Asterisk web sippekerjaan

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...

need a cloud pbx that can be connected to my grandstream ip phone. I need to be able to get calls in one area and then calls can be rerouted via extensions to other phones. This will include softphone solution that can act as a call centre. Multiple lines must be housed. This will not include SIP. It should be able to run all on the cloud and be housed on our server.

I need a cloud pbx that can be connected to my grandstream ip phone. I need to be able to get calls in one area and then calls can be rerouted via extensions to other phones. This will include softphone solution that can act as a call centre. Multiple lines must be housed. This will not include SIP. It should be able to run all on the cloud and be housed on our server.

We want to Build a total Voice call Solution with Incoming Missed call, IVR, Etc over Asterisk

looking for customizing Linphone for iPhone and android, Need "AUTO LOGIN WITHOUT USERNAME AND PASSWORD" (username and password from a2billing table), PUSH NOTIFICATION SHOULD WORK TO RECEIVE INCOMING CALLS AND FCM MESSAGES, SMS OPTION SHOULD WORK WITH ASTERISK SIP SOME OPTIONS OF LINPHONE to simplefy linphone interface. we have a sample source code and customized linphone dialer TO SPEED-UP THE PROJECT. After finish the project, we need the final source code.

Hi Erewin, I noticed your profile and would like to see if you are interested in helping with my project. I require FreeSwitch / Asterisk expertise to help debug a SIP trunk issue. There is further work with push notification later. We can discuss any details over chat.

I have an IVR with 3 avaya 9608 sip phones. I would like: - To have a call queue so when people call during business hours it puts them on hold until we can answer the call - To be able to transfer calls to another phone - make the voice recognition work. Right now it doesnt work and I dont know why. For example say "two" rather than press 2.

i need someone to teach me how to use the magnusbilling 1. the ...number like 1250, can not call out. 3. set sound record, and now some line didn't have the sound record issues 4. the Routes all submenus 5. Voice Broadcasting 6. time in the system not right. 7. if we want to made a call, create sip account, the tariffs, the trunk, the prefix, etc, what the steps, which one is first, which one is last, order by order? 8. and also every time my local ip changes, it can not ping, i have to add IP address to the server, if not do that way, the sip account sometimes can not connect to the server, have other way to solve that. 9. sip account for spy call. 10. how to configure call out with the GOIP gateway 11. if some one call the mobile number from the GOIP ga...

We need an Asterisk AGI expert to make some changes on our Platform, We already have asterisk setup and custom AGI, we just need someone who can make changes in the existing AGI and change the incoming call flow. Please post only if you have experience.

we need experts who can fix all problem in the actual setup

we are looking to have a new system administrator for our asterisk free PBX system we need expert for some hard task remains things to fix in actual setup

I currently have a script --- that runs on a SIP server -- to DIP my provider for the CNAM for the call. I am changing providers and only need the script modified to use the new provider. The new provider's directions are below -- Attached is the existing script that needs modified.

The business model is a PlayCafe & Spa, it will be designed with boutique play spaces for children. The play areas will be hands on, most spaces will be facilitated by a Playpal Instructor. Some areas will allow independent play and equipment exploration. The business will have an area that feels upscale, and modern for parents; especially for moms to enjoy a chair massage, tan, exercise, glamour, sip a latte, or enjoy her laptop. The PlayCafe Spa supports the needs of both; parents and children. The Playcafe Spa eliminates the need for an outside babysitter, decreases separation anxiety, teaches healthy independence to children and new parents, and creates a culture of wholeness. The boutique play based learning areas, will promote creativity and exploratory thinking amongst c...

I need someone that can configure Asterisk like a walkie talkie server with Intercom/Paging, and create a simple php web page with a button that uses a long press event on call button to send a dial command with an auto answer on the other side (auto answer) and another event of on key up to hangup the call.

React native VoIP( SIP ) client app needed

...load these in and calculate the overall length of the semester and then determine how far along the semester the snapshot was taken, creating a percentage of how far along the semester is. The second header line gives the headers for the following lines. Following the headers, the courses are listed in order by course name. Each line prints the following: A single character marker. This is an asterisk if the projected enrollment is greater than the total cap. The course name The current enrollment in that course (as of the last snapshot), summed over all offerings. The projected enrollment for that course on the add deadline. The total cap for that course, the sum over all offerings of the overall caps. The second header line and the following lines should be arranged into lef...

