Pjsip işler

Hi Friend, My mini project i need source code to config integrate Opensource PJSIP using WPF c#. Many tutorial on Github but i duno why im not success build Dll from pjsip project and integrate it into c# project. The project result: - Guide to build PJSIP to DLL using in c# ( i try here but not success ) - WPF project import PJSIP DLL/ Wrapper C++ function by Swig to call in c# - Can call function support by PJSIP in c# - Can register and make call from WPF Version pjsip is newest on The SIP infor account i will send later

Looking for a Mac OS Developer, with experience in VoIP and PJSIP Library. Need to build a custom Softphone that can work with our SIP Servers. Here will be the Flow: Application will call an API with username and password to get the SIP Credential. Those credentials will be used to configure Softphone to register on phone system. User should be able to Choose Codecs, Add multiple SIP Accounts. We will hire only people/company who has an experience in VoIP and Sogtphone Application for Mac. Must have worked on similar project and must provide reference. Here you can find windows version.

...see this error below when I do an outbound external call, and the call will hang up itself, the mobile client out close it by itself too. I've tried this on both iphone and android client. Can any asterisk/freepbx professional provide me a quick solution for a fee? Full log: -- SIP/6566811234-00000005 answered PJSIP/90101-00000005 -- Channel SIP/6566811234-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-aece-77a2e185c45e> -- Channel PJSIP/90101-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-aece-77a2e185c45e> [2020-04-05 13:30:53] WARNING[19800]: res_http_websocket.c:508 ws_safe_read: Web socket closed abruptly -- Removed contact 'sip:1se6p552@;transport=ws' from AOR �...

I need an Android app. I would like it designed and built. I want to implement PJSIP native library and build function for sending sms and calling using voip server

same project as before in regards to pjsip

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need to register 2 extensions as PJSIP on isable asterisk

I need you to develop calls functionality using VoIP, PJSIP, WebRTC, and Opus. Please bid only if you have prior knowledge in these fields. Type "call" in proposal so I know you have read project description.

I only need to read the IPC's rtp stream (H264) and send it to the sip device at the other end. It does not need to be displayed on the local end, not received, and does not need to encode itself. Runs on armv7 platform, Linux , Not the Android app modiy pjsip-apps/src/samples/vid_streamutil.c or simpleua.c,Give me a demo, just read and forward the IPC video stream, and the project is complete

I am looking for Python Senior expert. The developer should know well about pjsip and should work in US timezone. if you are interested in the project, please write "pjsip" in the heard of your bid sentences. Autobid should be ignored.

hello, i have working python code it's depend on package pjsip i need to add function for caller id from the package to be add to command line

hello, i have working python code it's depend on package pjsip i need to add function for caller id from the package to be add to command line

hello, i have working python code it's depend on package pjsip i need to add function for caller id from the package to be add to command line

We have the issue in the production FreePBX 16/asterisk 13. After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip. FreePBX is a virtual machine with a public IP (direct). Endpoints are Acrobits sofpthone users (android/iOS) connecting via WAN. there is nothing in between. All end-users use TLS+SRTP. and Acrobits Push. So ping is huge sometimes. I suppose qualify option may be the cause here. Official FreePBX forum treads ignore the issue and ask to order their paid support. As the issue is in the production system, there is no place ...

... - but other work has left the project incomplete. I need the project completed and updated to use the latest version of , 0.15.6. You will be provided with FTP access to the current source files including the HTML, CSS and current JS files. Additionally 3 SIP accounts will be made available for testing as you progress, these are own our own SIP server running Asterisk 16 with the PJSIP stack. The current version allows for successful calls to made, both inbound and outbound, placing calls on hold and resuming those calls and call muting. The biggest things that needs to be done are call transferring, both attended and blind transferring and conference calling. I have BLF (Presence) working OK and the web phone currently sends AJAX calls to a CGI script for things like call

Hi, i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?

Hi Mohammed. i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?

Hi Ambiorix R. i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?

