Asterisk sip tls srtpJobb

make a 3D design of this image with a 6mm bail to be used a pendant for parts unknown it’s up to the artist the idea is a trippy piece to create pendants the grass near the mushrooms is not needed. The height of the design should be 6cm not including the bail.

I wanted to build omni channel solution on Asterisk open source platform. It should have following module like Voice, Email, Chat, Messaging, Reporting, Voice Analytics and GUI based IVR. It is good to have WebRTC. I am also open if anyone has already developed like this.

Create a simple to use howto in both...user on your own Windows 10 machine, run (which uses ) and fetch TLS keys from and decrypt your own corresponding TLS web traffic in wireshark. To be able to complete the project you need to install on your own machine on windows 10: and probably use the tools And any other free tools you seem appropriate. Here are some ideas how others have done it (i.e. decrypt TLS traffic of windows apps that uses ): Always stay within what is legal.

Wanting a custom Asterisk API solution that combines whatever you think is necessary (ARI, AMI, AGI/EAGI, JACK, ICES, etc, doesn't matter to me) into one API solution that provides the following functionality: I have a remote website staff portal that has the functionality to initiate calls to our customers and play sound files to them (usually notices), and another feature that allows us to call up to 6 customer phone numbers and place them into a confbridge together with one of us, ringing all 6 simultaneously, to have a phone based conference call. I want to enhance that functionality and improve it especially by adding the ability to stream the calls live to the web portal (not sure how: EAGI, JACK, ICES, ARI externalMedia, Custom C++ module, I don't care) which we d...

Hair salon logo. crisp clean. shimmery gold or rose gold maybe. I would like to see the logo "Salon Rose" (e with an asterisk) and one with "Salon Rose"with smaller print under that says "and extension bar".

We need help with the Android Linhome app, namely my developer has a problem with PUSH afterimages when the app is running in the background or the app is in close mode PUSH notifications are not working. We are looking for a person who can fix it.

MBA SIP ON TOPIC study of different marketing channels & their impact on e-commerce.

I have a php website that has been analysed and has vulnerability issues in terms of old code and security protocols These were the findings The following are the findings that are recommended to be addressed. • Insecure Transportation Security Protocol Supported (TLS 1.0) • Weak Ciphers • Old Software – Adobe Flash • Stored Cross Site Scripting (XSS) - Public Disclosure • Information Disclosure • Missing Security Headers • Missing SPF Record I need someone urgently to tell me what needs to be done, how long it will take and cost etc

Hi, - should design a chart which is on homepage, also develop client profiling tab in the same site. please check the site and then only chat

Hi Eremin P., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

Hi Vinod Kumar I., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

hi , hope all is well. i am looking for someone to help diagnose the connection between my asterisk and sip provider

Need help building PBXact and custom Modules. Should know how to review PBX Status on Asterisk Server and more. We are looking for somone with past experience with Asterisk.

Looking for someone to setup kamailio / OpenSIPs for the following - Registration pass-through SBC for Freeswitch Servers and Remote Phones... kamailio / OpenSIPs will be hosted inside Amazon EC2 - SBC Setup for phone calls with Least Cost Routing... Freeswitch will use kamai...kamailio / OpenSIPs for the following - Registration pass-through SBC for Freeswitch Servers and Remote Phones... kamailio / OpenSIPs will be hosted inside Amazon EC2 - SBC Setup for phone calls with Least Cost Routing... Freeswitch will use kamailio / OpenSIPs as the gateway and kamailio / OpenSIPs will pass the call to the appropriate ITSP - Setup kamailio / OpenSIPs with local registrations for SIP Clients to handle Video from Door Phones for Remote Clients

We're looking for an engineer with experience writing, testing and shipping production-quality code in Go, with a background in Ethereum/EVM-compatible development. As a member of an agile engineering team, you will help design, implement, test and deploy new features in very short cycles . Objectives of this Role - Design and implement services that are scalable, and fault-toleran...Knowledge of distributed systems design - Creative, independent, and hardworking - Problem-solving attitude - Good communication skills in English (both oral & written communication) Bonus Qualifications - Experience building wallets, payment tools and similar crypto products - Experience running ETH nodes - Experience modifying node codebase - Experience with RPC protocols - Experie...

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Required Functions (server, port, protocol, username, password) → Boolean Task: • Register at SIP Server • Return true on success • Keep connection alive, reconnect on disconnect • Must support SIP Server, UDP, TCP and TLS connections () → Boolean Task: • Return true if connected to server (to) → Boolean Task: • Initiate a call to a given SIP account • Return true if call is answered (files) → Boolean Task: • Play a list of mp3 files on the SIP call [given as Python list] • Return true if all files are played (char) → Boolean Task: • Send a DTMF sound • Return true if DTMF sound is played () → void Task: • Ends the current call () → Boolean Task: • S...

we need configuration of freepbx virtual machine/asterisk we have about 3 phones to configure there

SIP trunk which is located in our Office needs to be connected to IP-PBX (fusion - pbx) which is hosted in Google Cloud Platform through VPN. Also Some parameters have to be passed through X-Headers. Some one who worked previously on scaling IP-PBX preffered

Hi Ayush G., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

3cx configuration sip trunk with cisco

...com/spce/). Someone who have enough experience about VOIP, SIP, PBX. You sould be dedicated, serious and working transparent and clean. The sipewise will have to be configured for the moment with one phone provider with one "big" SIP Trunk and multiple channels. In the moment some 30 phone numbers. On sipewise I will generate different SIP Trunk based on the request of the client with one phone number and 2 or 3 or 8 channels. This SIP Trunk "generated" and managed by sipewise will be connected in 90% to an other PBX which the client have or in our data center or in his infrastructure. To secure the SIP Trunk we should think how will be the best solution or to offer to the client based on his possibility up to 3 ways how to connect ...

I am having a hard time integrating asterisk with Odoo. I have set up asterisk in the same server as Odoo, also have installed certificates for webrtc as well as for the odoo portal. I will provide access to my sandbox server.

Small multi platform script that: - Registers with a SIP server - Initiates a call - Plays MP3 files - Does DTMF - Ends Call Must work on Windows and Linux "out of the box". All components and source files must be included.

Write a Python script that sends text messages to phones with sip authentication It only authenticates the ip, user and password

Helllo, we are manufacturers of milk and milk products and soon would be launching our own brand for products like- Milk , Paneer, Curd and Desi Ghee. The name which we have decided is White Sip. Need branding and logo assistance

Precisdo de uma plaforma tipo MIRTAPBX ou o prórpio MIRTA PBX, que seja multi tenant e possa configurar contas voip e diversas configurações de central virtual via painel de controle e dashboard (preferencialmente em Português). Sei que o mirtapbx tem uma vesão em português, porém estou sem retorno (contato) com a pessoa do mirtapbx.com.

Preciso de um profissional para instalar o sistema CDR-Stats para asterisk.

Looking for Call Generate system for Looping , System Generate calls and it go to supplier and from supplier that call come back to our server if same call come back to our system then only call hold me required duration else disconnect immediately also manipulate ASR and ACD as per our requirement

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: