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    1,265 pjsip dtmf trabalhos encontrados, preços em USD

    ...web que faça as ligações; 2. Este sistema vai ler uma tabela do banco de dados o nome e o numero que vai ligar e gravará em uma outra tabela as informações de log, respostas DTMF e gravação do audio (se for possível); Exemplificando, tenho um sistema que funciona no cliente atualmente, faz um controle financeiro básic...

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    Identificar chamadas FXO vindas da operadora CTBC em DTMF. O serviço de identificação de chamadas está habilitado na operadora. Ambiente: Centos 7, com Elastix v.4, Asterisk 11.25, placa Openvox 1200p. Tenho conversor DTMF para FSK mas gostaria de trabalhar preferencialmente sem o conversor. Condição: Freelancer Brasileiro

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    asterisk dtmf Encerrado left

    Gerar um arquivo com os dados de log dtmf do asterisk e enviar por email com variantes

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    Olá Hoje possuo 1 servidor que...eu escolho 1 contato da minha agenda, ele disca para um número específico do meu servidor. Meu servidor retorna a ligação por outro(s) número(s), e após isso precisa enviar o dtmf do contato selecionado no inicio para a disa completar a ligação. Pode haver necessidade de inserir algum digito ant...

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    Desenvolver um circuito para converter sinais DTMF para FSK, utilizando a própria energia da linha telefônica. O serviço inclui o design da placa de circuito impresso (não a fabricação). Também deve ser provado que o mesmo funciona perfeitamente.

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    VoIP termination solution 6 dias left
    VERIFICADO

    ...registration, incoming traffic and billing Server B: filter spam and fake calls coming from clients Use IVR to filter these calls (answer all calls and ask caller to enter dtmf. no dtmf drop call) Must have on/off feature Server C: register gsm gateways using pptp vpn and load balance outgoing traffic to gateways based on number of calls For server A

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    5 ofertas
    PJSIP Compile 6 dias left
    VERIFICADO

    We are looking for someone who has compiled "PJSIP" package as ".so" library before for Android with latest version and packages! If you didn't do it before, please do not offer!

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    Goautodial specific functions 3 dias left
    VERIFICADO

    Set up following functions on Goautodial 3.3 : 1) DTMF function: when agent click on button, the client can enter credit card details by DTMF and get back to agent to continue discussion 2) Regarding any number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load

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    im looking for someone to make landing page simple nice looking landing page have form and need image

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    We are developing the app with above voip [fazer login para ver a URL] require your service regarding. We prefer the PJSIP environment Both server/client side duration of development : ? quality of voice streaming ?

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    We are using a 3cx phone system and we have some mobile-only extensi...outside number. 2.) Have some kind of UI or API call which will establish a call to the mobile extension and then provide a dial-out tone so that the number can be dialed using DTMF. Any other way to reach the same target, incl. applications in parallel to 3cx, are equally welcome

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    we need a utility app that will reside on Tizen the function of the app is a bridge betw...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 less

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    need some one to write audio port for conference bridge in C using unix file discriptors [fazer login para ver a URL] please read this and see if you understand

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    I need to fix issues within 5 hours. Currently, I am developing VoIP mobile app which written in React Native. But It has some issues. I am a developer but not good in React Native. I only need an expert not learner! Please use "imrnexpert" at first line of your coverletter. And answer please "Are you available for 5 hours from now?" Thanks.

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    I need to fix issues within 5 hours. Currently, I am developing VoIP mobile app which written in React Native. But It has some issues. I am a developer but not good in React Native. I only need an expert not learner! Please use "imrnexpert" at first line of your coverletter. And answer please "Are you available for 5 hours from now?" Thanks.

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    ...playing a set of sound files, simultaneously listen for a single frequency tone (answer machine tone) and or DTMF tones. If a either single frequency tone or a DTMF is detected we need to exit and go to the matching extension in the dial plan. EG. DTMF #1 is detected, exit to extension 1 in dial plan context. If Single Frequency Tone (Answer Machine Beep)

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    I need to fix issues within 5 hours. Currently, I am developing VoIP mobile app which written in React Native. But It has some issues. I am a developer but not good in React Native. I only need an expert not learner! Please use "imrnexpert" at first line of your coverletter. And answer please "Are you available for 5 hours from now?" Thanks.

