Callback asterisktrabajos

I have asterisk pbx system and I am looking to update it by adding a phone application. for example, if the phone is not answered in 3 rings it starts ringing in the phone app.

...connection is estabilished, we should pick the h.264 video and the audio flows that come from the server and forward it towards a remote RTMP server using container formats like mp4 or flv. We can reuse the FreeRDP's code and be able to catch raw H264 content by implementing our own version of the SurfaceCommand callback (), there should be a little bothering because it catches all calls for surfaces, the callback that is wired by default is this one (). So we implement our own and just pick the codec we're interested in (H264 AVC420, AVC444v1 and AVC444v2). Then, once we have the h264 frame, before sending them to the server, since RTMP accepts

I need a experienced one to develop an Asterisk Open CTI connector with Salesforce CRM. SalesForce Lightning Platform Starter requirements - Enterprise Edition.

...ability to run asterisk 1.4 on the Community Edition of Clearos. the OS can be found (with the MARIADB installed) Asterisk files can be found: https://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.11.1+ Asterisk should be compiled with the following options files ( and ) NOTE: res_crypto MUST be included as a requirement Deliverables must include a step by step reproduce-able solution that compiles and allows the Asterisk to be run from the asterisk -cvvvvg command

... Actions · Good Use: Creating a portal. · Example: Modify, Add, Remove user/group/enterprise preferences and details. (Cannot be used for creation. Only Provisioning) · TEST: · More: · Call Logs · Enterprise Directory · Custom Contacts · Calls · Dial · Automatic Callback · Call Me Now · Call Pickup · Services · Call Park · Call Transfer · Music on Hold · Do Not Disturb XSI over HTTP The XSI-Actions is a Web Application that runs in a container like the BroadWorks Xtended Services Platform (XSP). XSI-Actions is responsible for authenticating the requests received from the remote application. It is also responsible for providing request and response mapping and

Deliverable - successful integration, code, documentation and training Integration functions: Develop API for integration between Cloud PBX and CRM Make calls from CRM interface in one click (Desktop App, Browser, App) call statistics integration with call recordings on CRM Callback features Please provide following details when you are quoting price 1. How many years of Experience you have in Third party API Development 2. Please provide description of APIs developed by you. Please specify telephony integration APIs along with others. 3. How many year support you will provide for API developed. Any bugs that arises need to be fixed

Configure server kamailio with modules : NAT TLS Integration MS Teams Register SIP User Local. Options Forwarding SIP Register to Asterisk Call Routing, OUTGOING. Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365 INBOUND MS Teams -> Kamailio( Integration Asterisk) -> Softphone Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365

Hi Nikita N., I would like to know if you have a couple of hours today to assist me with something I am struggling with in a JavaScript (node.js and ) backend script that I am busy with. My specific issue is arround having the functions in the script wait for other functions to finish before executing the next statements (I think it has to do with async/callback? but it is proving above my skill level). One of the functions is using axios to send a request.

Hi Jiahui T., I would like to know if you have a couple of hours today to assist me with something I am struggling with in a JavaScript (node.js and ) backend script that I am busy with. My specific issue is arround having the functions in the script wait for other functions to finish before executing the next statements (I think it has to do with async/callback? but it is proving above my skill level). One of the functions is using axios to send a request.

Hi Iana G., I noticed your profile and would like to know if you have a couple of hours this morning to assist me with something I am struggling with in a JavaScript (node.js and ) backend script that I am busy with. My specific issue is arround having the functions in the script wait for other functions to finish before executing the next statements (I think it has to do with async/callback? but it is proving above my skill level). One of the functions is using axios to send a request

I have a CRM created with php codeigniter framwork, I need to start calling via this crm (making outboud calls), so by connecting it to an asterisk system and a sip account.

Hi Viktor S., are you expert in troubleshooting a "toll fraud" attack on our asterisk/freepbx server?

Hi Anton F., are you expert in troubleshooting a "toll fraud" attack on our asterisk/freepbx server?

