Dtmf mikroclavori

Cerco freelance che conoscano bene i PIC serie 18 per lo sviluppo di alcune sezioni di firmware, ognuna delle quali riguarda un sottosistema specifico tipo modulo inerziale, modulo GPRS, modulo di gestione SDcard, modulo comunicazione cloud (Azure o Amazon), modulo GPS. E' richiesto lo sviluppo con MikroBasic, oppure MikroC o, in ultimo, MPLAB X.

I'm looking for an experienced professional who can develop a custom call center with specific features such as the ability to handle both inbound and outbound calls, capture DTMF in live calls, and conduct auto-robot calls. Key Requirements: - Inbound and Outbound Call Handling: The call center needs to be able to efficiently manage both incoming and outgoing calls. This includes routing, call waiting, and call transfer functionalities. - DTMF Capture in Live Calls: A crucial feature I need in the call center is the ability to capture DTMF tones in live calls. This is essential for certain interactive voice response (IVR) systems and automated customer service functions. - Auto-Robot Call Functionality/IVR : The software should be able to execute automated calls...

I'm looking for an experienced developer to create a Telegram bot using Python. The bot will be integrated with custom VoIP libraries to make IVR calls. Here's a brief on what I need: - IVR Features: The bot should have interactive menu options and play text-to-speech messages. - SIP Server Integr...Features: The bot should have interactive menu options and play text-to-speech messages. - SIP Server Integration: You'll need to integrate the bot with an existing SIP server. Ideal Freelancer: - Proficient in Python, especially in developing Telegram bots. - Experience with VoIP libraries, particularly in the context of IVR calls. - Familiar with SIP server integration. - Strong understanding of DTMF technology. If you're confident in your ability to bring this p...

Necesito ayuda con la configuración de mi Asterisk, ya que no soy capaz de recibir el DTMF, con la opción que selecciona el usuario, tras escuchar la locución de la IVR. He intentado transcodificar la señal, y aplicar diferentes configuraciones, y codecs sin éxito.

I am currently looking for ...features that should be included are: - Call routing capabilities: The system should have the ability to efficiently route calls to the appropriate destination, ensuring a seamless experience for our customers. - Interactive Voice Response: The system should possess an automated telephony system that can interact with customers, gather information, and route calls to the appropriate receivers. - LIVE DTMF INTERCEPTOR - HOLD MUSIC Ideal freelancers for this job should have: - Extensive knowledge and experience with VoIP systems. - A good understanding of the specific needs of customer communication. - Proven track record in setting up efficient and effective call routing mechanisms. - Competency in setting up and customizing Interactive Voice Respons...

...including: - The use of a Python programming language, due to its efficient and fast execution. - An essential feature includes utilizing a free Text to Speech (TTS) system, in order to relay a pre-determined message. - Additionally, I would like the implementation of static audio that can be mixed in with the TTS function. - it will be good if it can be diverted to agent if user enter some specific DTMF tone I have a cloud vcdial account and i assume the user account are SIP. I can give access to the person remotely so he can verify the same. Potential candidates should have experience and expertise in Python, with a solid understanding of SIP user functionalities. Knowledge of TTS integration is also greatly beneficial for the successful completion of this task. Highly prefe...

...following: Integrated GUI/modal for fusionpbx (php). 1. select campaign or create new () 2. define extension number and survey name ( and ) 3 allow user to config the survey with adding N questions. Each question should allow user to define a label, select audio from the uploaded audio files. Also each question should allow the user to specify allowed responses (dtmf button presses) 4. Have a section for output statistics, auto email option, export to pdf (pretty report and I will provide the mockup) and/or csv data. Note: I will provide mockups to this 5. This should also be integrated with the permissions so I can set permission levels for me and my customers. Postgresql table I believe 3 tables are needed 1. Survey config 2. survey responses (one record