We have the problem with server setup for Asterisk integration with Google API. We need to solve that problem. We started the project: By default, the project deploys all the necessary components, including asterisk 13. The only change we made is the channel driver, instead of pjsip we use sip. In the asterisk configuration files, we put the service key with the role of the DialogFlow API for connecting to DialogFlow services. The project contains this library Here you can compile a test client, in which if you insert the above key, they execute the DialogFlow requests. In asterisk, an error occurs during a call, presumably related to receiving authorization data: -- Executing [asterisk-irgkin@dialogflow-loop:2]

I have code that controls a button and the LED inside of a button. Essentially I need a program that allows me to press the doorbell button and create an outbound call via SIP. The program I'm attaching has Sonos API capability and other capability as well. I need a programmer that w...call via SIP. The program I'm attaching has Sonos API capability and other capability as well. I need a programmer that will modify my existing code and create a python program that will run on a headless Raspberry Pi 2. Please create an object oriented version of my code: - Sonos capability (to play the doorbell MP3 on all speakers in a subnet) - Press button and call using SIP to SIP client - PJSIP or Linphone (whatever works best) - Change LED color based on button press (code works)...

I need a simple SIP phone. Environment is Ubuntu 18.04. Qt 5.13.1 C++ environment Complete source code will be supplies PJSIP is to be used as lib.

Asterisk and FreePBX with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk w. WebRTC, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. See attachment for issues to be solved. The system must be checked to get it ready for production. The physical Servers (3) are installed and most of the software is also setup. A late Ubuntu OS and Teamviewer is required to login to our servers. --- Project will be divided into milestones, to be discussed.

Asterisk-FreePBX with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base and who know how to get things done on our servers. First things to fix are, attended transfers and PBX calls agent that did not register for his shift. Some requirements: FreePBX/Asterisk w. WebRTC, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. There are a few more fixes to make and the system must be checked to get it ready for production. The physical Servers are installed and most of the software is also setup. A late Ubuntu OS and Teamviewer is required to login to our servers. --- Project will be divided into milestones, to be discussed.

Hi , My system contain: - Freeswitch server - Sip Client: Web using sipjs , mobile react-native using to receive call. My problem is when call done i need to know the uuid of CDR recently add to Postgres DB of that call to attach some info to that call I try many way but can not success like , write http request to select into postgres DB, but can not find exactly which uuid because one extension can make many call one time. Can anyone help me solve this case?

Asterisk with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk w. WebRTC, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. There are a few fixes to make and the system must be checked to get it ready for production. The physical Servers are installed and most of the software is also setup. A late Ubuntu OS and Teamviewer is required to login. --- Project will be divided into milestones, to be discussed.

PLEASE READ THIS FIRST ! Asterisk with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. There are a few fixes to make (see attachment). The physical Servers are installed and most of the software is also setup. A late Ubuntu OS and Teamviewer is required. --- Project will be divided into milestones, to be discussed.

PLEASE READ THIS FIRST ! Asterisk with WebRTC - A real Specialist is needed.! For Inbound call center with Asterisk and WebRTC. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and most of the software is also setup. There are a few fixes to make. First: - Assisted Transfers to agent's extensions and to PSTN lines, calls can be taken in random order and Caller must not hear himself (echo). Ubuntu and Teamviewer is required. --- Project will be divided into milestones, to be discussed.

Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and most of the software is setup. There are a few fixes to make, like Assisted Transfers to agents and PSTN lines, calls can be taken in random order, Caller must not hear himself (echo), PBX calls agent who didn't register, No-store cache control for Apache, etc. Teamviewer is required. --- Project will be divided into milestones to be discussed.

Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and most of the software is setup. There are a few fixes to make, like Assisted Transfers to agents and PSTN lines, calls can be taken in random order, Caller must not hear himself (echo), PBX calls agent who didn't register, No-store cache control for Apache, etc. Teamviewer is required. --- Project will be divided into milestones to be discussed.

PLEASE READ THIS FIRST ! -- Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few fixes to make, like Assisted Transfers, calls can be taken in random order, Caller must not hear himself, PBX calls agent who didn't register, No-store cache control for Apache, etc. Teamviewer is required. --- Project will be divided into milestones.

...in TCLASS field. == Using SIP RTP Audio CoS mark 5 -- Executing [1333@test:1] Dial("PJSIP/200-00000001", "Mobile/WP530/1333,45") in new stack -- Called Mobile/WP530/1333 -- Mobile/WP530-3531 is making progress passing it to PJSIP/200-00000001 > 0x73f2cfd8 -- Strict RTP learning after remote address set to: > 0x73f2cfd8 -- Strict RTP qualifying stream type: audio > 0x73f2cfd8 -- Strict RTP switching source address to -- Mobile/WP530-3531 is ringing -- Mobile/WP530-3531 answered PJSIP/200-00000001 -- Channel Mobile/WP530-3531 joined 'simple_bridge' basic-bridge <c6406e9f-3653-48a4-bd0d-fdeceae7eac3> -- Channel PJSIP/200-00000001 joined 'simple_bridge' bas...

