32 connected to a SIM868E GSM / GPRS & GPS module through UART. We need someone to write C/C++ code to connect to AWS IoT core. We got it working on cloudmqtt and other public brokers, but the TLS encryption is difficult with aws. It would be best if you have a SIM868E breakout board to write the software and test. We use esp32 as the main MCU so it will help if you use that as well. We will provide tls certificates and aws iot core connection details. It can be written using esp-idf tool set or using Arduino IDE with any MCU. Please ask if you need more details.
Hello we are looking ONLY for experienced architect in wood Houses with SIP panels We are building tiny house with SIP panels We did a sample of tiny house already but now we need to get done some drawings by an arcchitect used to build houses with SIP panels The Tiny house is 7.2m long and 2.4m wide, and 4.2m high I will ask you to show me plans you already did prior giving you the job Once done I'll have same job to be done with a garden studio Thank you
Develop a hardware & software for custom GPS device . Device will use SIM808 of A9G GSM+GPRS+GPS module. Device will sent GPS data through TCP protocol to a remoter (IP address is known) through GPRS connection, Project will use 8051 series mcu like nuvoton N76E003AT , MS51FB9AE,MS51FC0AE silab busybee /laser bee 8051 mcu. MCU will read GPS data from SIM808/A9G module through AT command & transmit data to server through SIM808/A9G using AT commands , after establishing GPRS . GPS reading has to be taken after every 10/30/60 seconds interval sent to GPS server. Ready Made Hardware will be provided for & only software development has to be done. All source code to be written in C
I need to build a software which supports my clients and connects them with my employees, I am looking for people having strong skills in Python Node.js My SQL or MS SQL building a CRM based web application The software Includes API Ping post, Multiple user access, Voip/SIP call, Subtitles display and more I need a Person/Freelancer who resides in Chennai, Tamilnadu India.
