Asterisk sip tls srtplavori

need some cloud voip serveurs for swithing calls(fusionpbx,freeswith,elastix)

Module 07: SEED Lab VPN configuration using IPsec (100 points) Complete the SEED lab found below. Upon successful completion of the lab, you will submit screenshots (pasted into a Microsoft Word document) and then submit to Blackbo...document) and then submit to Blackboard as proof of lab completion. The learning objective of this lab is for students to master the network and security technologies underlying SSL VPNs. The design and implementation of TLS/SSL VPNs exemplify a number of security principles and technologies, including crypto, integrity, authentication, key management, key exchange, and Public-Key Infrastructure (PKI). To achieve this goal, students will implement miniVPN, a simple TLS/SSL VPN, in the Linux operating system.

We are using FTP over SSL/TLS. The data connection, which uses a random port, is failing. The data port in FTPs passive mode is a random port that the server assigns for each individual connection and I need help in setting up the FTP such a way where it sends traffic to the FTP server. I shall award the task if you can demonstrate you can solve the problem after I share remote access credentials.

We have a web app based on Laravel 4 and would like to update it to version 8. Maybe some bug fixing is required too. There are several custom controller, views and models which has been individually developed. It would also be good if you are familiar with Asterisk / VOIP technologies, which is part of the App.

Hello , We are facing https issue on our nimbra mail server , error message :- TLS Negotiation failed, the certificate doesn't match the host. same has to be fixed via anydesk only.

Situation We are aiming to dial into a Webex meeting from a SIP device. We have a self-hosted Asterisk server that is connected to our video conference application We are registering this Asterisk SIP account with our video conference application and Dialing into Webex cloud meeting and various telepresence hardware such as CISCO Telepresence, Polycom etc. Complication 1. We are able to successfully connect to telepresence hardware with no problem whatsoever. - PASS 2. We are NOT able to connect to webex cloud meetings (both personal account and organization account). ​We are getting we are getting this 408 request timeout error. - FAIL Screenshots attached for reference Requirement 1. We require you to work with us and help troubleshoot this issue.

I need a Residential floor plan with wall dimensions and elevations. 4 Bedroom, 3.5 Bath, dedicated Office, and a Flex / Bonus Space but no formal dinning. 2 - Story home compatible with SIP systems design installation. 1st Floor: Kitchen, Living room, Master, Master Bath, 1 Bedroom, Office, Utility, 1.5 Bath 2nd Floor: 2 Bedrooms, 1 Bath, Flex / Bonus Space

...Raspberry Pi and have installed (initially) the Asterisk + FreePBX Per this documentation: It got to the point where I was installing the security packages from the command line and it asked if I wanted to overwrite a certain Python file and realized that if I do and it breaks that I need to start over so it'd just be much quicker to work with YOU! So that I can see how to do this properly. Shouldn't be long and we can actually just do it from the web browser (GUI) - in fact we'll have to in order to screen share. I need the RasPBX - Asterisk Dialer to: - Auto-Dialer - Power Dialer - Voicemail Drop (doesn't ring their line, just leaves a voicemail) All of this is actually already baked into Asterisk+FreePBX so it's really just a matter o...

...instalado en un servidor el Issabel Asterisk para identificar las llamadas, y tenemos un portal web creado por PHP para que muestre toda la información recopilada. El funcionamiento actual de Asterisk y la web es la siguiente: En el portal web se de de alta un nº de teléfono, desde ese nº de teléfono se llama a un teléfono definido (siempre es el mismo, ya esta configurado en asterisk) y una vez recibe la llamada, la cuelga, mira en la lista de teléfonos dados de alta, lo localiza e indica la hora de la llamada entrante en el portal web. Ahora queremos ampliar la función con lo siguiente: En el portal web añadir un tick que se llame "Control" y lo podamos seleccionar para los teléfonos q...

...branding for the chat side of the app. We are now adding audio/video calls including one-to-one and conference calling. As a good reference, you can refer to how WhatsApp and Telegram make one-to-one and group audio/video calls. The one primary thing to note is that we do not use mobile phone numbers for making phone calls so there will not be a dial keypad. We use sip addresses to make calls and each user will be assigned a sip address when they register into the app. The audio/video UI/UX wireframe will include: 1. How to initiate an audio/video phone call. 2. Show one-on-one phone calls in action and be able to switch between audio and video for both the caller and receiver. 3. Include a way to add additional individuals to start a conference ...

