Dtmf detektor jobs
VoIP/Sip client Android APP first of all the app must be fully encrypted. this app must me done with the SIP protocol standards. Basic Features: Incoming Calls Outgoing Calls Mute Calls Hold Calls Transfer Cal...me done with the SIP protocol standards. Basic Features: Incoming Calls Outgoing Calls Mute Calls Hold Calls Transfer Calls Integrated Call History Contac List change CallerID before calling or fixed optional: Send random caller ID before call. Advanced Features: voice/speech changer integrated to the calling app Multiple SIP accounts setup SIP Account information hidden after initial setup DTMF (telephone tone) support using SIP INFO or RFC 4733 Provisioning via QR Code SIP TLS transportation SIP UDP transportation SIP TCP transportation Audio codecs: G711, G729,...
Hello, I would like to have a chatline (wich will be in spanish). Real-time voice chat. Something like this: 1) Caller calls into the chat line. 2) Plays welcome message / Disclaimer .Caller can skip to step (3) using prefered define key using DTMF 3) Select chat menu a) Chat room b) One to one Chat (a) 1) Plays Welcome [url removed, login to view] can skip to step (2) using prefer define key using DTMF 2) Caller can go to the next room pressing 9, and 7 to the previous one [10 rooms], caller can know how many people are in each room pressing #. 3) They can return to the main menu, anytime, pressing 0. (b) 1) Plays Welcome [url removed, login to view] can skip to step (2) Pressing 1 2) The caller 1 is connected to caller 2, if not available, he / she will listen to...
Looking for expert with experience in data transfer over sound to help me build algorithm to send data using DTMF frequencies on telco network
...call should be configured through an API that is supplied with the APP - Enable to route calls from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Proxy/Gateway - 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 a...
i want a python expert who is familiar with Pjsip . ASAP. further detail will be shared in chat box
...that interact with VOIP Using PJSIP. The script must call a number and use gTTS ( Google Text to speech ) to say predefined text. Also it must decode DTMF to get user input. Please if you feel you can do it message me. Full Description: Using Python Programming Language: * PJSIP. * gTTS ( Google Text to Speech). Register and Login to VOIP Server via SIP using PJSIP. Script initiate a call to a nunber using a call function. The call function take argument, FromNumber(international format ), ToNumber(international format). after the call is established. Using gTTS Play (This is automated call Press 1 for english or Press 2 for spanish). DTMF Listener will get the tone and determine which direction it will take. if user press 1. he go to English part which will say for exa...
Adapt the software project on to have it working with a new set of frequencies. It is, the new set should be the frequencies of the original DTMF tones, multiplied by 14. It is named as M-DTMF. There are the specifications: M-DTFM set of lower frequencies (9758 Hz, 10780 Hz, 11928 Hz, 13174 Hz) M-DTFM Set of upper frequencies (16926 Hz, 18704 Hz, 20678 Hz, 22862 Hz). Modified M-DTMF tones correspond to the frequencies of the original DTMF tones, multiplied by 14. The final delivery will be a very basic Android App using the M-DTFM code, as previous description. The app must run in the background and display the received numbers on screen. If the M-DTMF audio pickup by microphone matches the pre-recorded number stored in a table at program memory
...the Github "pjasiun" project to have it working with a new set of frequencies. It is, the new set should be the frequencies of the original DTMF tones, multiplied by 14. It is named as M-DTMF. There are the specifications: M-DTFM set of lower frequencies (9758 Hz, 10780 Hz, 11928 Hz, 13174 Hz) M-DTFM Set of upper frequencies (16926 Hz, 18704 Hz, 20678 Hz, 22862 Hz). Modified M-DTMF tones correspond to the frequencies of the original DTMF tones, multiplied by 14. The final delivery will be a very basic Android App using the M-DTFM code, as previous description. The app must run in the background and display the received numbers on screen. If the M-DTMF audio pickup by microphone matches the pre-recorded number stored in a table at program m...
