Asterisk sip authentifikationlavori

Seeking a the assistance of an individual who could assist with the learning curve with Git. I have built up a substantial repository of PowerShell scripts and automations a...PowerShell scripts and automations and attempting to generalize these for a diverse client base. Manually controlling versions is problematic at best. The successful applicant will have skills in Git, and integration with Visual Studio Code and PowerShell scripts. I am not seeking development assistance in PowerShell. I would suggest three one-hour sessions would be appropriate, either via telephone, SIP, or a method of the applicant's choice. I also have a set of bash scripts that may be worth migrating to a repository. Target is $50 USD for the project, in three pieces, plus a bonus upon successful know...

Building Video Conferencing software for schools and educational institutions need somebody to refine and modify code of open source video conferencing software and help set up the website. Need somebody who is familiar with WebRTC and SIP and other communication protocols. I just need someone to add login screen and also add a nice homepage and other minor adjustments to my website

We need an experience SIP Dialer developer to modify an existing IOS and Android SIP Dialer.

Need to set up SIP or Voip on my phone I have a number from my crm Service Fusion, I have voip and sip credentials I have not been yet successful with any app (Mizudroid, toktiv)to be able to send and receive sms and images through this number. in the desktop crm this number is able to deal with images, but not on phone. Service fusion gives me a dashboard for sending/receving calls and mms, but i nee this t happen on my phone through an app. Can you do this?

Currently we have a project to interconnect SIP Trunk from telco provider with Yeastar PABX throug Mikrotik. Up until now it still does not work and we need netw. engineer to solve this issue.

I have an Asterisk system running in the cloud - specifically an IncrediblePBX system (see nerdvittles<dot>com) running on a Crown Cloud server. I am an Asterisk noob and would like some help getting the cloud software configured to connect to my extension at home. The software is is installed and running on a commercial (Crown) cloud environment. I need to: 1. Help me access and set-up PIAF with my one extension in my office. I don't have any port blocking issues to my knowledge for port 5060. 2. I would like to set-up an extension for a couple of cell phones and a SIP adapter for one wireless home phone. 3. I would like some help getting the basic IVR solution running (press 3 for sales etc) There is no big hurry on the project as it is my personal syst...

We need a module developed for freepbx that would let us integrate it with zendesk so that it provides functions similar to: /apps/five9-for-zendesk/ /apps/asterisk-dtk-by-cdc/ The module should meet the following minimum criteria: - Be installable through the freepbx module admin - Must be installable in zendesk as an app - Must be able to assign zendesk agents to extensions - When an incoming call is coming in a popup should appear in Zendesk containing the following 2 scenarios: 1) if the customer is in zendesk the customer name should appear with an open profile link and reject call link, . 2) If the customer is not in zendesk the phone number should appear with a create contact link. - If the call is

Приветствую. Я живу в многоквартирном доме на 150 квартир. У нас вызывные панели BAS-IP с поддержкой SIP. Работает это все через малинку с True-ip, но он очень плохо пропушивает звонки на телефоны (в локалке проблем нет) Немного полистав инет, пришел к выводу что связка Flexisip+Linhome стало бы решением для жителей. У меня есть сервер на площадке ISP. Если Вы опишите параметры, то с предоставлением доступа к виртуалке проблем не будет. Подскажите, на каких условиях мы можем это сделать?

My name is Alex. I live in Kyiv/Ukraine. We have residential building for 250 apartments. Trying to make an intercom system for neighbours with entering panels that support SIP. Searching on the web I've discovered that the Linhome app is most suitable for this purpose. I think that Flexisip server will be correct suite to make it work. Is it possible to ask you to setup flexisip server for me with simply preferences (push enable) and teach me how to add/edit users on it?

I have an installation of Issabel running on a windows virtual machine, connected to a sim gateway. I need an option to import a csv to precisely control call flow The data structure will be like this: Call ID (unique reference number eg 13664.34244), Sim-reference (eg 1003), Start (eg '2021-26-04:22.21:00), End (eg 2021-25-04:22.25:00), recordingID(eg ""). I do not have sample data, since this is the structure. Sometimes more than one call will be made at the same time, the system needs to be sufficiently robust to handle many simultaneous calls. We would also need to be able to upload audio WAV files so that a recorded message could be played during the call as specified in the CSV file. The whole system would need to be uploaded to ISSABEL by remote machine over a ...

I need some help with my business. IVR SYSTEMS using freePBX and SIP

i need a sip calling app that based on the linphone customization, and also with the source code, if some one can do it, just contact me.

I'm in need of someone who can install & configure Linux Software for Our Asterisk PBX at our Hyd Office part will be in our scope,please mention the hardware Specification for installation Linux Software. Regards Reddy

We need to reconfigure the Cisco system from ISDN2 lines to SIP trunks.