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    I have a VoIP project which written in React native. And Have some issues. Need to fix them asap. I only need talented developer. no learner!

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    for this project i need to be able to pitch shift active channels in real-time. currently using asterisk pbx 13 with pjsip.

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    need to design front end UI/UX of a PJSIP based video call application in C++. May also include Android/iOS in later phase.

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    PJSUA PROBLEM Encerrado left

    Hi everyone, I have a problem with calls in PJSIP . When I freshly lunch my application the calls works perfectly then after 2 hours I can’t makes calls anymore and I had this message. 17:21:01.617 pjsua_aud.c .Set sound device: capture=-1, playback=-2 17:21:01.617 pjsua_aud.c …Opening sound device (speaker + mic) PCM@16000/1/20ms 17:21:01.634 ec0x629fc888

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    ... The mobile app uses PJSip to communicate with Asterisk, and interacts with a back end API created in PHP. Ideal candidate would have knowledge of React Native, Redux, and PJSip (Optional but definitely recommended). The final stages of the app are implementing payment processing with Google Pay/iOS In App Purchases, and PJSip (outsourced to another

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    laravel expert Encerrado left

    ...button) - Asterisk AMI / Call Bridge? Stop transfer calls (only when attend transfer) (click of button) - Asterisk AMI Hold/Unhold calls (click of button) - Asterisk AMI Dial DTMF (type dial and click of button) - Asterisk AMI Log in/out of Queue - Asterisk AMI View call logs Set status (available, on call, wrap up, etc..) View accounts assigned to Pre-screenpop

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    JoyGSMDial Encerrado left

    ...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

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    Zabbix fine tuning Encerrado left

    ...fine tuning. Host include: Windows 7 through server 2016 with active directory. Printers Xerox, Canon, HP Peplink routers, Balance and SOHO series FreePBX (asterisk running PJSIP) Ethernet switches Nortel/Avaya, Arista, 3com Microsoft SQL 2016 Standard with AG (to be added) Multitech rCell 100 Cellular router HP Servers with ILO (to be added) Proxmox

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    I need a IVR system which build on Rasberrypi. System should be able to decode DTMF from customer and respond accordingly (system should be able to modify the respond). System should be capable to call some Http APIs . There will 3 level menus. Normal market available sim will be attached to the SIM Module and were we expecting to receive call.

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    ...Analyzing requirements, developing detailed designs for implementation of technical work, defining implementation and testing procedures. • Designing the IVR caller experience for DTMF and speech-enabled applications. • Creating and design effective speech user interface designs including call flow, dialogues, functional logic, prompts, and define grammars

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    ...com/asterisk-java/asterisk-java). We need to fork OpenNMS (stable version 23) to support asterisk-java master branch and it to be able to monitor PJSIP registered devices (This commit adds support for PJSIP devices: Ideally this fork of opennms should be submitted as a pull request to be incorporated into official opennms, but this is not mandatory

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    Pjsip2.8 is tested with "pjsip-app/bin/pjsua-mipsel-unknown-elf" and the following error is generated during the call. Alsa_dev.c ca_thread_func: overrun! Alsa_dev.c pb_thread_func: underrun! Cpu occupancy reached 96%. I can provide ssh to let you log in remotely. Codes is wm8960,Alsa drive is ok, I can play music with madplay Can anyone help me find

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    ...am looking for a developer who has built their own voice broadcast dialer. Dialer must be able to create Press 1 campaigns and survey campaigns. Need full cdr reporting and dtmf capturing. Based on asterisk as well with admin panel to create sub users and load accounts. Again only reply if this is a platform you have built and is up and running with

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    i have an existing online banking script in php. i want an IVR application to be built so that users can call a phone number and using the DTMF, perform the following task check balance, make a transfer, change password, open an account, call customer care, etc

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    I provide a cross-compilation toolchain, help me compile PJSIP-2.8, require support for WebRTC, video CC=mips-openwrt-linux-gcc ./configure --host=mips-openwrt-linux --build=x86_64-linux-gnu --prefix=/home/pjvideo --disable-libyuv Just need to compile, do not need to make other changes, send the compiled file to me, if you can

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    I provide a cross-compilation toolchain, help me compile PJSIP-2.8, require support for WebRTC, video

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    I have a taxi company, with 500 customers and I want my clients to receive an automated call with our voice where they could key in with ther keypad. And allow DTMF to work, If that makes sense..