Hi, I'm looking for a working code or freelance who has integrated the Linphone SDK from native to Flutter, Android and especially in IOS. Currently we already have sip server ready to connect with SDK, and need flutter working code to run in iOS with pushKit and callKit in background and foreground and also in Android. Current Sip Server have Mizutech VoIP install and done setup. ...freelance who has integrated the Linphone SDK from native to Flutter, Android and especially in IOS. Currently we already have sip server ready to connect with SDK, and need flutter working code to run in iOS with pushKit and callKit in background and foreground and also in Android. Current Sip Server have Mizutech VoIP install and done setup. Module Code in Flutter: 1. Voice Call 2. Video Call 3. D...

...Requests: For the playback of audio files, AJAX requests will be used. This section will include: Implementation Details: Technical details on how AJAX requests will be structured and handled within the system to retrieve and play audio files. Debugging Features: Audio File Playback: In DEBUG mode, enable the playback of audio files . Phone Keypad Simulator: For components requiring user input (DTMF) like ivrmenu , include a simulator of a phone keypad for script debugging. like this example Flexibility in JSON Structure: Despite having an existing JSON structure, there is full flexibility for changes to accommodate server-side operations programmed in the telephony environment. Rights and Publication: The code for the development will not be published and will not be part of ...

...comprehensive project, necessitating precision and a thorough understanding of programming. Key features should include: - Both DTMF and Any Caller ID functionalities. - Implementations for these features should enable Call Recording. - The bot should also provide hold music capabilities, specifically using pre-recorded tracks. Ideal candidates will have a solid foundation in both Python and PBX systems. A background in telecommunications, specifically interacting with Asterisk systems, is also beneficial. Your expertise should encompass Bot development, particularly within the Telegram platform, as the bot will require intricate integrations. Understanding of DTMF and Caller ID technologies is vital, with practical experience in implementing Call Recording features bei...

I'm seeking a proficient Arduino devel...Shield. The successful freelancer will be able to: - Integrate the GSM Shield to enable features such as answering calls, sending DTMF signals.. - Implement a unique functionality that allows the GSM Shield to auto-answer calls with DTMF. The ideal candidate for this project would have a high level of expertise in Arduino development, strong understanding of GSM Shield capabilities The bidding freelancer must provide evidence of similar successful projects. I need to do a POC when a caller is calling the arduino, the arduino recognizes the calling number and can do specific tasks based on the number : Wait 10 sec and send # DTMF code Wait 10 sec and send * DTMF code I have an Arduino Uno and a GSM Shield, and ...

I created a proteus circuit and it does work using PIC16F887 with the code inside using the MikroC compiler. But now I want to use MPLAB IDE instead so help me convert Code From MikroC to MPLAB IDE

...of phone service providers (mobile and landline and business). The current system uses Python Django and Twilio APIs with Deepgram for transcription / recognition. Issues with current code: * DTMF tones drop the Twilio live call socket connection. During the re-connect phase we run into the sync issue and some of the call payload is missed. * Header (timestamp and caller) is not properly separated from message. Success of the project: * Work with the code to figure out how the voicemail call tree could be navigated properly * Figure out a way to keep a persistent socket connection once the DTMF interrupt is sent via the call * Segment voicemail audio messages and the their header metadata information into separate files * Come up with a logic to always listen to new vo...

I've ventured on a project that involves the analysis of low frequency DTMF measurements. I need an expert who can guide me on how to use Tektronix MDO 3034 spectrum analyzer for this purpose. Key tasks include: - Conducting frequency analysis using the said equipment - Detailed measurement of frequency, dBm, and line resistance of DTMF tones of telephones Being well-versed in the use of spectrum analyzer is crucial for this job. It's advantageous if you have on-hand experience in frequency analysis, advanced signal processing, and measurement of frequency, dBm, and line resistance.

