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    327 project based pjsip pekerjaan ditemukan, seharga USD
    PJSUA PROBLEM Berakhir 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 Berakhir 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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    ...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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    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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    im looking for asterisk pjsip developer for support our system

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    ...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 calls and video calls; - have a contact list to make calls and if possible

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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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    ...Setting up the System base currency to South African Rand (ZAR) 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

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    ...Setting up the System base currency to South African Rand (ZAR) 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

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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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    I AM LOOKING FOR A DEVELOPER WHO CAN FOLLOW INSTRUCTIONS AND WHO WOULD DO THIS AS A PROJECT BUT MUST BE WILLING TO WORK FULLTIME IF NEEDED WHEN PRODUCT IS RUNNING LIVE. This project would NOT be paid by milestonesn because 99.99% of the project is 0 to me and GARBAGE. So if you are one of those scammers looking for money upfront, stop reading now

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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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    I AM LOOKING FOR A DEVELOPER WHO CAN FOLLOW INSTRUCTIONS AND WHO WOULD DO THIS AS A PROJECT BUT MUST BE WILLING TO WORK FULLTIME IF NEEDED WHEN PRODUCT IS RUNNING LIVE. This project would NOT be paid by milestonesn because 99.99% of the project is 0 to me and GARBAGE. So if you are one of those scammers looking for money upfront, stop reading now

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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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    I AM LOOKING FOR A DEVELOPER WHO CAN FOLLOW INSTRUCTIONS AND WHO WOULD DO THIS AS A PROJECT BUT MUST BE WILLING TO WORK FULLTIME IF NEEDED WHEN PRODUCT IS RUNNING LIVE. This project would NOT be paid by milestonesn because 99.99% of the project is 0 to me and GARBAGE. So if you are one of those scammers looking for money upfront, stop reading now. I

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    Hello. One of our clients some time ago bought an app from a company based in India. As it was of a bad quality, right now my developers are dedicated to write him a new one, but this will take some time and the client needs his old one to appear on Apple Market asap (to be replaced with new one when it's ready). To make it appear there it requires

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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 untuk melihat URL] Please contact us if you have experience and you can start immediately

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    I AM LOOKING FOR A DEVELOPER WHO CAN FOLLOW INSTRUCTIONS AND WHO WOULD DO THIS AS A PROJECT BUT MUST BE WILLING TO WORK FULLTIME IF NEEDED WHEN PRODUCT IS RUNNING LIVE. This project would NOT be paid by milestonesn because 99.99% of the project is 0 to me and GARBAGE. So if you are one of those scammers looking for money upfront, stop reading now

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    I AM LOOKING FOR A DEVELOPER WHO CAN FOLLOW INSTRUCTIONS AND WHO WOULD DO THIS AS A PROJECT BUT MUST BE WILLING TO WORK FULLTIME IF NEEDED WHEN PRODUCT IS RUNNING LIVE. This project would NOT be paid by milestonesn because 99.99% of the project is 0 to me and GARBAGE. So if you are one of those scammers looking for money upfront, stop reading now

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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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    ...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 Nodes, and

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    I AM LOOKING FOR A DEVELOPER WHO CAN FOLLOW INSTRUCTIONS AND WHO WOULD DO THIS AS A PROJECT BUT MUST BE WILLING TO WORK FULLTIME IF NEEDED WHEN PRODUCT IS RUNNING LIVE. This project would NOT be paid by milestonesn because 99.99% of the project is 0 to me and GARBAGE. So if you are one of those scammers looking for money upfront, stop reading now

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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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    Bersegel
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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 untuk melihat 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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    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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    ...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 and develop the app. The IM module have to work using XMPP on our own servers. The app have to make

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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 Berakhir 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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    We've the exact same problem as this question on stackoverflow [login untuk melihat 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

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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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    Edit PJsip script Berakhir left

    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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    Need an expert in PJSIP Library to complete a video rendering project. We've gotten almost 3/4 of the way but are missing some key components. The ideal candidate for this project is well versed in rendering video on mobile for a two way/multiplex call.

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    Expert in C, C++, PJSIP Stack Must be able to analyze the Wireshark captures to identify any SIP signaling/media issues. Should have worked on Server side Should be aware of RFC 3261,3264, NAT traversal, Media Codecs AMR, G729, Opus.

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    We are looking for the following 1. VPN based - Mobile SIP Dialers 2. Tunnelling Server. 1. VPN based - Mobile SIP Dialers for blocked countries We have existing sip dialer based on CSIPSIMPLE using pjsip. We need to integrate vpn client into this. 2. Tunnelling Server - Linux based a. Bandwidth optimization b. Secure connection and communication

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    Modify the code so it works with PJSIP and Attended transfer within the asterisk C code. We can provide more details for the right developer.

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    PJSIP - Application development/ Customization

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    Need Application Developer who has worked on PJSIP for mobile app for IoS and Android Simple app with standard screens - 1. Theam change 2. Login change 3. Voice mail 4. Authentication pattern change 5. Contact 6. Setting.

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    Hi Freelancers. I need to complete Chat/Voice/Video call app now. Of course I have source code (Android/iOS/Backend). Backend is python. I will share all source code to the right candidate. Please write your favorite fruit at the top of your cover letter so that I know you are not robot. Only bid who has 100+ hours of work history. Thanks

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    I'm looking for Windows Developer who can build a SIP Client based on PJSIP lib

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