Asterisk definity sip trunkεργασίες

tool to connect my asterisk via AMI user and will send HTTP command on each EXT event to a specific address. eg: ext 200 is ringing - the tool will send HTTP command to the PC [the PC is running 3rd tool to react- u will not deal with it 3rd tool] the tool will get a list of the PC's that r in the same network. accurding to the events that will pop up [AMI] the tool will send a specific command to each ip address according to the event. ringing - incoming call Incoming call Outgoing Call DND-ON DND-OFF Terminated Hold UnHold Reject Incoming Call Answer New Incoming Call Call Remote Canceled each event has a specific formula- just the ip address will be changed- by the list of PC

I have a dying maple tree that will need to be cut and I would like to preserve six to ten feet of the trunk and have it carved with various images.

...and grow together! We are looking for an agile developer with skills in multiple technologies including Node.js and angular, ideally some php as well with the ability to adapt to different technologies by self learning. We are looking for attention to detail, communication and reliability. Mandatory expertise Nodejs Angular React NestJs Websocket Mongo OPTIONAL Sails Ionic Openvidu/mediasoup Asterisk Chatbot php symfony !NOT NEGOCIABLE! You are expected to work in GMT+2 timezone. We pay 2'500USD per month (can be reviewed depending on seniority) and offer 20 days of paid holidays per year. TO APPLY 1. Let us know your experience in years and short description of your work performed on all mandatory elements and ideally on optional elements. 2. Send your CV 3. Put your...

i have a freebox asterisk setup that i need some debugging. needs to be available right away.

...through SIP and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Summary The goal of this project is to develop an Android application that can send calls through SIP and forward them to the GSM network AND receive calls from GSM and forward to SIP. The application should then forward the audio and convert from VoIP to GSM and vice versa General Deliveries The application working in APK format Full source code Simple manual for compiling and generating the application from source A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project. Features - Route call from SIP to GSM - Ro...

...colour 5. The font for Paragon Green that I want to use is New Times Roman or similar. 6. flat, (not 3D) 7. Once design agreed, a version with the "TM" trademark symbol and a version with the ® symbol EXAMPLE - The attached mock up is what I was thinking, but the tree came off the internet and I don't know who the owner is, so some sort of variation of that would be good. The Sn is in the trunk and very subtle which I like. Probably prefer single green colour leaves. I'm also happy to run with something different if someone comes up with something really great. NOTE 1 - Once the design is agreed, I'd like versions in white, green and greyscale - all with clear background. NOTE 2 – Also, once the design is agreed, these 3 colour v...

...for someone to make for me a python script that interact with VOIP Using PJSIP. The script must call a number and use gTTS ( Google Text to speech ) to say predefined text. Also it must decode DTMF to get user input. Please if you feel you can do it message me. Full Description: Using Python Programming Language: * PJSIP. * gTTS ( Google Text to Speech). Register and Login to VOIP Server via SIP using PJSIP. Script initiate a call to a nunber using a call function. The call function take argument, FromNumber(international format ), ToNumber(international format). after the call is established. Using gTTS Play (This is automated call Press 1 for english or Press 2 for spanish). DTMF Listener will get the tone and determine which direction it will take. if user press 1. he go to...

I would like to hire someone to help me select a Cloud Directory (LDAP), VOIP, and SIP solution. I would like a cloud-based, centralized directory service, like an LDAP, so that I can create a new user and have the option to select from the below-listed services and have an account created for the user in those services. I don't want all users to have the same services so it is important that I can choose which services are added to the user. Here are the services that I use and would like to integrate into my Cloud Directory service: Azure Google Suite Slack Asana NextCloud Zoho Suite Twilio Additional conditions: I would prefer that this LDAP service be self-hosted, probably on Azure, and be an Open Source solution, if possible. I need the ability to c...

I face a problem installing FusionPBX on Debian 10.8 OS. Always invalid sip profile for external and internal. I need to fix that and know what was the problem and how to fix that in the future

I would like to develop an app, like a drin...event" kinda game. Like x-factor, the national team in sports, or just some big. And make it like a live drinking game for events, so when something happens as someone gets voted further into x-factor you get a live notification. I will post 1 picture for 1 event that went down in Denmark. In the first picture, it contains a list which says basically: It's the new years eve speech by the queen. 1 sip: When she talks about covid. When papers turned. Takes some water. 3 sips: Names "Denmark" Names "Greenland" Wipes her mouth 5 sips: Names "The youth" Names "Vaccine" Names "Clima change" Bottoms up: Say "God take care of Denmark" (She always says it at the end of ever...