PLEASE READ THIS FIRST ! You must use a PC with a late Ubuntu OS. -- Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, FreePBX/Asterisk, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few upgrades and fixes to make. Teamviewer is needed. --- Project will be divided into milestones.

PLEASE READ THIS FIRST ! -- Inbound call center with Asterisk and WebRTC for agent access. This is only for professionals. Some requirements: Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, FreePBX/Asterisk, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few fixes to make. --- will be divided into milestones if needed.

3 server System, for an inbound call center, with Apacbe, Asterisk, WebRTC, MySQL, on Ubuntu OS. First thing to be fixed, is for agents to take calls that are in a queue, on our Operating Panel. This is for FreePBX / Asterisk specialists ! Some requirements: Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, FreePBX/Asterisk, SSL, Ubuntu, etc. -- Project has a few other things to fix and will be divided into milestones.

Need help in asterisk pjsip , Experience needed 1. Asterisk PJSIP Registration 2. kamailio 3. Rtpengine 4. NAT

Not interested Indian, Pakistan, This job is for US based developer, Thank you! This is a new app for international calls. Starting from scratch. We'll use REST API, SIP (liblinphone or pjsip or WebRTC). We use git, Trello, Figma to collaborate on the project. More details available upon request.

Language: Javascript (NO others) Use I need a simple proof of concept for a sip client on a html5 mobile device with android. We are using an asterisk server, and can provide a pjsip or sip extension. I want run the webrtc code as a function on a bigger web page based application, so you can create a simple , and contain the phone display, logic and functionality inside of a div. I want it to pop up a div when an inbound call comes in, with an "accept", and "deny" button, just as you might expect if you were receiving a phone call. There should be a way to hide/show the div. There should be a little div, with the most current status displayed This div should also have a "call" button, which triggers a dialpad appear on he screen, with a dial

I want a module of godot game engine for pjnath library of pjsip project. Especially, that module should work on android and iphone. Please not disturb me with general proposal. I want to work with an experienced developer.

Using electronjs build a windows/mac compatible softphone using SIP(PJSIP) and/or WebRTC(chromium) with desktop notifications. Simple interface to include Dialpad, Phonebook, History, Instant Messaging, Video.

I have a setup of kamailio and Asterisk, registering works, but making calls not. Sip signalling is delivered to asterisk but there is some rtp issue. I need somebody, who will look at it and repair this setup. Kamailio 5.1 and Asterisk 16 with pjsip

Need an iOS SIP client based on pjsip 2.8 library. iOS client app should seamlessly work with asterisk 15.4.1 i.e should handle registration with authentication and support voice calling functionality. iOS client app should handle NAT64 as needed by Apple for app approval. Code should be in Objective C

Our SIP Trunk provider recently made some changes overnight. Originally, we were configured for PjSIP, and could not register. After changing to chan_sip, we could now register, however can only make outgoing calls. When we try an incoming call, the SIP trunk provider tells us that our FreePBX instance is returning 401/unauthorized to them, and so incoming calls are not working.

I need a freelancer pjsip expert who can build PJSIP stack with SRTP support and TLS support.

This project will be written in Java and have VoIP calling feature. I don't want to lost my time. So I need to work with a Serious developer. If you are experienced in Android Java and VoIP SDK, Please ping me. Before submit proposal, Please check shared source code and start your cover letter with "iknowvoip". This will be a long-term contract. Looking forward to hearing from you! Thanks.

Please check this open source : This project is based in React Native and have VoIP calling feature. It has Old React Native version so current have serious issues for launching. I don't want to lost my time. So I need to work with a Serious developer. If you are experienced in React Native and VoIP SDK, Please ping me. Before submit proposal, Please check shared source code and start your cover letter with "iknowvoip". This will be a long-term contract. Looking forward to hearing from you! Thanks.

I have a React Native based VoIP calling app but current have serious issues when dialling. Need to work on VoIP SDK core. So I need to work with a Serious developer. If you are experienced in React Native and VoIP SDK, Please ping me. Looking forward to hearing from you! Thanks.

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!

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

We are developing the app with above voip require your service regarding. We prefer the PJSIP environment Both server/client side duration of development : ? quality of voice streaming ?

need some one to write audio port for conference bridge in C using unix file discriptors please read this and see if you understand