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    We need a simple one page android app to decode dtmf signal on call.

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    ...functionality should be: Via Nemo’s API Make a phone call to a number with a command and send the audio of it into a discord channel(.call [number]) Hang up the phone call(.hangup) Send DTMF tones into the phone call (.dial [digits]) Exclusively send the audio of a single user in the channel back to the phone call (.listentome) Required skills regarding node.js:

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    im looking for asterisk pjsip developer for support our system

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    I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load bala...version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

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    ...list - Speakerphone mute and hold - UDP and TCP transports (use TCP for better battery life!) - Supports g711 (ulaw, alaw), speex, iLBC and gsm codecs - Supports sending of DTMF - DNS SRV - Built-in echo cancellation - STUN support - Change ringtone per account - call waiting - Video support (gold users only) - ZRTP / TLS support (gold users only) -

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    Build me a program Encerrado left

    I have a project I need help with, I'm using Arduino and I want you to write a program. DTMF acoustic signal detector: listen to touch tones, decode them, and speak out the number sequence.

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    ...Sip Requirements o Register on Sip Proxy/Gateway o Receive call authenticated by IP, user/pass or no authentication. o Make calls with or without authentication o Forward DTMF digits using RFC2833 or inband o Use codec G711 and GSM o Be able to use codec G729 o Create a mini dialplan to choose which chip to use based on techprefix, per SIP Account

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    Web Call back or click to call features: When number clicked on "Free Call" button, user get connecting message and generate DTMF button on screen so user can press options from IVR and user use inbuilt microphone then connect call to SIP address on Incredible PBX OR When number clicked on "Free Call" button, viewer types in his/her phone number

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    Hi, I need a Freelance Electronic Engineer to solve this issue:[fazer login para ver a URL] llook mainly at the [fazer login para ver a URL] and [fazer login para ver a URL] (phone access and delete and to this (videos) [fazer login para ver a URL]

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    Existing Python based Plivo on FreeSWITCH project has web UI to build touchtone IVR surveys. This project is to add the support of ASR to be used in addition to or instead of DTMF tones by way of the FreeSWITCH mod_pocketsphinx. https://freeswitch.org/confluence/display/FREESWITCH/Mod+pocketsphinx [fazer login para ver a URL] Eventually another

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    Existing Python based Plivo on FreeSWITCH project has web UI to build touchtone IVR surveys. This project is to add the support of ASR to be used in addition to or instead of DTMF tones by way of the FreeSWITCH mod_pocketsphinx. https://freeswitch.org/confluence/display/FREESWITCH/Mod+pocketsphinx [fazer login para ver a URL] Eventually another

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    We'd like to allow a parked caller to press a DTMF which sends the caller to the user that originally parked them. Please provide a full project plan and how much time you'll need to accomplish this.

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    Trophy icon Logo - Dreams To Miracles Foundation Encerrado left

    Dreams To Miracles Foundation (DTMF) is a registered Charity. A sample of the current Logo is attached, and, even though the Foundation Committee likes the general idea/colours, this Logo is not quite right for our Charity. The Committee has no specific instructions in relation to what is required, however, would like the design to reflect that the

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    ...their subscription. As an example, can we identify the mobile number from the ANI of the incoming call or do we use the IVR and have the caller enter their mobile number using DTMF. Once the mobile number is obtained, either the VoIPNow API conducts a customer record lookup and validation and with an additional prompt cancels the subscription OR the IVR

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    ...their subscription. As an example, can we identify the mobile number from the ANI of the incoming call or do we use the IVR and have the caller enter their mobile number using DTMF. Once the mobile number is obtained, either the VoIPNow API conducts a customer record lookup and validation and with an additional prompt cancels the subscription OR the IVR

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