Need a fly made for the grand opening of a clothing store. Colors must match logo: black, red,, white. Sip & Shop Feb 5th from 12-6 pm raffle and prize giveaway 1& 4 pm. Shop Envy (Located inside Tampa Mall booth 433) 5400 E. Busch Blvd Temple Terrace Fl 33617 813-616-2525 Instagram _shopenvy1

We have customized android linphone version hardcoding the sip server and integrating SOAP and REST web services, we need expereinced developers to look at the android code and apply teh modifications to the ios version. If you don't have experience with Linphone app don't submit your proposal. Refrences are requested as well

Hiring freelancer on call basis to support our Asterisk bases call center solution. We are using Dinstar UC2000-VG (32 port) GSM gateway for our telecalling centers. We have installed 10 Nos new Jio Sims (4G). We need to enable 4G in the device to activate sims.

We are looking for a Freelancer who has previously done projects with the 3CX Call Control API. We want to capture in real-time multiple events, such as incoming and outg...real-time at the time the call is made. There will be an ini file for the service to read and save settings to eg reading any settings from the 3CX registry/ini file to then use on future start. We also need to be able to initiate calls by calling the application using a REST interface. The service must work on 3CX v18 Linux. You will need to have your own working version of 3CX, extensions, SIP trunks, etc to test on. Please tell us about your experience with 3CX API. Full project details and final pricing will be negotiated once we review freelancers. If you DO NOT have experience with the 3CX API, DO NO...

For my VoIP, im using FreePBx on Cisco 7911. If I create new extension, the phone is stuck on "registration" Logs show that: Registration from '<sip:>' failed for '' - Wrong password and if i check the phone status itself: Error Updating Locale even the locale show is right.

Hello, We need to implement CGrates with our current SIP infrastructure. The infrastructure in mind is an SBC server connected to a cluster of 2 Freeswitch. We would like to implement this as soon as possible. Thank you.

Hi, We are looking for a professional developer who is an expert in React.js, electron & knows some WebRTC to debug an issue while answering calls on our electron app when used on an iPad/iPhone or iOS device. Having physical access to IPad or iOS device is a requirement for this job. Please apply only if you meet the requirements. Detail...electron & knows some WebRTC to debug an issue while answering calls on our electron app when used on an iPad/iPhone or iOS device. Having physical access to IPad or iOS device is a requirement for this job. Please apply only if you meet the requirements. Details will be shared with person shortlisted for the job. As you can see the error in the screen shot from when we push the Answer button for our SIP Soft-phone. Attached are 2 scre...

We are looking for someone with experience in ip communication. The main areas we are looking for experience in are - pbx including grand stream asterisk and fpbx - we need someone asap - thank you

The goal is to fix the error in Drupal user registration. Error: DrupalCoreEntityEntityStorageException: SQLSTATE[HY000]: General error: 1366 Incorrect integer value: '...[:db_insert_placeholder_2] => user [:db_insert_placeholder_3] => 0 [:db_insert_placeholder_4] => en [:db_insert_placeholder_5] => 123 test ) in DrupalCoreEntitySqlSqlContentEntityStorage->save() (line 846 of core/lib/Drupal/Core/Entity/Sql/). Steps to reproduce: 1) go to site 2) select login 3) select create new account 4) fill in all fields that are mandatory (have an asterisk in red) 5) Submit Error above occurs. Please note that freelancer must work weekends and be able to start upon award and finish within 1 day of award. The freelancer must understand Drupal 8 and PHP with MyS...

Custom CRM with Lead management and following functions: 1. send SMS 2, Send email 3. Send Wapp 4. call record with SIP calls

Hi, We are looking for a professional developer who is an expert in Electron & WebRTC to debug an issue while answering calls on our electron app when used on an iPad/iPhone or iOS device. Having physical access to IPad or iOS device is a requirement for this job. Please apply only if you meet the requirements. Details will be shared w...expert in Electron & WebRTC to debug an issue while answering calls on our electron app when used on an iPad/iPhone or iOS device. Having physical access to IPad or iOS device is a requirement for this job. Please apply only if you meet the requirements. Details will be shared with person shortlisted for the job. As you can see the error in the screen shot from when we push the Answer button for our SIP Soft-phone. Attached are 2 screen...

Доброго часу доби! Необхідно налаштувати asterisk push notification для "пробудження" iphone перед дзвінком Щось на кшталт цього:

Looking for a ASterisk person to assist us, We installed Issabel PBX Created Trunks Created Dial Plans Created Out and Inboundroutes however having issues with calls failing to and from the PBX from our gateway.