ATI is looking for development, implementation(s) and support of Asterisk, Flexisip, Turn/Stun, Linphone and Library repository (IOS only). Phase I will be slight modification and branding of the GUI and App. Setting up the core on our test servers, placing the App in the App store and testing the total solution. Phase II will involve deploying the solution at multiple customer sites and programming specific to each client. Phase III will include further modification of the App and building a management console (node.js) that will be deployed at each location. Phase IV is continued support, modification’s and future deployments at new customer sites. Key components: • All services and applications running on our client’s servers and network. • Setup of cli...

...release the call and continue the transfer - x202 still sees "201" as Caller ID, until he picks up the call. Once he picks up the call, 222-222-3333 appears on the CID. I want to change the way the last step in the above scenario works. Once x201 presses Transfer again to release the call and continue the transfer, x202 should see 222-222-3333 on his CID display. This worked for us when we used asterisk. In , setting connected_line_method=update made this work correctly. If anyone has the right experience, I will be happy to hire you for this project. I do expect that you already have the experience and not wait to do the research after you are given the job. If you already know how to get this done, I don't expect to spend more than an hour with you working on ...

Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem happens when the person who is using the Webphone has intermittence of internet, generating multiple connections in closing state that are observed in the Kernel log. We think that to solve this is to use an Opensips such as Webrtc Gateway and that the flooding is controlled from this point. We need a professional to help us install and configure an Opensips as a mid-registar WSS, which allows us to log the extensions found in Asterisk of the SIP type. Observation: The asterisk actualy is not Real Time, and for local devolpment with cannot change that.

Need to Confgure Goautodial V4 for a press 1 survey campaign and a IVR Blaster Campaign. need to configure dynamic login to self authenticate and stop sip attacks on the server

Currently Mailchimp is setup for a subscription services in AWS Lambda (see example bellow). Now id like you to setup the same thing for Hubspot. This would need to be done in the next couple of hours. var Mailchimp = require('mailchimp-api-v3'); // mailchimp environment variables var API_KEY = '----', LIST_ID = 'a8a412dc50'; // function for subscrib... (err); const response = { statusCode: 200, headers: { "Access-Control-Allow-Headers" : "*", "Access-Control-Allow-Origin": "*", "Access-Control-Allow-Methods": "OPTIONS,POST,GET" }, body: ('failure'), }; return response; }); return res; } = func...

I m working in voip in Morocco from 2014 I have a system of asterisk Isabel the problem is when my client send me trafic give him an erreur 486 must be 503 In this case my client lost his traffic https://www.freelancer.com/users/l.php?url=https:%2F%%2Fusers%%3Furl%3Dhttp:%252F%%252Fpt%252FA42hBUju%26sig%3D50c67e1b3a7f27f20771fd2ee320f07b9eb302904f9e19edea3a28d328aafbdf&sig=dbffe148af4b5367f232a97a6acda81aea4364dfabcb09ccf07b413f3f1c3be9

Need a complete SKIN customization of opensource microsip. need also a little button that send information about operator status directly to asterisk pjsip server. will send details about image during chat.

...conditions) which will filter the data frame by the rows which its column (add_to_general_flag) =1 , and when the user hoover by the mouse in any of the filtered row it will show the values in the column(Subsetions)-attachment4 You can see attached-2 as a sample of what I want Bokeh library with version 1.40 can implement the same, if it can be implemented by plotly dash it is okay Custm_js callback to implement the above...

I have a simple iOS App using SwiftUI. I have tested with some librarys. I got 2 libs to work. First Library: works, including callback. but it crashes frequently. second library is written in standard swift, it works, but I have problems handling the callback. So I need to get help on this

I am looking for Java Spring developer who has more than five years of experience. I need you to work full time working(8 hours/day, 5 days/week), we will make a payment based on your hourly working time. We are making a web-based application to authenticate users. The following skills and understanding are needed. - Spring MVC - Restful API - HTML, Callback - Preferred: Encryption/decryption using SEED, AES, etc and Hash(SHA), RSA(Symmetric-key algorithms are algorithms) - JPA(Hiberanate) - Git source management - MySQL

I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: You will probably also need to check the G729 codec configuration

Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (Dinstar gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted ...