I need someone who knows about asterisk PBX for packet related issues on Asterisk

After restart Zimbra 8.8.10.GA.3039 services are not starting. zmzontrol restart gives ------------------------------------- Connect: Unable to determine enabled services from ldap. Unable to determine enabled services. Cache is out of date or doesn't exist. ------------------------------------- after ldap start, starting the zmcontrol giving: ------------------------------------- Unable to start TLS: SSL connect attempt failed error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed when connecting to ldap master. ------------------------------------- It must be something w certs, cauce we have different backups and they are not starting also.

Hi Folks, We have a standalone asterisk server recently installed on a Ubuntu box. We want to achieve the following in terms of network connection 1. Ethernet interface - This connects to telecom provider for asterisk line and should be used for everything related to SIP 2. Wireless - We want to use this for connecting to the internet Need help from a networking/asterisk developer who can help us split the traffic in the above-mentioned fashion.

Hello, I am looking for a developer who can create a script to sniff phone numbers (sip/pjsip over tls). In order to avoid reading the CDR regularly, I would like a webhook that calls a url. I need date, time, calling phone number, called phone number, transaction id, call duration, status: response codes The documentation for the installation of the script. Environment : Server : PBXware (tls / ssl, pjsip/sip) Smartphone : Communicator / Glocom Linux server (sniffer) : Debian Language: python Library: scapy or other. Transport: tcp / tls

Want to able to transfer a call to another agent/queue using a prefix so we can record a note that is then played to the new agent when they pick up. FreePBX 15 Asterisk 16.13 PHP 5.6.40

Thanks for reading, we have a problem to solve as below: 1. We are running Asterisk IP PBX soiftware on MT7628 Cpu running OpenWRT 2. For audio we are using PulseAudio and we have some problems: 2a - The echo cancel algorithm for the on-board speakerphone gives echo to the distant end, i.e. the echo cancel is not fantastic and needs looking at 2b - We are struggling to play recorded audio over our speaker 2c - The PulseAudio always requires a second reboot to work properly. If you are an engineer with solid experience in this area we would appreciate assistance on solving these items.

...connected a "feature type" phone using its native dialer to our VoIP (SIP to SIP) Platform (rather than an app on a smartphone). The phone is Android OS and does have a SIP client. The phone has a messenger app for sms and the manufacturer has noted we could make adjustments to the apk as this is publicly available for developers. On our VoIP side the custom application should support the SIP SIMPLE protocol for messaging. It should send outbound SIP MESSAGE to our SIP node IP as well as receive inbound SIP MESSAGE from the SIP node. This maybe helpful … https://cs.android.com/android/platform/superproject/+/android-8.1.0_r48: and dive into the phone and messages source code. The SIP stack is in the ext...

Создать Asterisk PBX в виде приложения для Android по сути что бы PBX, работала, в фоновом режиме на устройствах Android, можно было это приложение скачать с google play, и подключиться к Asterisk удаленно (например по SSH) для настройки конфигурации 1) подключение sip клиентом на том же устройстве по адресу 2) аптайм приложения 99.99% 3) работа всех модулей Asterisk 4) возможность подключение как клиента так и siptrunk к Asterisk

Hello, I am running a Small call center with Freepbx 14, needs to develop a customer satisfaction survey application for incoming queue calls. Call landed to the queue > agent answer > agent completed the call > should go to satisfaction survey automatically or agent can transfer the caller to satisfaction survey. Custom IVR should be able to play to the caller, should have a dashboard to view the reports. Reports needs to be contain following Date and Time / Number / Agent's Extension / Score needs to be able to run reports monthly / weekly / daily and custom date range.

We would like someone to organize various events at our retail location on weekends.