Hi Aleksandar, I have read about the project "Create SIP Server with WebRTC"..and seems exactly what I need. please could you tell me more? thanks

I am planning an startup company and looking for Freelancer who can help me build an operational API for my Video conferencing solution. Our requirements are as under: Sr.# Module Name 1 One to One V...(User Picture, Full Name, email address, mobile number, company name, designation, and default meeting password). 17 Speaker Statistics 18 Option to raise a hand (virtually for asking questions or grabbing the attention of the host). 19 SIP Integration 20 See statistics of the video conferencing activity 21 100% secure communication (end to end) We require Source Code (Operation based API calling/ Endpoint), API referencing (API input/output details), local deployment guide, minimum five (5) sessions with our technical team and complete cross platform (web/mobile/e...

...need it to be designed and sent in good quality please. Once completed, Please use the logo to create labels for each of the following variants. Note that you will need to remove the word "Juice" and replace it with "Dessert" or "Fruit" etc. Dimensions of Label: Height=53mm Length=130mm -Dr. Dessert -Dr. Fruit -Dr. Ice -Dr. Green -Dr. Tabak -Dr. Salt -Dr. Dairy -Dr. Strange -Dr. Candy -Dr. Sip The following information needs to be within the labels. Warning: This product contains nicotine, which is an addictive substance. Keep out of reach of children. Not for sale to persons under the age of 18. Product Info: Nicotine...( mg) VG/PG Ratio...(70%/30%) Flavor... Symbols (Bottom Corners): Dangerous to health Not for sale to persons under the...

Olá, Preciso de um profissional para desenvolver um Softphone SIP que eu consiga ler QR Code do meu PABX (Asterisk) e configure automaticamente o ramal no softphone sem a necessidade de configuração manual. Este softphone tem que funcionar para IOS e Android.

The PJSIP is an open source SIP stack , we want to compile it for Android NDK The PJSIP is developed in C and support all SIP features as per the RFC standards. SIP, SDP, RTP, STUN, TURN, and ICE. It combines signalling protocol (SIP) with multimedia framework (WebRTC) and NAT traversal functionality. Please bid only if you have good hands on experience in compiling SDK and good knowledge on VOIP/SIP protocols and working with low level network layer/API. Much have C/C++ experience with JAVA.

Hi. I need opensips expert for hiding or modifying the RTP IP before reaching the termination site. Means I need to modify the RTP IP from the sip trunk running on Opensips as a different IP, before it reaches the termination side. So the termination site could not see the actual IP address.

I am creating a brand named "trinklethings" for various types of accessories for homes, people, and kids that is fun, happy and unique. I'm looking for a simple logo that includes and asterisk like star * and maybe some rounded lines. I prefer the esthetic of a lower case "t" vs uppercase. My brand is all about cuteness, simplicity, and practicality. It would be nice to have something that would look appropriate on each of the above mentioned accessory types.

i want developer for my own mutualfund software who make this kind of below report 1. SIP Performance Report Based on XIRR Calculations: which effectively calculates internal rate of returns where there are multiple transactions happening at different times (SIP) 2. Capital Gains Report: Category split of investment in long term and short term based on holding period of investment 3. Total Returns Till date Effective % of Returns calculated till date based on amount investment and current market value 4. Day on Day Change Effective % change day on day in market value of funds 5. Fund Holdings in major sector High level fund holdings in sector such as Tech, BFSI, Automobile etc

I want developer for my own mutualfund software who makes below report 1. SIP Performance Report Based on XIRR Calculations: which effectively calculates internal rate of returns where there are multiple transactions happening at different times (SIP) 2. Capital Gains Report: Category split of investment in long term and short term based on holding period of investment 3. Total Returns Till date Effective % of Returns calculated till date based on amount investment and current market value 4. Day on Day Change Effective % change day on day in market value of funds 5. Fund Holdings in major sector High level fund holdings in sector such as Tech, BFSI, Automobile etc

i want frelancer for my mutualfund software who can import data from cams & karvy and make report for client as valuation report, capital gain report & sip report ,Aum report any many more report any one who can work before this type of report pls contact us

Tôi cần phat triển phần quản trị cho giai pháp ai smart dialer va cần nhân sự biet ve asterisk python va php mysql