We are trying to build a new Asterisk server using PJSIP instead of the older SIP module and need some help to make it work. The setup is as follows: Asterisk certified/16.8-cert3 running on CentOS 8.2 with a public IP address. Phones in the office need to connect to the Asterisk server, they are on a private LAN (192.168.1.x/24) and access the Internet via a NAT router. Two types of phones: Aastra 6757i and Grandstream N300. The server also needs to send and accept calls over a SIP trunk to another Asterisk server also on a public IP (PSTN Provider). You will need to be an expert with Asterisk and PJSIP and work with us to get this working and explain what you did - you can have SSH access to the Asterisk server if required. For the right pe...

I am building up an Arduino ESP32 software and need some help with the code structure and having problems setting up the code with different header and .cpp files. I have taken the standard BLE server app and split the defi...it cannot find the services reference. I am sure this is really simple but I need to use global variables though out the software, as it's going to be easier to read and write data in the BLE call back and then manipulate it though the software. I have a global variable header folder, but I cannot get the variables to work inside the BLE callback. So I would like the following: 1. Setup global variables though-out the software i.e. main loop, BLE callback etc 2. How to setup a separate file where I can define all the characteristics and call it fr...

I need the “Referral ID” cell moved to the woo commerce checkout page, seen in first image.  When a user signs up the first time there is this cell “Referral ID”. Because ou...“Email Address” and above the Additional Information.  The only current connection with the PAP API where it feeds in data to PAP is on the registration sign-up page. So to connect that, I believe you would need to create a connection with woo then update that PAP profile using the API on the checkout. Hopefully it is simpler than this all sounds. Also my last web developer wrote “there’s a callback function in (around line 120) that calls the Post Affiliate Pro API to create an affiliate user using the same form data that ‘User Registration’ u...

...es/BMO/(124): FreePBXmodulesCertman->__construct(Object(FreePBX)) #4 /var/www/html/admin/libraries/BMO/(37): FreePBXSelf_Helper->autoLoad('Certman') #5 /var/www/html/admin/libraries/BMO/(298): FreePBXSelf_Helper->__get('Certman') #6 /var/www/html/admin/libraries/BMO/(39): FreePBXHooks->preloadBMOModules() #7 /var/lib/asterisk/bin/retrieve_conf(81): FreePBXHooks->updateBMOHooks() #8 {main}...

1 Page Agency Website For Lead Generation Digital marketing agency company name/Services/contact us form supply content supply images if needed 4.4x services with buy now or schedule callback buttons connected to paypal for 1 time and recuring charges-may need to configure paypal now connected to calendar (we supply code to embed) widgets on website (we supply code to embed) All devices friendly Ready to upload on WordPress WHAT IS 1+1 =?

Hello, We need to create extensions and edit some params through an api. The parameters we need to be able to edit/create/delete are; -Extension number -Extension alias (display name) -Direct DID -Outbound DID -Inbound Direct DID -Password -Follow me settings (initial ring time, followme list, ring time, confirm calls) Our Asterisk is an Issabel distribution.

We need to develop a softphone application to work with Asterisk freepbx.

...Social Media: Restream + Facebook Live - - Multi-Streaming Within Your Grasp! - StreamerSquare - Read Less ://dauth.user.ameba.jp/login/success&state=06f601f94db41faa782dd7b6178d9ce3&registration_type=da#

...Social Media: Restream + Facebook Live - - Multi-Streaming Within Your Grasp! - StreamerSquare - Read Less ://dauth.user.ameba.jp/login/success&state=06f601f94db41faa782dd7b6178d9ce3&registration_type=da#

We have a bug in our implementation of Kamailio + Asterisk + Mysql Proyect, we don't know why, but the Asterisk crashes with Memory problem.

Fix audio issue on Asterisk. Some calls have no audio.

Hi there. I have a really simple problem. I need help with a callback function, it's pretty simple. Screen share with TeamViewer. It's almost working, just need 1 tiny bit of help. 20 mins work I think.