I need predictive dialer and reporting software for our issabel pbx server. Asterisk verison 16.16.1 1- Predictive dialer and reports. 2- After the automatic dialer, the desired announcement will be played, dtmf dialing will be recorded and reported. 3- Search will be made according to queue availability. 4- The number of calls to be made at the same time and the number of redial calls can be entered in the subscriber or queue information to be transferred if the number of calls is answered. 5- Campaign can be created. 6- The dialer will automatically call a list of customers and when they pick up, it will route the call to an available agent. 7- It will be web-based and needs to be written with nodejs react.
I'm looking for an Asterisk and IVR specialist with voice recognition and "text to speech", who can develop a dynamic IVR with an interface capable of editing prompts, navigation menus and reports. The user will be able to navigate through the IVR via voice recognition or DTMF.
I need an expert to help me to develop the auto play system for me, the process as follows 1) the function will get a phone number to call 2) when customer answer the call the wav file will play 3) the wav file will ask the user to enter code on the keyboard of his/her phone 4) if the user didn't enter anything for 4 second...play system for me, the process as follows 1) the function will get a phone number to call 2) when customer answer the call the wav file will play 3) the wav file will ask the user to enter code on the keyboard of his/her phone 4) if the user didn't enter anything for 4 seconds the wav file will repeat 5) if user enters the code 4-6 digits it will print it on my screen to save it you can use DTMF for this this is the process Please just...
I need a call tree flowchart with call forwarding, major voice recognition and minimal dtmf. Would prefer process to be visible and changeable in the flowchart itself. Caller calls in and says name of individual or department and is transferred to external phone number
...true if call is answered (files) → Boolean Task: • Play a list of mp3 files on the SIP call [given as Python list] • Return true if all files are played (char) → Boolean Task: • Send a DTMF sound • Return true if DTMF sound is played () → void Task: • Ends the current call () → Boolean Task: • Stops the playback of all mp3 files immediately • Return true if all files are stopped () → String Task: • Returns the status of the current call --> ringing, connected, playing, dtmf, killing, stopped, aborted, … () → Boolean Task: • Unregisters from SIP Server and closes the connection to the server • Return true if successfully done General Information • Must be written in ...
Small multi platform script that: - Registers with a SIP server - Initiates a call - Plays MP3 files - Does DTMF - Ends Call Must work on Windows and Linux "out of the box". All components and source files must be included.
1. Rename App 2. Adding the logo 3. Adding another 3 DTMF buttons (now the Linhome app allows you to add up to 3 buttons, we need up to 6) 4. Add our icons to the buttons 5. Change of the graphic design (colors) 6. Checking whether we do not violate the license rules 7. Test 8. Help in publishing in GooglePlay and in APPGallery 9. Share the sources on our account
I need a design of PCB with STM 32 MCU DTMF audio response using 2g or 4g module
...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 design 3. Linhome APP can add 3 DTMF opening buttons, we need 6 buttons in total. 4. Adaptation of the application only to our server .. eg: the ability to create a SIP account after scanning the QR code or entering the pin from the device sticker. 5. Applications must be adapted to phones and tablets. 6. A good solution will be to match the application with the horizontal and vertical tolerance of the screen. (...
Hi Guys! we need Softphone client for windows and Mac for our MT freeswitch platform (Hudusoft) the client need to include :...VP8 WideBand and Ultra WideBand codec (speex, opus) Audio tuning wizard Dynamic Jitter Buffer Packet loss concealment (PLC) Automatic Gain Control (AGC) Acoustic Echo Cancellation (AEC) Voice activity detection (VAD) Noise supression Push notifications Auto QoS Dynamic Threshold Algorithm for Silence Detection Network handling: UPNP, STUN, ICE, IP Translation, Firewall and NAT detection DTMF (Inband DTMF or SIP INFO messages) CRM solution: Click to Talk Personal address book Import contactlist from various sources (LDAP,WAB,Outlook,CSV,Active Directory, etc) Settings and contactlist backup and restore Please contact us if you have any questions Than...