Hi Josh, I'm a VoIP communications engineer and developer that can add a new SIP Trunk and update existing SMS API to a new API RESTFul endpoint as you want.I have 3 locations listed in my file. 2 are meant to load balance eachother. At the moment it works perfectly as the logs show the calls doing a round robin between the 2 locations. We use 2 sip providers. 1 gives us 2 trunks to load balance and the 2nd we use just for failover in case the first provider goes down or is unable to route a call for any reason. Here is my file: The config file I am using was created by someone else for a different system we use to use. # $Id: ,v 1.1 2004/08/10 16:51:36 dcm Exp $ # sample config file for dispatcher module debug=2 # debug level (cmd line:

Hi Josh V., Hi Josh,.... I am an expert VoIP developer. I will add a new SIP Trunk and update existing SMS API to a new API RESTFul endpoint.I have solution for your problem. public virtual HttpResponseMessage Get([FromUri]TwilioRequest request) { TwilioResponse tr = new TwilioResponse(); if ( == "Hello World") ("Hello back!"); else ("Text 'Hello World' for a friendly message."); return (, ); } SMS API enables you to send and receive text messages to and from users worldwide, using our REST APIs. Programmatically send and receive high volumes of SMS globally. Send SMS with low latency and high delivery rates. Receive SMS using local numbers. Scale your applications with familiar web technologies. Pay only for

Within our existing web...do some minor improvements. We aim to establish a recurrent relationship with a Front end developer that deals with it. Technology stack -Frontend React (hooks) Typescript Redux Cookie, Sessions Some experience with UI design would be a plus -Backend (as information, not required) Go Python -Security Understanding of basic security concept and how to use them in a frontend app ( Access token, TLS ) Understanding of classic security concepts related to frontend applications (CORS,CSRF,...) -DevOps Proficient with Git Familiar with concepts of Pull Requests/code review Familiar with multi-environment setup (local/dev/qa/prod) Basic understanding of CI/CD -Cloud Basic understanding of Kubernetes would be a plus (not required) Some experience with a Clou...

Hi, can someone do the following: * Remove Logo top left of each page and replace with logo on the business card that blue loop with business name - Resource Appraisals * Change the top of the color of the bar to match...Replace fax line (800) 621-7070 with (866) 906-9326 * Edit the copyright information at the bottom of each page from 2010 to 2002 * Remove the line after All rights reserved with Resource Appraisals and the logo are trademarks of Resource Tech Appraisers Then make the document into a fillable form for Mac / PC, Internet. It will be sent to people to fill it all out and submit it to send back. Red Asterisk are required fields - All fields should support Alphanumeric characters. Under Additional equipment and Notes should support max characters for a long message...

Hell...a "feature type" phone using its native dialer to our VoIP (SIP to SIP) Platform (rather than an app on a smartphone). The phone is Android OS and does have a SIP client. The phone has a messenger app for sms and the manufacturer has noted we could make adjustments to the apk as this is publicly available for developers. On our VoIP side the custom application should support the SIP SIMPLE protocol for messaging. It should send outbound SIP MESSAGE to our SIP node IP as well as receive inbound SIP MESSAGE from the SIP node. This maybe helpful … https://cs.android.com/android/platform/superproject/+/android-8.1.0_r48: and dive into the phone and messages source code. The SIP stack is in the exter...

Istzustand: 3CX Telefonanlage auf einen firmeneigenen Server. (läuft bereits) ca. 5 Snom Telefone (IP per DHCP jedoch statisch fixiert), teilweise über VPN von extern angebunden. SIP-Trunk von Sipgate. Ich möchte gerne eine einzelne HTML Seite programmieren auf der der Status von ca. 5-10 SNOM Telefonen angezeigt wird. Unter Status meine ich den Zustand (Frei, Besetzt, DND, ...) Vewendet werden folgende Telefone: SNOM 760 und SNOM D735 Diese HTML Seite sollte auf einen eigenen vorhandenen Linux-Apache-Server laufen. Die Seite sollte von allen Mitarbeitern im internen Netzwerk zugänglich sein. Zugriffsgegelung ist ncht notwendig. Jeder im internen Netzwerk darf auf diese Seite zugreifen. Auch möchte ich gerne, dass der Zustand DND auf dieser Seite für...

Hi, I have an issue with Call Deflection when a mobile calls into a DDI which is then forwarded back onto another mobile. This is getting rejected when Call Deflection is turned on. Works when off but I need it on to show the A party calling number. Need some to trace and fix my SIP Trunk parameters / settings. Using Freeswitch with the same SIP Trunk it works so it is supported by the carrier. I just don't know the 3CX system enough to debug and fix.

I am looking for someone to quickly put together a recording demo using either Asterisk or Freeswitch where the user is able to do the following: 1. Login into a basic site (will provide access to a DO server) 2. Have the option to start/stop a recording (simple button should be present for the user to click) 3. Once the recording is stopped it should be saved. 4. Ability to view and playback recordings with timestamps. We do not require any options to delete or edit recordings. 5. The user will login into the site on their PC and using a USB microphone will conduct the recording.