...looking for a developer who can create a WordPress plugins to integrate different social logins. This is long term project and it is for quick developers. I am not looking for agencies. I am looking for honest freelancing developers who can be also part of our team for further development. Similar to OneAll Social Login but should store data in only running WordPress server. No third party server & callback will be used. A developer must be aware of WordPress plugin guidelines. We are looking to build currently for free version of plugin for our site. We have used OneAll plugin to our site But I don't have any privacy of users data there. It should not store users data to any third party database. We want currently Google, Facebook, Github & LinkedIn integrations

...familiar with the ARCamera object and the orientation or transformation APIs. _______________________ We use machine learning to generate clothing measurements. For this job, we'd like you to write a preprocessing method using the depth data provided by the ARCamera object. 1) We'll provide an app that collects camera data and calls a callback with a binary human segmentation, depth data, camera object, etc 2) You write the implementation of that callback: process the segmentation and return the modified CVPixelBuffer Overview: Use the depth data to model the image in 3D, and generate an image viewing the 3D data at a different angle and location. More details are in the slides here (also attached as a PDF): https://docs.google.com/presentation/d/1O15btQL4lZ...

We want to have sip trunking between Avaya and asterisk pbx.. If anyone have experience and stays in Jeddah let us knw

Freelancer willl provide expert advice for GOIP equipment and settings. Please let me know about your experience.

I need to install and configure Asterisk

] WARNING[21199][C-00000007]: func_base64.c:79 base64_helper: Syntax: BASE64_DECODE(<data>) - missing argument! this is what i get when i am supposed to receive a ussd asterisk 16

Objective: Develop n...Jetson Nano/Jetson Xavier NX. Current FFMPEG command used: ------------------------------------------------------- ffmpeg -i rtsp://URL -vcodec copy -acodec copy -map 0 -f segment -segment_time 300 -segment_format mp4 "ffmpeg_capture-%" Technical Challenges: ----------------------------------- - IPCamera rtsp is used to store continuous video clips with x seconds duration. How to get callback when individual clips are written by FFMPEG in python. - Stream playback stream (between start and end time) continuously to web browser Deliverables: -------------------- 1. FFMPEG clip written to disk & Store clip path to SQLite 2. Stream live video and list of clips as a continuous stream (MJPG, Webrtc) 3. Compile to Cython for efficient ru...

Custom Themes for Vicidial and Call flow changes for asterisk for Inbound Call Centre SIP based Outbound Call Centre Sip based IVR UI based Predictive Dialer Pls share the demo of your work on Vici Dial

We are getting an error on a woocommerce install call_user_func_array() expects parameter 1 to be a valid callback, function 'register_my_session' not found or invalid function name Some basic functions aren't working

I have a server with all my asterisk files that I need someone to fix up and make admin panel for me

...a POC of using Asterisk for our phone support. I am an experienced software developer and an engineering manager, but I have no experience in Asterisk or PBX. This project is to run Asterisk on a new Ubuntu/20 server running on AWS, listen on SIP calls and handle a few basic operations: 1. Voice greeting, coming from mp3 or other audio file. 2. Connectivity to Google Voice Recognition, listening for next commands 3. Logging the recognized voice in text and in audio (mp3 or other audio file) 4. Responding with another greeting from mp3 files, depending on recognized voice (3 options: option A option B or unrecognized) Also, we own 2 VOIP DIDs in , which has SIP connectivity and DID POP. We would like to connect those, so that calls to these VOIP will reach the As...

I'd like to create an app for Zoho that will be listed in marketplace.zo...is/api 1. User signs up at , purchases a PRO subscription and obtains their API Key. 2. User enters their API Key in the app and configures their custom checker page (in the app as well), including a callback URL. 3. When end-user browser details are needed for a specific ticket, a button in the ticket can be clicked (for example, "request browser details"). 4. In the ticket an individual checker URL is generated and included. 5. When the end-user opens the URL, a report is generated and saved in database and the callback URL is requested. 6. Upon callback, the app checks for new reports via API. If a new report is available for the specific ticket, it is loaded and included in t...

I am looking for a python programmer to ...documentation. Other than the example provided in the code, I'm not sure how other Classes should even be used. The assertion tests will (in part) provide an example of how each Class and Method are used. I would like assertion test for the module in You can find sample .ini files here: My primary interest is how to use append, insert, pop, and remove in the Category Class. I assume the module is capable of modifying sections and key/value pairs. Using coverage and pytest, your objective is 100% coverage, pass or fail for the module.