J2me dtmftrabajos

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 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

3 Stage IVR, needs to be a customizable outbound call along with a live DTMF output feedback & too re-routing to the calling agent (at the other end of the line) if the customer-client demands so. Further details will be given within chat, including diagrams & flowcharts.

Write incoming dtmf to a file on Asterisk. They need to know Asterisk programming and Linux shell commands. We are needing to capture and write DTMF to a file on asterisk. Aim : To Read a variable in the form for DTMF tones as pressed by the caller. For example if you would like your users to call up the system and record there inputs in the database and then make use of Asterisk to perform what ever tasks with those recorded inputs.

I am in need of an android app that can record the DTMF code entered by the caller. 1. The android app will answer the incoming call if the incoming mobile number is registered as a member 2. The android app will play a pre-recording greeting - "Welcome, please enter your pin number." 3. The android app will then detect and validate the dtmf tone (pin number) and 4. Despatch a fail or success Server / SMS message.

Hello All PLEASE READ CAREFULLY! need Android application which can do the following: 1. send API request to get 4 numbers ( from a server ) 2. the app must dial the first number ( using android phone keypad ) 3. after call connected and x seconds the app must dial the second number ( with DTMF ) 4. after x seconds the app must dial the third number 5. the call must be recorded ( and saved with file trans_no and name of the second number ) 6. after this the app must hang the call up and save the audio file on local MMC, and send it back to sever ( api request ) 7. save log locally on phone MMC 8. restart from first step!

LOGO/SIGN – DTMF COMMUNITY ASSISTANCE "DTMF COMMUNITY ASSISTANCE" is a future charity project name for community assistance services (to assist the community and/or community projects), and will be auspiced under a registered non-profit Charity (DTMF – Dreams To Miracles Foundation). SPECIFICATIONS: The logo/sign must be an ORIGINAL, professional-looking design, using bright colours, and reflect community assistance services and/or community spirit. Preference is for a circular-type design (with an image and the wording DTMF COMMUNITY ASSISTANCE). The design MUST be of high-quality resolution (for print/copy use). Designs must be submitted singly on plain white background (as per Entry Submission description below). * WORDING/TEXT: D...

Hi Ambiorix R., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I am looking for a custom IVR for FreePBX to ask questions, take input from DTMF, and a voice message or two, convert to WAV, then email all results, and save to database. I've never created one before, nor know how to integrate into FreePBX and be able to route calls from an IVR to this 'script'. I'm sure I could figure this out myself, but I'm trying to save time asking a pro!

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,...

english >--------------------------------------------------------------------------------------------------------------- -- SIP client as ...----------------------------------------------------------------------- -- SIP client as APP for Nextcloud with JSSIP library. The following functions should be supported: To dial a keypad 0-9 *,# Answer and hang up with DTMF transmission. 1. register extension 2. make call 3. accept call german >--------------------------------------------------------------------------------------------------------------- SIP Client als APP für Nextcloud mit JSSIP library Folgende Funktionen sollen unterstützt werden: Zum Wählen ein Tastenfeld 0-9 *,# Annehmen und Auflegen mit DTMF Übertragung 1. Nebenstelle registrieren ...

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...

We have need Android Developer for Kolkata location, This is not a freelancing job. Position: Android Developer Company : Gartech Web Solution pvt ltd., Kolkota location: Salt Lake or DumDum Job type: Permanent Qualification: Gradutation Exp: 1-3 years Skill: Java (j2ee/j2me), Database Technologies (sqlite, Mysql) development, Mobile Application Life Cycle & Mobile Ui Design Android studio, Html5 & Css3, Java Script, Xml4. Android Sdk, 5 Web Services Data Parsing using Xml, Json. Candidate should have experience in publishing App (mandatory) Salary : In hand salary 25K as per EXPERIENCE Kindly Contact us Immediate Joiner/ serious candidates Only last updated CV at (nitish dot ardorhr at the rate outlook dot com)

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. (...

find coders and fix java server errors I need to find someone who writes the server code based on an available client. client running java me (j2me). The budget is 500$ looking forward to cooperating with you, immediately

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 to find someone who writes the server code based on an available client. client running java me (j2me). The budget is 500$ looking forward to cooperating with you

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.