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    1,209 pjsip python 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...

    $671 (Avg Bid)
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    6 ofertas

    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

    $140 (Avg Bid)
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    1 ofertas
    asterisk dtmf Encerrado left

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

    $447 (Avg Bid)
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    3 ofertas

    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...

    $150 (Avg Bid)
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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.

    $308 (Avg Bid)
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    Есть процесс ffmpeg с входным сигналом,нужно по DTMF метке подмешивать другой источник. Возможно написать скрипт на python и подмешивать в pipe через stdout. Рассмотрю иные варианты

    $188 (Avg Bid)
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    6 ofertas

    Pjsip c/c++ Voip doorphone echo cancelation and noise cancelation example I develop doorphone application with embedded Linux using pjsip but I have problem with Ambient noise and with echo. I need solve

    $350 (Avg Bid)
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    Voip app Qt/QML PJSip c/c++ example for Android /iOS and Windows. I need too example of echo cancelation for use in doorphone e with pjsip

    $252 (Avg Bid)
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    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio to google speech api (file or stream) 3. get the recognized text back Silence should be detected to stop the file recording or the stream to google Websockets might be used as well

    $523 (Avg Bid)
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    ...can call orher app members or other non app numbers, they would also be able to receive call from any phone or device including mobile and landline phones We can use the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling

    $997 (Avg Bid)
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    35 ofertas

    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio in chunks to google speech 3. get the recognized text back I would like to either detect the silence to send the audio to speech service or continuous send and get the text as it is ready

    $528 (Avg Bid)
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    We need an iOS developer and Android Developer,Who posses experience in PJSIP integration for Video call.

    $693 (Avg Bid)
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    6 ofertas

    ...they can call orher app members or other non app numbers, they would also be able to receive call from any phone or device including mobile and landline phones We can use the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling

    $1757 (Avg Bid)
    $1757 M\u00e9dia
    12 ofertas
    iOS dialer app fixes Encerrado left

    ...do not have time and free workforce in-house as we need to focus on developing new app from scratch. I want to be transparent here about quality of the code, so as it uses PjSIP it must be GPL2 and you can find whole source code here: [login to view URL] I am interested in quotes for fixing "High Priority" issues

    $540 (Avg Bid)
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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

    $1500 (Avg Bid)
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    1 ofertas

    We would like you to help us to make a Java project who was able to do a call throught a VOIP channel and play a text (text2Speech) and recognise DTMF tones that the person would do in his keyboard. For example: Hello Mr John Smith, press 12 if you have received the package otherwise press 49. Can you help us?

    $762 (Avg Bid)
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    We have a VoIP based on PJSIP, which works ok on Android 7.x but on Android 6.0.1 it freezes when receiving a call. According to the debugs, it is a problem related to PJSIP and access rights: [login to view URL] Please contact us if you have experience and you can start immediately

    $55 (Avg Bid)
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    Simple script Encerrado left

    Write on python 2 or 3 simple script: -Connect to sip server with credentials (username, password from sip) -wait for call -answer a call -send DTMF signal (for example 1234) -end call

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    Create a Customized IVR solution For references examples are : IVR solutions used in Banks, Railways, Food chains etc. Software must include both - DTMF signalling and IVR Language requirements = C# , VOIP , .NET

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    ...announcements to a caller and have the two way conversation recorded along with the announcements. It seems easy to record a conversation and easy to payback a wav file when a dtmf key combination is pressed (I worked out how to do this).. but is seem very tricky to do both. I have tried monitor and mix monitor. The problem is either the announcement

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    ...can call orher app members or other non app numbers, they would also be able to receive call from any phone or device including mobile and landline phones We can use the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling

    $2497 (Avg Bid)
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    41 ofertas

    ...Miracles Foundation (DTMF) - with some form of artwork included in the design (perhaps something to reflect the DTMF charity name, creating miracles, though this is definitely not mandatory). Font/words must be easy-to-read, and design preference is for a colorful design (preferably not black/brown/beige or dull colors). DTMF is a registered charity

    $153 (Avg Bid)
    Destacado Secreto
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    87 inscrições

    ...receive calls from GSM and forward to SIP. The application should then forward the audio and convert from VoIP to GSM and vice versa. The app also needs to be able to recognise DTMF tones when calls come from SIP. General Deliveries - The application working in APK format - Full source code with detailed comments - Documentation - Detailed manual for

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    Hi, I need help to include this library: [login to view URL] in my Android things project. I have the protoboard and the components of this site: [login to view URL] mounted here and I will replace the Arduino with the [login to view URL] card and I will show the digits typed / received from the telephone line on the LCD monitor of this card's Kit using GOERTZEL ALGORITHM

    $25 (Avg Bid)
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    1 ofertas

    ...can call orher app members or other non app numbers, they would also be able to receive call from any phone or device including mobile and landline phones We can use the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling

    $1301 (Avg Bid)
    $1301 M\u00e9dia
    25 ofertas

    ...(Like as auto dialer/Vpoll). From pbx panel and from URL with cron) -- Customer can upload data and voice. System call thats with channel limit. Called person answered with dtmf 1/2 or repeat again. After finish send an email report. -- Customer set an URL and time from pbx [login to view URL] will import data from this link and create, call and report.

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    19 ofertas

    I want to make a project through which i can give option to user to listen information based on DTMF and also enable users to patch the call with Support Team.

    $22 - $187
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    0 ofertas
    Trophy icon Smart Calling Card Application Encerrado left

    ...features such as ability to select Outbound CLI  Change Status associated with Extension. Busy, In Meeting, Out of Office (This would trigger preconfigured call routing rules)  DTMF digits could be sent as data or faster tone dialling...