I need predictive dialer and reporting software for our issabel pbx server. Asterisk verison 16.16.1 1- Predictive dialer and reports. 2- After the automatic dialer, the desired announcement will be played, dtmf dialing will be recorded and reported. 3- Search will be made according to queue availability. 4- The number of calls to be made at the same time and the number of redial calls can be entered in the subscriber or queue information to be transferred if the number of calls is answered. 5- Campaign can be created. 6- The dialer will automatically call a list of customers and when they pick up, it will route the call to an available agent. 7- It will be web-based and needs to be written with nodejs.
I need predictive dialer and reporting software for our issabel pbx server. Asterisk verison 16.16.1 1- Predictive dialer and reports. 2- After the automatic dialer, the desired announcement will be played, dtmf dialing will be recorded and reported. 3- Search will be made according to queue availability. 4- The number of calls to be made at the same time and the number of redial calls can be entered in the subscriber or queue information to be transferred if the number of calls is answered. 5- Campaign can be created. 6- The dialer will automatically call a list of customers and when they pick up, it will route the call to an available agent. 7- It will be web-based and needs to be written with nodejs.
Our project based on swift language and already implemented linphone SDK version 5 in our project and want to extent a features. The features is Sending DTMF while ongoing a call.
DTMF LOGO – BEAUTIFICATION Dreams To Miracles Foundation (DTMF) is a registered non-profit Charity, and the auspicing body for many projects to assist those in necessitous circumstances. SPECIFICATIONS: The DTMF logo (attached) is required to be beautified. The DTMF Committee likes the existing design (attached), but feels it could stand out more. The design MUST be of high-quality resolution (for print/copy use). Brief/Preferences: 1. Please maintain the base design, keeping it similar to the current design (attached), in regard to shape (the design MUST be circular in shape) and style. 2. Make the circles (inner and outer purple circles) thicker/bolder. 3. Make the ‘DREAMS TO MIRACLES’ wording thicker/bolder. 4. Change the colour of the &...
Geomant is a leading provider of unified communications and contact center software, well known in the sector for systems integration and providing effortless customer interaction solutions. With 2...vendors (ISV) and cloud service provider partners shows Geomant’s strengths in working with others. Geomant is a long-term Avaya DevConnect member, an Avaya Business Partner and a Microsoft Gold Communications and Application Development partner. Internally, these strengths are reflected in a people-centric attitude. The Task: We are looking for a developer how can develop independent a complex DTMF based IVR application in AAOD tool based on a Technical design document, who has knowledge of CTI integration, REST API, SOAP WS integration, MS SQL database integration and AI...
I want to built a sm...small proof of concept for a potential client. It should allow an external phonesystem (GENESYS) to forward a call to a number (TWILIO), and then the user gets prompted to answer 2 questions with dtmf tones or voice, and 2 where they can just leave comments (voice). The results need to be recorded (also twilio feature). Practically, a TwiML Bin consists of XML depicting if-else decision trees in a language called TwiML. You can ‘Say’ a batch of words, ‘Gather’ user inputs (either voice or DTMF), or even ‘Play’ audio files – among many other possibilities. In our IVR, we will set up a TwiML Bin that will take advantage of that functionality.
I need a solution which can receive a call and let the caller enter a "case number" and based on this case number it should route to the appropiate department. This means a lookup in a database to search for that case number and lookup the corresponding department, when found, route the call to that department. The receiver should get a popup of some sort to see what case number was entered. Can you create this? Kind regards, Remco
I'm searching for a device that would accept a sim card and operate on the 4g GSM network. This device would provide a "pots" or "tip and ring" output. It would need to accept incoming DTMF tones from the receiving part and from the connected dialing device. It will also need to provide line voltage or simulation. Ideally it would have an antenna port for an external antenna.
Looking simple app in React native. Where i can receive video call from Asterisk sip and send back DTMF code. Only incoming video call required # DTMF code can be sent by app support android and ios.
HI, Looking simple app in React native. Where i can recive video call from Asterick sip and send back DTMF code. Only incoming video call required # DTMF code can be sent by app support android and ios.