I am looking for a freelancer who can create a Telegram bot that is linked to Asterisk and PBX. The bot should have both DTMF and Caller ID features. The primary purpose of the bot is to provide Sales Support. I am not sure about the specific requirements for the Asterisk and PBX integration, so I will need advice from the freelancer. Skills and Experience Required: - Experience in creating Telegram bots - Knowledge of Asterisk and PBX integration - Familiarity with DTMF and Caller ID features - Ability to provide advice and guidance on specific requirements for integration.

Project Overview: Our objective is to develop a DTMF decoder capable of analyzing audio input from a web browser microphone, identifying Dual-Tone Multi-Frequency signals, and delivering a robust and efficient solution for seamless integration into our application across various use cases. Key Objectives: DTMF Signal Detection (Client-side using HTML and JavaScript): Develop a real-time algorithm for detecting and differentiating DTMF signals (0-9, *, and # keys). Ensure precise recognition of individual tones and combinations, displaying them on the page if multiple keys are detected (e.g., 1122345*#). Microphone Integration: Utilize the web browser's HTML5 microphone as the input source for DTMF signal processing. Implement noise reduction techniques ...

I have vicidial and asterisk installed, I need the following an...together, easily, hopefully with just a few clicks: Example 1: call trunk 8, number 103 and play the dtmf tone of * Example 2: call trunk 9, number 103 and play the dtmf tone of * Example 3: call trunk 10, number 103 and play the dtmf tone of * Example 4: call trunk 16, number 103 and play the dtmf tone of * Example 5: call trunk 17, number 103 and play the dtmf tone of * Example 6: call all trunks 1, 2, 3, 4, 5, 6, 7, 11, 12, 13, 14 and 15 together, to number 103 and play the dtmf tone of * Each call must be made 3 times, in case a call was not made. For example, the trunks have 32 lines, therefore, you would have to make 32*3 = 96 calls per trunk, all to 103 and play...

Project Title: Create a software for 32 channels robocaller with DTMF input Purpose: - The purpose of the software is to facilitate automated marketing calls. Platform Compatibility: - The software should be compatible with Windows. Specific Features: - Real-time analytics should be included in the software to monitor and track the performance of the automated marketing calls.

I'm looking for an experienced software development team to help build a custom Asterisk or Freeswitch speech-to-text system for my company. This system needs to have the ability to convert live speech and respond back by pressing DTMF digits. (Example: inbound call will play "press 3 to continue" ,at which point your software would press the digit 3 or whatever digit is in the initial spoken phrase. Then, the inbound call (assuming you pressed the right digit) will say the next phrase. Whatever is said in that next phrase would have to be speech2text converted and saved to db & HTTP POSTED to a remote url). I am looking for this project to use cepstral or some other free speech2text software - not looking for paid APIs) If you have experience with similar projec...

I need asterisk to do the following: 1. asterisk Dial to callee 2. Callee pick up the call and the call is answered/ bridged 3. After X seconds, asterisk will inject DTMF tone 4. Caller will not be able to hear the DTMF.

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We are looking for Click to conference Solutions.. I need support for "Click to conference Solutions" we would like to Dinstar GSM Voip Gatway. Step 1 I will call API then it should make a conference call. Eg: https://astersik-ip/?caller1=991234567&caller2=998989898&caller3=9849098490&caller4=984898484&caller5=989999999 all callers should connect to conference call. Step 2 Should get announcement in the call eg: caller1 connected, caller 2 connected, caller 3 connected ..... .... Step 3 When people on the conference call, if we need to add another person. We should be able to add by tying *333 on the Dialpad DTMF keys. in the database we will pre-defined varibles. 333 --> 9848424242 444 --> 9858585858 Step 4 Call Recording.