...computer but I would like to place a SIP proxy and run a webphone in the browser. For that I will need an executable docker-compose file or detailed instructions for: - Frontend - inputs SIP parameters as ip, login and password - button to connect - disconnect button - button to answer the call - button to end the connection - no need to make calls, just receive calls - it doesn't have to look nice, I will adapt the frontend to our internal application. - you can use sipjs, jssip or any other open source SIP library. - backend a backend application to access the frontend, it can be nginx but it needs to be configured - an example asterisk server - running on port 5060 with UDP - a configured extension - SIP proxy / gat...

I need an asterisk developer to write the script for me …. It's urgent project for the header acceptance... asterisk doesn't accept the Tel: Header (from <tel:055XXXXX;noa=national;Srvattri=national>) we need to get accept and receiving the incoming call

- Setup a dialler on my server - further setup the dialler's campaign - dialler with recording is preferred (3rd part app with recording is also welcomed) - find a VoIP sip for this dialler's campaign with an unlimited budget plan for both inbound and outbound calling

my website is developed using CMS caller Perfex and I need you to integrate softphone so that my team can call leads using CRM only. Softphone Type: -with SIP login - work with VoIP channel with call recording

Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. Please let me know only if you know how to manipulate "SIP Response Code" in Asterisk SIP Trunk. Thank you.

I'm looking for a service to install Asterisk or Elastic PABX for my call center Must have recording feature and reporting module, also looking to integrate this PABX to existing in-house CRM

We are having some issue with OLD Analogue Phones. The call is answering right after the dial is initiated. So our customers are being billed for hearing the ringing tone. The caller gets the 200-OK message at the beginning of the call. None of above is expected. We need our customers hear fake Ring back, not billed & their CDR not updated as answered until the call is picked up by a real human. That's pretty much it. Please read it carefully and bid if you want. Budget $100. If you have any question then feel free to ask after reading the details I already provided.

Dear Candidates, We are looking for an asterisk, Freepbx and AWS cloud engineer who can take the role of supporting a telephone firm in a particular time. The engineer will require to take support during the +GMT 8 time zone. With that timing, the company allocates specific service level agreement to customers. The infrastructure consist of extension registered to assigned FREEPBX machine (EC2 instances) to one customer or a group of customer depending of the size of the signup. Call travels from their FREEPBX machine to a single central A2billing system with a facility of DID management. The trunk which this SIP service is running are directly connected to the A2billing or a gateway attaching a physical SIP/ISDN cables routing back the calls to A2billing. Each Free...

Hello I have issue with my Asterisk certified/13.18-cert2. it 's rejecting the incoming call with the header Tel: ... I need a small modification on the asterisk to accept the calls. I need developer who will develop small patch for me . find the attached invite

I am investigating on setting up 3CX for our company's needs. I am looking for a freelancer to help me set it up. I am a techie, but not an ex...environment does not allow SSH logins. (If I am mistaken and it does, hosting at 3CX is ok). Requirements: - Configure fixed DNS on Amazon EC2 / 3CX instance - Allow 3CX instance to send e-mails (I have Amazon SES account) - Configure 3CX according to the instructions document (see attachment) - Configure 3CX with Ringdesk (details in ) - Configure SIP trunks to allow outbound calls - Make sure 3CX can send e-mails - Finalize by making and receiving a call using softphone on Windows and cell phone I would prefer to work on a shared screen using Zoom. (You give instructions and I'll share my screen)

IVR Industries (and sub-industries) Manage audio lists by industry 1. Menus - each menu has playback file to navigate the listener what button to press. 2. Menus - a...hebrew?) * ALL THIS BUTTONS AVAILABLE DURING THE LISTENING 6. When the listener hang up or go back, it will save the point where he stopped to listen. When pressing # in main menu it will go back to position where he stopped last time. configuration: in Admin panel i will have option to configure the playlist to run in this variations (you have to explain me how to integrate it with Asterisk DB data): a. old to new. b. new to old. c. press number of track. if i press "150" it will start playing 150th track. 8. Ability to generate daily/monthly report of usage and minutes. 9. Ability to view online usa...

Hi. I'm running a SIP Trunk with asterisk. The problem is when we call a Analogue Phone the duration starts at the beginning of the ringing. We need someone who could setup the Ring Detection module of asterisk to bridge the call and let the duration start only when the ringing stops. More details will be described with the selected freelancer.

Hi. I need someone to configure STUN & ICE on asterisk. More requirements will be discussed over chat.

I need to configure asterisk to hide or mask the RTP IP of the vendor by masking the IP with a name. Please let me know if you have experience in this.

I need someone to help me preventing the 200-OK Message passed to the dialer in some cases. More will be described in chat.

I need an Asterisk Expert for couple of works. More details will be described later. You need to be able to create asterisk scripts and have to be expert on making dialplan.

I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tenant B <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk)

i need integration erpnext crm with issabel and make pop up on call show customer details and call record and ather and i need sorce code We should have incoming call popup with data being picked up from the ERP Next CRM Module for a existing number and We should have a option to save a un-known new number as a new Lead or Save to a existing customer / supplier / other contact similarly we should be able to do a click to dial from within ERP Next

I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tenant B <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk)

Need someone who can modify a client software for unified communication client software. Has knowledge of SIP client, video conference compatible with Asterix and Cisco CME

Requirements: - call reject/hang up, - call transfer, - no activity detection/silence detection, - DTMF detection, - Linux platform kernel version from 2.6.