HI, Looking simple app in React native. Where i can recive video call from Asterick sip and send back DTMF code. Only incoming video call required # DTMF code can be sent by app support android and ios.
...tables throught a periodic sync operation. Then InTime reminds customers of their scheduled appointments via email, SMS and automated phone calls. The forewarnings is sent a configurable number of days in advance and also the prioritization of the channels (Email, SMS and phone calls) is highly customizable. The contact have the option to confirm appointments answering to the communication, via DTMF recognition, contact can reply to SMSs or Emails inserting specific keywords or use an embedded URL. Alternatively they can cancel the appointment or talk to an operator to reschedule it. The communication sequence is logged and the result is saved and, optionally, written into the external Agenda. InTime offers a UI to login, config its parameters, verify the communication acti...
Hi, I'd like to do the following on FusionPBX. Think would need to utilize the IVR then do a couple customizations? (1) Receive inbound call (2) Play recorded questions (3) Capture DTMF responses to the questions (4) Export out the results to a .csv (or store in a DB that can be accessed)
Ability to see DTMF input live maybe through a web GUI. (Without using IVR) so that we can verify that they've received our verification code. Outbound IVR system When a call is answered IVR sound file is heard and the customer is asked to press 1 to speak to an agent or 2 to end the call and allow them to give us a call back when they have time, if we call multiple people and they all press 1..they will need to be put into a queue if all of our agents are busy. Must be able to easily load new audio files to use in the IVR. Needs to be compatible with FreePBX/Asterisk, ideally an easy to add module but I am flexible. Scenario After customer presses 1 they are connected to the agents extension, when agent requests the customer to input our verification code, it'll sho...
For an outbound...API will provide information to FreeSwitch like name and surname of the customer and also amount of his debt, etc. This information will be used later in the IVR. 5. After those REST APIs FreeSwitch should play an announcement and the number returned from the REST API above, for example 150, 250 78 (without TTS) etc. 6. After the announcement, a response (DTMF) will be waited/received from the customer. 7. The result of the received DTMF will be returned to OCM again via a separate REST Web service. 8. The call will be closed and the information that it has been closed will return to the same OCM product. A preliminary meeting will be held with those who want to work with us on this project. Then the details of the work will be discussed and it will be star...
Make calls through telegram which will go through a SIP provider and asterisk api. You will have to use IVR and DTMF on the project as well. The necessary skills, langages and packages will be. Sip trunks configuration Python Asterisk config DTMF FREE PBX
Requirements: - call reject/hang up, - call transfer, - no activity detection/silence detection, - DTMF detection, - Linux platform kernel version from 2.6.
We are looking for support for Amazon connect IVR and call flow implementation. The job includes connectivity for 15 API with two separate systems, creating a simple lookup DB for a location to user lookup. The IVR would need to validate the customer Input (NLU and DTMF) that was provided and call API using Lambda functions and validate the information store this till the end of the call to process subsequent information's requested from the backend systems. The Dynabo DB table would also need to be used for this. The call flow is already defined and is pretty straight forward, We also need support with the reporting and call flow management. The systesm should also be integrated to a work flow manager.
...gateway: 1. Connect via SIP (or WebRTC) to my voice switch, which is hosted on the internet and not local to the phone 2. Accept calls from the voice switch server and place call out over the GSM connection. 3. Provide two way audio for the caller/callee Call generator: 1. Accept commands to place call on GSM side and make call 2. No need to connect audio to voice server 3. Play recording (or make DTMF noises) from phone over GSM call 4. Hang up call when required SMS gateway: 1. Connect via smpp to my SMS server (or use HTTP polling to a HTTP endpoint) 2. Accept SMS from my SMS server via SMPP or HTTP polling and send it out over the GSM network 3. Accept SMS from the GSM side and relay to my SMS server via SMPP or HTTP 4. Provide DLR if network is able to provide this Contro...