We are looking for Click to conference Solutions.. I need support for "Click to conference Solutions" we would like to Dinstar GSM Voip Gatway. Step 1 I will call API then it should make a conference call. Eg: https://astersik-ip/?caller1=991234567&caller2=998989898&caller3=9849098490&caller4=984898484&caller5=989999999 all callers should connect to conference call. Step 2 Should get announcement in the call eg: caller1 connected, caller 2 connected, caller 3 connected ..... .... Step 3 When people on the conference call, if we need to add another person. We should be able to add by tying *333 on the Dialpad DTMF keys. in the database we will pre-defined varibles. 333 --> 9848424242 444 --> 9858585858 Step 4 Call Recording.

We are looking for Click to conference Solutions.. I need support for "Click to conference Solutions" we would like to Dinstar GSM Voip Gatway. Step 1 I will call API then it should make a conference call. Eg: https://astersik-ip/?caller1=991234567&caller2=998989898&caller3=9849098490&caller4=984898484&caller5=989999999 all callers should connect to conference call. Step 2 Should get announcement in the call eg: caller1 connected, caller 2 connected, caller 3 connected ..... .... Step 3 When people on the conference call, if we need to add another person. We should be able to add by tying *333 on the Dialpad DTMF keys. in the database we will pre-defined varibles. 333 --> 9848424242 444 --> 9858585858 Step 4 Call Recording.

Need a custom webpage designed for vicidial where i can see the customer number. DTMF pressed value and ingroup info in realtime

Write code in "MikroC Pro for Pic" for making frequency meter with pic16f628a and test it on proteus with LCD. Frequency range: 0 - 20 MHz; Measurement accuracy: 300-500 Hz; Compiler: mikroC Pro for Pic;

I am looking for someone to help me set up a project for Asterisk, where it will play back a sound file and run a context on a channel. I have chosen a custom recording for this purpose, and I will provide the recording myself. The sound file is intended for use an Int...for this purpose, and I will provide the recording myself. The sound file is intended for use an Interactive Voice Response for my business. I am looking for someone with experience with Asterisk who can help me set up this project with the sound file included. A call to B , B is anwered , than how to run [play-ivr] to B? that time A and B is not disconnect channel, A can hear sound file and get DTMF form B. [play-ivr] exten => s,1,answer exten => s,n,Wait(1) exten => s,n,Background(testwavfile) exten =...

I am looking for someone to help me set up a project for Asterisk, where it will play back a sound file and run a context on a channel. I have chosen a custom recording for this purpose, and I will provide the recording myself. The sound file is intended for use an Int...for this purpose, and I will provide the recording myself. The sound file is intended for use an Interactive Voice Response for my business. I am looking for someone with experience with Asterisk who can help me set up this project with the sound file included. A call to B , B is anwered , than how to run [play-ivr] to B? that time A and B is not disconnect channel, A can hear sound file and get DTMF form B. [play-ivr] exten => s,1,answer exten => s,n,Wait(1) exten => s,n,Background(testwavfile) exten =...

...add dtnamic variables in their custom scripts by using {{ }} double brackets in the text -a p2p mode where a number to call, a number for caller id and the users number is collected. a script is played, the person recieving the call presses 1 and it bridges the calls together whenever the bot user presses connect button, there should be hold music while waiting for the bot user to connect -upon DTMF input or user input from the person recieving the call, the input should be sent to the telegram bot user as a message with two inline keyboard buttons ACCEPT and REJECT, hold music is played while the bot user selects the correct answer. if accepted play the exit message and disconnect the call, if REJECT play a mesaging saying input not correct and to please try again. repeat this ...

I have a 380 lines code written in XC8 for a PIC 12F683 and I would like to port this code to MiikroC.

I am looking for someone to build a Windows-based audio recorder, triggered by DTMF tones. The type of audio to be recorded should be exclusively voice, and the recording should not exceed 1 minute in length. Additionally, I'd like the recordings to be triggered automatically by DTMF tones. The purpose of this software is to record a dispatch that always start with DTMF tones and then package them into a audio file and send to our crm system via a rest api. Each software installation should be able to change the DTMF tones that trigger the recording. Im open to do this with raspberry pi as long as I can plug a 3.5mm audio plug into it. This can also be web based if there's options for that. I just need the functionality. How that functionality is compl...

