Asterisk sip tls srtpJobb
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.
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,
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
Hello, we already have asterisk and freepbx installed and we need someone to help us configuring extensions and test calling.
I already have a ready-made programme developed, I want a person to Installation of this programme into a fresh server, Few changes and also a long term maintenance contract.
I need someone to develop a custom Asterisk - FreePBX module/application, this must have; - I want to make a keypad recording and audio player on call, for example (I call someone and want to play an audio stored on database and if user inputs #123 on keypad I will get that output)
We have an esp32 connected to a SIM868E GSM / GPRS & GPS module through UART. We need someone to write C/C++ code to connect to AWS IoT core. We got it working on cloudmqtt and other public brokers, but the TLS encryption is difficult with aws. It would be best if you have a SIM868E breakout board to write the software and test. We use esp32 as the main MCU so it will help if you use that as well. We will provide tls certificates and aws iot core connection details. It can be written using esp-idf tool set or using Arduino IDE with any MCU. Please ask if you need more details.
Our self hosted exchange 2013 server was setup using a domain we no longer want to use. We have since changed our domain and emails are sending/receiving fine using this new domain but our metadata still reflects the old domain. We need someone to make the necessary changes so the following can be updated. Received: from (X.X.X.X) by SERVERNAME.local.olddo...setup using a domain we no longer want to use. We have since changed our domain and emails are sending/receiving fine using this new domain but our metadata still reflects the old domain. We need someone to make the necessary changes so the following can be updated. Received: from (X.X.X.X) by (X.X.X.X) with Microsoft SMTP Server (TLS)
Hello we are looking ONLY for experienced architect in wood Houses with SIP panels We are building tiny house with SIP panels We did a sample of tiny house already but now we need to get done some drawings by an arcchitect used to build houses with SIP panels The Tiny house is 7.2m long and 2.4m wide, and 4.2m high I will ask you to show me plans you already did prior giving you the job Once done I'll have same job to be done with a garden studio Thank you
I need to build a software which supports my clients and connects them with my employees, I am looking for people having strong skills in Python Node.js My SQL or MS SQL building a CRM based web application The software Includes API Ping post, Multiple user access, Voip/SIP call, Subtitles display and more I need a Person/Freelancer who resides in Chennai, Tamilnadu India.
...and everything else related to keeping the business running. A few months ago I purchased a GoIP4 SIP GSM Gateway, but been having a hard time configuring it, so it hasn't worked as expected up to now. I've got 4 local sim cards in the machine already but I never got them to work properly. I directed the calls to an SIP address on one of my virtual PBX, I want those sim cards to receive and make outgoing calls and I want all 4 people to be able to do so right from their stations at home even though we are not in the same network. At this point, I can receive calls but calls always get disconnected only after 10 to 15 seconds for unknown reasons. When I make calls, the calls are charged to my SIP provider instead of being charged to the SIM card, so I�...
Hello We are building tiny house with SIP panels We did a sample of tiny house already but now we need to get done some drawings by an arcchitect used to build houses with SIP panels The Tiny house is 7.2m long and 2.4m wide, and 4.2m high I will ask you to show me plans you already did prior giving you the job Once done I'll have same job to be done with a garden studio Thank you
Требуется разработчик с глубоким пониманием SIP, VOIP уведомления для Apple в приложении Flutter. Задача следующая: есть связка ip домофоны - сервер Asterix - мобильное приложение для ios, android. Уперлись в следующую ситуацию: не получается разбудить приложение на ios.
Dotnet core web application publish to smarter asp failed error message Severity Code Description Project File Line Suppression State Error Web deployment task failed. (Connected to the remote computer ("") using the specified process ("Web Management Service"), but co...computer ("") using the specified process ("Web Management Service"), but could not verify the server’s certificate. If you trust the server, connect again and allow untrusted certificates. Learn more at: The underlying connection was closed: Could not establish trust relationship for the SSL/TLS secure channel. The remote certificate is invalid according to the validation procedure. Tarqeem 0
I am looking for someone who has experience working with VoIP and a number of different PBX systems. The PBX will most likely be installed on Amazon AWS unless you have a good reason to use a different host. ========================== Wishlist for the PBX is: - SIP based - Free or low cost licensing for the whole thing (PBX and modules - Free is better!) - Ideally on some kind of linux os (Most comfortable with CentOS but other distro like Debian are ok as well. Prefer to avoid Ubuntu - I seem to have to many problems with it) - The PBX should either be multi-tenant or you should have a solution for how we can have different users on the same server but keep them separate. - Ideally a nice, easy to use web GUI is ideal, especially if we can have a separate login for the user...
I am looking for someone who can configure and provide the PBX.
Hello, I need a voip expert to setup asterisk with didwww need it urgently now,
hi i have exchange server 2019 firewall sophos i cant receive and send from The response was: TLS Negotiation failed: FAILED_PRECONDITION: starttls error (0): protocol error
Hello Viktor, we haven't talked for a while. I just show a project where you integrated Asterisk + MS Teams. We've got a client who wants to use MS Teams with a trunk from our Asterisk. We're not sure if it is worth the effort. So, can you please let us know the price for that integration. It would be enought with a step-by-step guide and support from your side. Thank you so much Ricardo
I need codes which can login Nike website by using Requests only. It can done by Python or Javascript. You may require Golang to deal with TLS check too.
Hi, we have new installation of freepbx and asterisk. We created extensions and used sip app to connect. One phone can call another phone, but both parties cannot hear each other. Error is "Couldn't negotiate stream 0:audio-0:audio:sendrecv"
we already have a running and functioning Asterisk system installed. we just need someone to look at the code and find the reason why some calls are getting dropped (2-3 %). and some other simple issues.
...Access Control -Active shooter detection -Nurse Call -AI enabled cameras for weapon recognition -Industrial Automation -License Plate Recognition -IP based Intercom and Announcement Systems with SIP Phone Integration -Panic Buttons -Multi-facility systems unification -Area of Refuge Solutions -Unified Security and Operational Intelligence Education -Intrusion Detection -Fire Detection -Video Monitoring/CCTV -Security Access Control -Active shooter detection -AI enabled cameras for weapon recognition -Industrial Automation -License Plate Recognition -IP based Intercom and Announcement Systems with SIP Phone Integration -Panic Buttons -Multi-facility systems unification -Area of Refuge Solutions -Unified Security and Operational Intelligence Banking -Intrusion Detection -F...
Need somebody who can configure a Cisco C60 Codec as Standalone device (No Video Communication server or similar) in order to join Webex meetings via SIP call. A free sip registrar like should be used, alternatives are appriciated. If possible there should be no reucuring cost in the solution (e.g. Hosted SIP registrar/H323 Gatekeeper like pexip) As deilvery there is either a complet configuration document expected or and lodable config file). A network configuration suggestion for the firewall should be proposed as well.
Hi I need someone to modify python's "requests" module to change the TLS connection method by implementing a valid JA3 according to the given session headers.
