Pjsip android source code trabalhos

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    336 pjsip android source code trabalhos encontrados, preços em USD

    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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    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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    9 ofertas

    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.

    $62 (Avg Bid)
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    3 ofertas

    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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    15 ofertas

    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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    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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    OpenNMS ([fazer login para ver a URL]) is an open source monitoring and systems management platform written in Java. It includes support to monitor SIP devices registered to and asterisk server and it does it by connecting to Asterisk's AMI interface and querying for SIP registrations. If the list of registrations change, OpenNMS generates an

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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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    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

    $132 (Avg Bid)
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    2 ofertas

    im looking for asterisk pjsip developer for support our system

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    ...SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step of the project: - make and receive

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    I need Sip Dialer With Include OpenVPN. Sip Dialer connect by OpenVPN. OpenVPN Connect only Sip dialer..... you need use PJSIP OpenSource SDK for SIP Dialer. I need Very Simple Apps... i need support g729 voice code. I can you OpenVPN server side. if you have good experience please contact me.

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    I need android VoIP Softphone With Include VPN .. You need use PJSIP OpenSource SDK for Softphone.... You can use any VPN technology for softphone. I need Support g729 voice code. I need Good Voice Quality. If you have Old Experience for Softphone with vpn then contact me.

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    ...so it updates all other currencies to cost based on the ZAR price through the standard Yahoo Finance Integration already present. 2. Ability to create new Realtime ChanSip, PJSip and IAX accounts that FREE PBX can connect to without needing to re-load A2Billing. (DB Accounts) This should however not affect the Free PBX configuration 3. Configure and

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    ...so it updates all other currencies to cost based on the ZAR price through the standard Yahoo Finance Integration already present. 2. Ability to create new Realtime ChanSip, PJSip and IAX accounts that FREE PBX can connect to without needing to re-load A2Billing. (DB Accounts) This should however not affect the Free PBX configuration 3. Configure and

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    We need to developed a better echo cancellation for our PJSIP base applications on PC which currently using speex echo cancellation which is not working very well. We need to hire a person who have experience on write AEC for us or improve on speex AEC so it will work well on PC applications.

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    I need a echo canceller for use with PJSIP I need funcional exemple in C/C++ to connect ECHO CANCELLER with PJSIP VOIP LIB Ill use in various platforms.

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    ...admin /management system must be extensive and very detail A rotating banner ads system would be added to the product( to be discussed before adding) Here is our old app on android, it shows the watermark and a rotating banner ads [fazer login para ver a URL] I would like to keep [fazer login para ver a URL] We will be implemented

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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

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    .../management system must be extensive and very detail A rotating banner ads system would be added to the product( to be discussed before adding) Here is our old app on android, it shows the watermark and a rotating banner ads [fazer login para ver a URL] I would like to keep [fazer login para ver a URL] We will be

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    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

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    We need an iOS developer and Android Developer,Who posses experience in PJSIP integration for Video call.

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    ...admin /management system must be extensive and very detail A rotating banner ads system would be added to the product( to be discussed before adding) Here is our old app on android, it shows the watermark and a rotating banner ads [fazer login para ver a URL] I would like to keep [fazer login para ver a URL] We will be implemented

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    iOS dialer app fixes Encerrado left

    ...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: [fazer login para ver a URL] I am interested in quotes for fixing "High Priority" issues listed on Issues

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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: [fazer login para ver a URL] Please contact us if you have experience and you can start immediately

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    .../management system must be extensive and very detail A rotating banner ads system would be added to the product( to be discussed before adding) Here is our old app on android, it shows the watermark and a rotating banner ads [fazer login para ver a URL] I would like to keep [fazer login para ver a URL] We will be

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    39 ofertas

    .../management system must be extensive and very detail A rotating banner ads system would be added to the product( to be discussed before adding) Here is our old app on android, it shows the watermark and a rotating banner ads [fazer login para ver a URL] I would like to keep [fazer login para ver a URL] We will be

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    ...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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    ... we need someone to understand the code-base and continue working forward. The trading platform comprises of usual Front-End, 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

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    7 ofertas

    .../management system must be extensive and very detail A rotating banner ads system would be added to the product( to be discussed before adding) Here is our old app on android, it shows the watermark and a rotating banner ads [fazer login para ver a URL] I would like to keep [fazer login para ver a URL] We will be

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    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

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    Secreto
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    ...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: [fazer login para ver a URL] Work expected: - Make necessary changes to successfully compile the PJSIP source code that is available in the above link. - Update OpenSSL

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

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    I'm looking for Android and iOS developers to make a social chat/calls/videocalls/paycalls app. We already have the design, so you have to program everything. IMPORTANT: The connection for the calls, pay calls (to landlines and mobiles) and video calls must be using PJSIP on our own servers, we will give you the access to them to makes test

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    I need you to develop some software for me. I would like this software to be developed for Windows . Sip gateway in pjsip that receives call from my itel softswitch and forwards to viber desktop.

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    Write some software Encerrado left

    I need you to develop some software for me. I would like this software to be developed for Windows . Sip gateway in pjsip that receives call from my itel softswitch and forwards to viber desktop.

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    ...[fazer login para ver a URL] If we answered the video call from CallKit, we'll get OpenGL renderer initialization error and not be able to enable video after switch from CallKit to the app. Here is the log: 11:38:56.632 pjsua_call.c !Answering call 0: code=200 11:38:56.632 pjsua_media.c ...Call

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    We are trying to get customization to PJSIP source code to be able to do transferring using PJSIP within Asterisk. I have a requirements doc but need to interview anyone interested in the project before I send it to them.

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    i have pjsip code to do call throw command line i need to add trasfer call feature to it so when call connected it automatic transfer anther number Require : Knowlage of PJSIP package *Job urgent today

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