We are looking for a custom-built SIP Softphone (PJSip Asterisk extensions) project using .Net 5 and C# to with the following features: 1) Source Code in Visual Studio project using .Net 5 for Windows Application 2) Downloadable internal directory/contacts 3) Simple UI 4) Minimizable to Tray 5) DTMF and Multi-line with hold, conference, transfer 6) Configurable ring notifications 7) Cannot use proprietary/paid license components 8) Completely configurable from web endpoint (can be XML/YAML/text for the extension, server, etc) 9) Code standardly documented in English
The project includes CIPHERING A VOCAL SIGNAL, ANALYSIS OF A FILTER and GENERATING AND RECOGNIZING DTMF TONES
this project includes: CIPHERING A VOCAL SIGNAL ANALYSIS OF A FILTER GENERATING AND RECOGNIZING DTMF TONES
1. Set up IVR Flow based on the requirement. 2. Integrate the IVR with two external systems for validation and retrieve data based on the inputs provided • Flow will need to recognize caller’s input using DTMF and LEX Voice bot • Basic Flow o Validate the caller using TAX ID based on information available SYSTEM 1 o Validate ID (SSN) using SYSTEM 2 o Validate DOB based on the information available in SYSTEM 1 o Once the information is verified, look at a specific table and read out the information using TTS o Must be able to send information through fax to caller when needed
...trigger from Agent side to force the telephone call to become a conference call. Audio between the agent and the trunk will be muted for DTMF tones. A third SIP call leg will be added to the existing call connecting the audio path between the customer and third call leg parties. DTMF and audio will be active between customer and third call leg. Third call leg will be sent a XFER header in the SIP information. XFER header will be pushed with the API trigger to start the conference process. XFER will be used to allow the Agent data session to be cross referenced to the data collected over DTMF on the third call leg. c) When third call leg disconnects call DTMF/audio will be reinstated on call leg A and B custoemr and Agent. In summary we want to control the c...
Sip Softphone to be installed in low cost nokia type bar phones to be configured as a feature phone Required features: 1. Based on SIP. 2. Should support Voice and SMS. 3. CodecG.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. The whole system should work with standard softswitch. 7. Registration of softphone clients through the encryption server in the softswitch.
...EXISTING LOGO/SIGN – ASTALA ALLIANCE "Astala Alliance" is a project name for a community shop (a co-operative which will sell local and handcrafted products), and will be auspiced under a registered non-profit Charity called Dreams To Miracles Foundation (DTMF). SPECIFICATIONS: The work required for this brief is to BEAUTIFY the existing logo/sign for “Astala Alliance”. The maypole (pole with ribbons) theme must be maintained as close to the original design as possible (a new/redesigned logo is not required). The Dreams To Miracles Foundation (DTMF) Committee would like this design to be more beautiful and elegant. The design must be colourful, and all colours of the rainbow must be included in this design (red, orange, yellow, green, light...
I want a combination of dtmf receiver, dtmf decoder and dtmf motor driver circuit combined in a single circuit board to control a drone
- Must be compatible with PBXAct and FreePBX. - Can run in background a...- Can run in background and stay registered even if multiple programs are open after. - Be able to Forward a call to an internal extension or to a phone number. - Be able to do a Conference call. - Have a Contacts list. - Compatible with UDP/TCP/TLS for Transmission Protocol - Can register multiple account at the same time - Support multi-language (French and English) - Support RFC2833 as DTMF - Support PCMU (ulaw) as Voice Codec - Call History And a (nice to have) : - Use a QR code to add an account easily - Multiple ringtones - Support QoS - Import contact list from iPhone, Androïd or Outlook. - DND option If you can give me a time frame to build the basic and another tim...
Create a simple VOIP Application using React Native 1, Must use latest React native version 2, Should able to work with existing SIP Server (FreePBX and VitalPBX) 3, Should able to make an outgoing call 4, Event Trigger when DTMF is received.
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone wi...from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - 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 ...
We're developing an app which has an unepected need for Asterisk integration. We need 1 extra developer to help setup asterisk with SIP trunk to take data from a python program, and make a phone call with that data. The call will need SpeechRecognition implmeneted, and play DTMF tones.