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Nero Drinks Company is a family run business. We produce Nero Premium Vodka, a handcrafted, small batch potato vodka made exclusively of British potatoes. Nero Vodka is velvety in texture and smooth enough to sip neat. Equally its neutral taste and slight natural sweetness makes it ideal to mix in a cocktail or long drink. As a brand we see ourselves as the David to the Goliath of the big premium vodka brands. What we lack in size we more than make up for in determination. Our strap line is the 'Undefeatable Spirit' and our mission is to celebrate the undefeatable nature of our customers and consumers. Our target market is the 25-40 age bracket of males and females who like to celebrate but prefer quality over quantity. We advocate responsible drinking. Our produc...

we want to setup Bitrix24 with Sip Connector integrated to our IPBX

Wish to have a integration of a Asterisk PBX like FreePBX or something with ERP Next. We should have incoming call popup with data being picked up from the ERP Next CRM Module for a existing number and We should have a option to save a un-known new number as a new Lead or Save to a existing customer / supplier / other contact I need a source file similarly we should be able to do a click to dial from within ERP Next

We are a professional product development company with several projects. One new project is a client with the idea to incorporate a aerator into a wine glass - so when the customer uses the wine glass, it puts air into the wine, giving it a better taste and experience. Simply poor the wine into the glass, and then when you go to take a sip, the wine passes through the decanter or aerator, causing the wine to take on more air - this improves the flavor and texture. We will do this with glass so any cool ideas you have, let me see them. Looking forward to working with you. Global Product Makers

I am looking for an audio , SIP , WEBRTC, EXPERT to build a product like 1000% of features and operations. with some additions. with my own dashboard design and home page layout. You must know sip snd webrtc. This is NOT for first timer. I will NOT pay for you to learn. Apply ONLY if you have done similar and can provide sample link and code. If you don't like to listen and follow instructions DO NOT APPLY. if you are inexperienced and cannot work under pressure DO NOT APPLY.. This is only for serious professionals who are willing to work long term and not try to make money now and disappear. I am in this business 40 years and have heard all. I will provide sample layout of homepage and dashboard to NOT CONTACT ME IF YOU DONT HAVE SAMPLE OF SIMILAR

Hi. I'm looking for someone who has ALREADY BUILT a WebRTC app that I can buy, then pay to have it customized. I will only consider those that can send me a login of one they have built so I can test it. Here are the features/requirements: - Softphone that connects to my SIP - Chat/messages which works via API with my carrier - Video which works via my PBX/WebRTC - File sharing on chat - Screensharing while on call - Mobile apps (iOS and Android)

softphone that will register to my sip PBX as an extension. I'll load the EXCEL sheet with leads that I want to call to. the dialer will call them one by one. when it gets sip event of 200K (so it's a valid number)- I can set in the settings if I want to receive the call on the softphone- or it will transfer the call to another EXT or ring group that pre related in the settings. click 2 call big advantage activated by code! (i'll get 1M random codes in EXCEL)- 14 days trial.

hello I have a project that I want to use asterisk pbx or free pbx to call a caller random caller to answer 2 or 3 questions and capture the results in mysql or pass to an api call can you do job if so what would be the cost?

Hi, I need support setting up a basic on premise (RasPBX) Asteriks installation. I have the following devices: - Raspberry PI 4 with RasPBX installe...support setting up a basic on premise (RasPBX) Asteriks installation. I have the following devices: - Raspberry PI 4 with RasPBX installed - 2N technolgies Doorbell with Sip Support (works already and is registered to sipgate) - Several Devices with Sip Softphone installed - Home Assistant on several devices (I want to connect via webRTC (Browser) to the Asteriks) So I need: - basic configuration of Asteriks / FreePBX - 2-3 extensions / endpoints for the devices / softphones - 1 extension for the webRTC connection (e.g. ) - 1 configuration for the 2N Doorbell with SIP Support

we need an asterisk expert to install and configure the budget is 100 dollar you will show our tech how to do it

Implement asterisk solution - Asterisk Dial-plan, Asterisk Gateway Interface and Asterisk Manager Interface. - Configure Queue, IVR and Voicemail -Asterisk installation, configuration and troubleshooting. - installing and configuring Asterisk with PSTN card.

Need to correct a problem with fail2ban and registration of extensions in asterisk, 1 to 2 hours estimate.

We are currently facing VoIP issue on Mikrotik RB750G3. The incoming & outgoing calls can be done but no voice from both sides come out. The PBX system is NEC SL2100 & the SIP Phones are Flying Voice IP622CP.

Need asterix specialist, to design an architecture on Cloud and propose technical details how to set it up and integrate it with AzureAD...