We are currently using a custom PBX solution. The PBX server is hosted in the Vultr Cloud and is based on Freeswitch with the frontend as FusionPBX. There are few issues which have come up which needs to be fixed urgently. The SIP trunk is being provided by Voxbeam 1) All incoming calls should ring the SIP softphones 2) Outgoing is not working properly for some regions. Need to diagnose that and resolve 3) If possible, need to configure an alternative SIP trunk in case Voxbeam is failing. The process should be automatic.

We are a telecom, marketing and analytics company, we’re looking for a PHP/Laravel full-stack web developer with experience in VoIP/SIP & Twilio or similar Telecom company and having knowledge of telecom and how telecom systems work especially FreeSWITCH & OpenSIPS. You must have the following skills to qualify for the job: PHP Laravel  Javascript/JQuery SSH + Ubuntu Command line MySQL GIT - GitHub Twilio Telecom knowledge of VoIP, SMS /MMS, Call center style call queues, call conference rooms FreeSWITCH OpenSIPs AWS React These skills are optional: Facebook APIs We will offer long term work every week but only if you prove your skills in the first week or two. We are only looking for serious developers and to prove that please fill the attached document, u...

I need to create a B2BUA but for SIP to Microsoft Teams. I know direct routing will do most of what I need but Video is my goal. I need to be able to take SIP Video and send it to Teams Video phones. Early video would also be required. Willing to license whatever I need from Microsoft. PLEASE DONT BID IF YOU HAVE NEVER DONE THIS TYPE OF WORK BEFORE.

...integrated as required: Admin Area: Create, Suspend / Un-suspend, & terminate PortaOne account. Send Welcome Email with SIP account details. See and adjust client credit balance. Create products with SIP configuration options (such as PortaOne client call rate card, account type, DID monthly/annual charges, subscriptions, and credit limits) Create products for fixed subscriptions / Add credits Allow enable/disable of international dialing and call forwarding. Client Area: Show PortaOne Credit Balance In real-time Show SIP Account Details Topup dropdown with fixed amounts for easy use and fast checkout, top-up must create invoice and allow WHMCS payment gateways. Change SIP Secret / password Call Rates (call plan) Call History - XDR (Includes search, Do...

...as required: Admin Area: Create, Suspend / Un-suspend, & terminate PortaOne account. Send Welcome Email with SIP account details. See and adjust client credit balance. Create products with SIP configuration options (such as PortaOne client call rate card, account type, DID monthly/annual charges, subscriptions, and credit limits) Create products for fixed subscriptions / Add credits Allow enable/disable of international dialing and call forwarding. Client Area: Show PortaOne Credit Balance In real-time Show SIP Account Details Topup dropdown with fixed amounts for easy use and fast checkout, top-up must create invoice and allow WHMCS payment gateways. Change SIP Secret / password Call Rates (call plan) Call History - XDR (Includes searc...

I am looking for a script from where sip trunk gets connected and make calls through different numbers to one number. Its basically known as TDos. Any one who know about this work and can do it , PM

OPEN TO DIFFERENT BUDGETS; AND FIXED I have CPANEL installed for many years but I've never setup the IMAP/POP3 Secure SSL/TLS Settings For email I always use Non SSL. But I am setting up Zendesk Sell and they require IMAP so I need help to set that up. 006: The hostname on your SSL/TLS certificate does not match your email server's hostname If you can please help me set this up on my server correctly and once Zendesk Sell recognises it successful that would be very appreciated? Thank you!

Hello, I need to develop an AGI script to integrate a dynamic IVR with google's STT. Asterisk versions: 13, 16 and 18.

I need help to configure and

С помощью криптографической библиотеки OpenSSL 3.0 реализовать клиент-серверное оконное приложение с модернизированным модулем передачи данных по защищенному протоколу TLS 1.3, которое реализует следующее: Для TLS предусмотреть использование подписей RSA (RSA-PSS-RSAE-SHA256), ECDSA (ECDSA-SECP256r1-SHA256) (доступно по умолчанию), а также реализовать алгоритм подписи ДСТУ 4145:2002 ( - Додаток Г.1, №7, кривая в полиноминальном базисе 257 бит, возможное название для алгоритма: ECDSA-DSTU4145-SHA256), насколько знаю подпись используется всообщении Certificate Verify () во время хендшейка. Подробное описание указано в документе ТЗ.