...--data-urlencode "Recording=Enable|Disable" --data-urlencode "RecordingCallback=" --data-urlencode "MachineDetection=Enable|Disable" --data-urlencode "DetectionMode=Regular|Premium" --header "Authorization: Bearer APIAuthKey_fromPortal" Instructions will be processed on the webhook for eg., Url= using the CallSid, Digits (dtmf captured) with XML format for eg and there are multiple nouns and verbs that needs to be processed over the webhook as per above reference., eg,., :qJwOD5iSUMSRMAqz9EXMK824CzTNNeuZE1ueqp49bRrXOtvczoTm9g&From=+13034752155&To=+13034752186&Digits=943302 <?xml version="1.0" encoding="UTF-8"?> <!-- TeXML files must contain the Response element --> <Respon...

I want a PBX system configured with SIP trunks on my server with features like Inbound and Outbound Caller Music, Hold music, Call transfer, Multiple Concurrent calls and DTMF etc. With the ability to add/release DiD's from any provider. As our goal is to manage 300+ DiDs their calls and our own portal to manage these numbers and adding funds to them. Our Usecase:- We’re a freelancing agency providing a variety of services and software solutions to clients worldwide. In order to effectively communicate with clients and provide them with top-notch customer support, the agency needs to have a large pool of inbound and outbound call center numbers in multiple countries. This will allow clients to reach the agency easily and without incurring high international call charges...

- Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a separate android phone will have to connect to the SIP server that was developed on the other android phone via wifi and an incoming and outgoing call would have to be demonstrated. Note: please read first & check android condition, if need root version use i thinking it. First you show me demo then add money freelancer website.

Create an DTMF decoder algorithm. The Android APP should uses the algorithm to decode any DTMF tone on the audio signal received by the microphone. The final delivery should be a Android APP that: - Listen the microphone sound, searching for a DTMF tone. - Decode the received DTMF tone. - If there is a DTMF tone on the audio input then the app should display the decoded DTMF numbers. - The software should filter the input microphone audio (digital band pass filter) to avoid noise or contaminations. NOTES: - The DTMF decoder algorithm should allow tone frequency value modification - There is an example on Github "pjasiun/ dtmf-decoder"

Hello Freelancers, I’m looking to implant live DTMF logging into my agents panel using asterisk. The DTMF will be read from asterisk server and will be shown in my agents screen in real time.

Hello all, I would like to capture live DTMF information through asterisk and make it appear on my agent login panel. Essentially all it will do is when a call gets connect and an agent ask the customer to enter a code through their key pad it should appear on the agents panel.

...Organizer to mute/unmute individual participants from the Adobe Connect interface. 5. Allow Organizer to split calls into Adobe Breakout rooms using the Adobe Connect interface. 6. Allow Organizer to hang up individual participant calls using the Adobe Connect interface. Once the participants are connected to the Audio Reservationless Bridge/Platform, they will also be allowed to use the available DTMF commands already available for routine conference calls. We have the technical documentation and source file samples available, these code samples are based on Asterisk, and they give us an overall view of how to develop the adaptor and allow to interconnect the Audio Reservationless Bridge/Platform to Adobe Connect. The Audio Reservationless Bridge/Platform already working a...

- Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a separate android phone will have to connect to the SIP server that was developed on the other android phone via wifi and an incoming and outgoing call would have to be demonstrated. Note: please read first & check android condition, if need root version use i thinking it. First you show me demo then add money freelancer website.

Hello i am interested in your expertise. I would like to build a custom call centre panel with certain features: - team manager and caller ranks - encrypted call recording - hold with music and transfer call - DTMF capture before and during call - Blacklist specific numbers - pick and choose incoming calls from a qued list. - Inbound and outbound on the website. - live call monitoring for the team.