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    Garantido
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    1 inscrições

    ...must not bridge the final connected call. Asterisk must remove itself from the call-flow once the call has been transferred. The other pbx is currently setup under SIP and PJSIP with the same remote ip address. We are using the Asterisk pbx as an auto-attendant only. Our main pbx will send calls to the asterisk pbx and expects the calls back as

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    15 ofertas

    Hi, we have develop an call blasting application based on php and we are using freepbx and asterisk in backend. we want to enable DTMF response to be recorded in callblasting thanks

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    ...Middleware and Backend architecture, where are exposing a customizable interface to our end users (Traders), via a Windows Client Application. The VoIP implementation is based over PjSIP, connecting to tiny Asterisk based nodes, and serving voice control mechanism via the Windows Client. The Windows Client takes care of decision making with respect to Voice

    $15 / hr (Avg Bid)
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    ...page and searchable destination - NOT ACCEPT Incoming call - NO Contact list address book - Dial Page Separate screen as block layout to show (Video, Jpeg, GIF) - Can sent DTMF tone while on call (e.g. press for extension number) - History will sent as txt files to SQL database on Internet (date/time/serial no/destination phone number/duration) -

    $564 (Avg Bid)
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    6 ofertas

    ...can call orher app members or other non app numbers, they would also be able to receive call from any phone or device including mobile and landline phones We can use the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling

    $1815 (Avg Bid)
    $1815 M\u00e9dia
    23 ofertas

    When someone leave a voicemail message, I'm currently using the vm notifier to call a cellphone and announc...I'm currently using the vm notifier to call a cellphone and announce there's a message into the voicemail. Now, I want the vm notifier to call a pager and on answer send DTMF digit like 1234. So my pager would show me 1234 alert. Thanks

    $81 (Avg Bid)
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    4 ofertas

    This project is a series of task around signal processing; it essentially involves DTMF encoder, simple bandpass FIR filter design (window based method), a scoring function etc. Some examples will be provided of the work I would like, as well as much more information.

    $36 (Avg Bid)
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    5 ofertas

    This project is a series of task around signal processing; it essentially involves DTMF encoder, simple bandpass FIR filter design (window based method), a scoring function etc. Some examples will be provided of the work I would like, as well as much more information.

    $36 / hr (Avg Bid)
    $36 / hr M\u00e9dia
    14 ofertas

    VERY IMPORTANT This Project is for UK developers only. Outside UK proposals will be ignored. We are looking to for experience app developer that has experience with PJSIP CSIPSIMPLE and Phone Gap. We are looking to add WiFi option so users can call from our app. We also need to add some additional features into our existing Phone Gap app

    $200 - $300
    Secreto
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    0 ofertas

    we have: asterisk 11 on freepbx 2.11 on Centos 6.5 serv...all our extensions are recorded by default. (include internal calls and calls from queue) we need to enable the agent (extension) when he is on a call to be able to press on DTMF like (star,2) and pause the recording. when he wishes he can press again (star,2) and continue the recording.

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    The proposed project aims to drive a vehicle using a mobile phone. The project requires two mobile phones, one to control the robot that sends DTMF commands to another one mounted on the robot vehicle. A DTMF decoder interfaced with the microcontroller of 8051 families receives the commands and with the help of a motor driver operates the vehicle movement

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    ...communicate with monitoring station over PTSN line, and use Ademco Contact ID, DTMF protocol to send data from Alarm system to Monitoring center. We need to make SIP Client software for simulate Alarm Center receiver. Data what we receive from alarm systems and need to send this DTMF data to RS232 This software can making on raspberry ENIGMA II

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    Hello Everyone, I need somebody who can compile again PJSIP library for PJSUA with all support library similar to following link with latest OpenSSL library. CSipSimple Sample: [login to view URL] Work expected: - Make necessary changes to successfully compile the PJSIP source code that is available in the above link. - Update

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    Based on open source Pjsip to build a softphone. To enable Audio layer on Rpi, and use USB mic. No desktop GUI, only a service daemon, and simple web interface or API to interact. Video support as well. Can consider based on Linphone as well Please check attached PDF file here. There are a lot of tutorials and instructions how to enable audio

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    Destacado
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    We need a cloud based calling system through which we can provide a helpline number to our clients and whenever the client calls he/she should attended by an automatic voice service and when the client press dedicated key then the call should transfer to our respective employee

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    4 ofertas

    We have several SNOM M25 and M65 Phones connected to a SNOM M700 Base. The Snom uses FreePBX Asterix 14.0.x (latest) as SIP-Gateway via PJSIP connected to SIP Gateway of German Telekom. All the Phones are properly connected and the Peers registered. Somehow, we can't manage to dial out or to be dialed in and we don't know why. Need an SIP Expert to

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    Trophy icon Design a Popup / HTML page Encerrado left

    We need a web page design for popup. The said popup will be used as a GUI to web phone, th...4 In Active Call --------------- * __title:__ In Call with: {Caller Name <caller_number>} * __button:__ Load Contact * __button:__ Hangup * __3x4 array of buttons:__ DTMF (1-9 and * 0 #) * __dropdown select box:__ Select an Extension for Call Transfer

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    ...PulseAudio server to connect to, which devices (and at which volume) to connect the sink+source to, whether the connection is bidirectional/monitor/announcement, whether in-band DTMF is recognized, etc. The code shall include a "pulse XXX" command for the Asterisk command line which is able to show active channels and their state (including Pulseaudio

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    ASTPP Behind NAT Encerrado left

    I need help to resolve problems with ASTPP behing NAT. The call complete but don't has audio or DTMF.

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    need to create a dialller which have an option for external dtmf,got sample to look at

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    ...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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    24 ofertas

    i need python developer to code me custom script for pjsip package

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    3 ofertas