I have gitlab self managed, and i want to enable the container registry. but i face a TLS handshake error. now every change i make to enable gitlab it stops it from runing

Hi, we are developing mobile application for SIP calling in React Native and SIP js and desktop application through Electron JS. We need help in following parts. 1. To setup proxy server to send notification when call is incoming for mobile app to wake up and register. But the call to be connected to main SIP server instead through the proxy server. 2. To implement the call conference in both mobile and desktop applications.

We got an issue with our PBX. Our SIP trunk provider indicates that our router blocks some traffic from his IP. We need to fix this. The router works well for many months without any issues. hardware: Mikrotik V6.48 RB3011UiAS PBX grandstream UCM6204 , up to date I join a pcap file with the issue. See example at line 15912, 15917 and many other lines. The PBX work well during about 18 minutes after this the issue happen and we need to reboot the PBX.

I want a Softphone WEB with integration of Janus Gateway + Api Gateway. Request: Contacts - Add, Delete, Update, Click call, Click send Message, Alert new Mensage, Resume. Call - Make Call, Mute, Transfer, On Hold, Conference, Show Participants Call, Dial Pad - Use Gateway Janus Sip Legacy. Enable video call using Janus Video. Chat - Send Message, Received message, List menssage - Integration Backend Services - API Gateway The application need to use Ionic 3+ and Angular 6+.

This project is the installation and configuration of fusion PBX on Debian 10. You will be provided access to a VM. What must be delivered Full functional system integrated to 1 sip trunk, VoIP phone and softphones A fully secured system that is only accessible to admin and users registered in PBX Full documentation of how to in Freepbx eg: add new PBX user, configure softphone, setting to be used for softphone, VoIP phone config and settings Auto-attendant with MOH for inbound Leave a VM by pressing a number VM must be sent as an email to admin and PBX registered users. New users can only be added by the admin Outbound calls are restricted to users registered in the PBX. Call recording when a call is answered. You must have experience of doing so as well. The development must ...

I have an old Visual Basic Code. A server I was connecting to changed protocols and now I have problems connecting with the TLS 1.2 Just cannot reach Web Service. Attached Is y my code Need to correct problem, compile and send me DLL.

Hello, Me and my friend we have created a SIP video intercom, we would like to start the production and sale of devices, but to get customers we need free applications and a server that will allow easy configuration. We are currently using the Russian application, but it is difficult to configure due to the need for port forwarding in the router. Not every client has the option of port forwarding, so we want to implement a free server in our database + free applications for our clients. The Linhome system is the perfect solution for this task, we have been testing for some time and it works stably with our intercom. I hope our requirements are not too high: A.) APP. We need versions for iOS, Android and Huawei are needed, and in them: 1. Adding our logo 2. Change of the graphic des...

Quiero configurar mi centralita con las siguientes características. mailbox y configurar la cuenta de mail. 2. horarios de la centralita. 3. Solucionar problemas con el asterisk que no muestra debug en el cli 4. No puedo conectar las extensiones en los teléfonos ip. 5. Configurar gateway Dinstar de 8 Puertos y configurar las llamadas de cada uno de los puertos a una extension específica. 6. Actualizar asterisk de la version 13 a la 16

Hello, My freebpx is not connecting to asterisk and i need a expert asap to help me with it, will be via anydesk,

Input sip provider setting in vicidial

We are looking for someone who can setup an asterisk server for us. It must be setup with correct security. Furthermore, you must show how we setup new agents and DID numbers with queues and opening hours. We do not need an interface; it can be directly in the conf files. A ”phone” must also be made, like this image, where it is possible to Answer calls Transfer calls – both hard transfer and soft transfer (speak first) Put the agent on DND Pause the call Make outgoing calls Furthermore. the phone must be built into 2 different system, it is not something you must do, but you must ensure that it is possible, either via an API or similar. It is naturally important that if an agent is assigned to a queue and a if a call comes into this queue, then it must ALWAY...

Hi Sadiq, I really need to connect to AWS, I've received the firmware that allows tls 1.2 to work on the sim868e which is required by AWS, so I think I'm very close to getting it done but I'm stuck with the code.. can we try get this done somehow? Thanks in advance. Regards, Emile