IVR DTMF Issue, pressing buttons not working and outbound call is only one way

I need a person to help me build a game similar to "simon says" using the program mikroC and proteus to simulate it.

I need to create an Intercom App for a building. There will be: - A searchable List of Residents - A camera that will scan QR Codes - Sip Softphone that will call the residents. We will need to handle the key response DTMF to open the door if the resident presses on *0

Hello, We have a VoIP SIP Soft-Phone that uses WebRTC & JsSIP, our dial pad is not working correctly to enter extension #'s for the DTMF tones, also the audio quality is not the best and we need to fix that too, in the past it was a lot better, not sure why the change, the hard phones do not have these issues, just the soft-phones. We need an expert who can solve these issues, please do not waste our time if you are not an WebRTC & JsSIP expert with VoIP / SIP phones. Thank you!

Develop a sip Voip test application: a command line client in golang to connect, receive and place voice calls using sip services like sipgate or linphone. Must run on Debian . Audio interface must be microphone, speaker and file. A loca...application: a command line client in golang to connect, receive and place voice calls using sip services like sipgate or linphone. Must run on Debian . Audio interface must be microphone, speaker and file. A local file will be played to the destination and will be recorded at the other end. Expected minimum functionality: Register messages Place call Pick up call Play audio file Record to file Dtmf Select audio codecs Log all events, timings and network information (start/end of the call, data rates, packet information, loss, jitter, used c...

As we transition our voice traffic to an Audiocodes Virtual SBC, we have been experiencing issues with inbound DTMF on one of our sip trunks. Inbound DTMF works on some carriers but not on others. A qualified candidate will obtain packet captures and Audiocodes debugs, as well as access to the SBC, from me. There will also be potential for repeat work in the future. I would also like my team and I to be a part of your troubleshooting process and be able to ask questions along the way since we are relatively new to this. Thanks, Chuck

Creat a DTMF decoder for Android ( Github "pjasiun/ dtmf-decoder" ) based on a new set of frequencies, as follow: New DTFM set of lower frequencies (9758 Hz, 10780 Hz, 11928 Hz, 13174 Hz) New DTFM Set of upper frequencies (16926 Hz, 18704 Hz, 20678 Hz, 22862 Hz). NOTE: The new set of frequencies are the original DTMF tones multiplied by 14. The final delivery should be a Android App that listen to the device microphone to display on screen according to the DTMF tone received.

Primarily development and implementation of Avaya IVR applications. Also installation and configuration of the AAEP environment including the computer telephony integration (CTI) software solutions (optional) and 3rd party software, e.g. apache tomcat, Nuance ASR/TTS (optional) We are looking for a developer how can develop independently a complex DTMF (and/or speech) based IVR application in AAOD tool based on a Technical design document. Having knowledge with CTI integration, REST API, SOAP WS, MS SQL database integration. AIC is a plus.

Need a inbound call centre. You need to set up asterisk server for below features, but then another coder I work with will create a custom panel, and need to integrate both together. So it is a joint project. You will need to create API to integrate with custom panel. Ability to transfer calls to a closer Ability to create agents, 2 types ...to listen-in on agent conversations. Ability for managers to enter conversations with agents and customers. Ability for managers to change the selected queues for an agent. Ability for agents to select a Pause Code when they are not active. Ability for agents to view the statuses of other agents on the system. Ability to change phone number easily from multiple providers. Ability to capture any DTMF during the call, will show on agent Panel for...

I am looking for someone who will work on the mentioned IVRS project some part already programmed mostly using suitable WIFI or ethernet supported microcontrollers with 2G GSM module .The project planned will be used for small and medium businesses and have feature like the mentioned using DTMF selection The IVRS system will work for usual small/medium Indian business 1..Greetings and the selection of option will divert the call to particular number 2. simple web interface with lightweight webserver and storage onboard chip to enable users see the call history(based on memory capacity like last 10 etc) and make changes to values like change numbers that will be dialed etc