ENG A plugin for VoIP Client is required. We already have the VOip Server (Freeswitch + FusionPBX). Our application is built upon @ionic/angular 5.6.3, capacitor 2.5.0 Our application needs to be deployed in iOS 14 SPA Se solicita crear plugin para proyect...required. We already have the VOip Server (Freeswitch + FusionPBX). Our application is built upon @ionic/angular 5.6.3, capacitor 2.5.0 Our application needs to be deployed in iOS 14 SPA Se solicita crear plugin para proyecto en Ionic Framework (@ionic/angular 5.6.3, capacitor 2.5.0), la cual permita realizar llamadas (solo audio) por protocolo SIP, conectado a un FusionPBX. Su uso debe ser de forma similar al plugin de cordova (cordova-plugin-sip). El plugin debe funcionar en versiones android 11 y superior, ...

We are looking for a PHP Developer to create an Asterisk call center page that allows us to generate statistics based on inbound and outbound calls to our telephone number. We will provide a copy of the database with a specific layout to work on and once the task is completed the functionality will be implemented on our system. We need to generate statistics based on the daily/weekly/monthly (and so on) comparation of data such as amount of inbound and outbound calls, call duration, missed calls, avarage waiting time for the answer to the call. We will work in milestones (5x €100,00) Depending on the level of skill of the freelancer and the work provided we can decide to cooperate for further tasks Indian, pakistan and chinese applications will not be evaluated

We currently use Twilio/OPENVBX but are searching for an upgrade as the system is 5-6 years old and has some bugs. We are a small office, but would like a system we can scale. Asterisk, VoiP, Freeswitch, etc.

WebRTC • Experience with Strong WebRTC platforms • Experience with WebRTC (direct use and/or debugging of the WebRTC APIs) • Knowledge WebRTC serv...Knowledge WebRTC server technologies. • Actively involved in all phases of our WebRTC product lifecycle. • Experience developing applications on a VoIP infrastructure • Integration of Web and Mobile Applications with WebRtc Technologies for Audio ,Video , Screen Sharing and Meeting • Knowledge and experience of the following protocols and platforms: SIP, WebRTC, RTP, REST APIs, Linux • Knowledge and experience with one or more of the following: Asterisk, OpenSIPS, Kamailio • Good knowledge of media codecs, formats, transports and container protocol Webrtc • Good knowledge on Video...

I need a simpe script to make phone calls from a webpage. It will only be calling one number, so on the page I only need a green phone icon to start the call and a red phone icon to hang up again. Here is some info I found: It needs to run on my shared hosting, so it's not possible to install additional software.

Hello Freelancers, We have linphone based android and iPhone application. Both apps are working fine. But the issue is background calling. Means If app closed then its not working. so we are looking for the developer who can help us to achieve it via push notification. We are using Asterisk based system. Thanks

This project is to setup a web SMS management software and gateway (Gammu + PlaySMS) along with Asterisk (freePBX) system that also uses the USB modems for calls in and out. The project must support SMS in and out with multiple usb modem dongles and sim cards simultaneously with calls in and out

I have attached a PDF with all the information to get started. I am looking for a VoIP Server that can be set up on my AWS Server and instead of normal phones connecting I am going to have some custom hardware made to connect to it for Two Way Radios.

I have attached a PDF with all the information to get started. I am looking for a VoIP Server that can be set up on my AWS Server and instead of normal phones connecting I am going to have some custom hardware made to connect to it for Two Way Radios.

I need someone that can remote in and reset the loop trunk cards.

We have a custom build of MicroSIP. This is used for Win/Mac/Linux (using Wine) devices. Source code for our custom build of MicroSIP can be provided. We would like a similar application for Android/iPhone devices. Audio codecs supported should be G.711 and G.729a. Video codecs should be H.264. We will maintain ownership of the source code for both platforms.

...phone numbers - users should be able to choose phone number through dial prefix) 2. We have SIP Trunks that needs to connect to the new Issable image. 3. Follow the image for the IVR flow and create the IVR flow 4. Creating Ring-groups 5. Configuring Standard firewall. Sip traffic should only traverse the firewall if the other side of the call is outside of the office or at a remote location accessible by VPN 6. Make new recordings according to the script provided. 7. Create a xml file for contacts which will be integrated to the system so that when contact added / removed from the file it will be automatically refreshed. 8. A shared voicemail for sales and tech accessible for all users 9. Check SIP connectivity to see if all the trunks are up after the whole setu...

looking for someone who can set up server with Indian SIP domestic calling service

looking for some one who can set up server with Indian SIP domesting calling service

Looking for a skilled developer to help develop custom features and apps to work in hand with FreePBX. This would be an ongoing relationship, as our customer base is constantly putting in feature requests. First project would be to develop an application that links to PBX users and utilizes SIP Trunks for sending and receiving SMS messages. Within the sending and receiving, we will need the ability to create groups for mass outgoing, but in a way that each end user is messaged individually so the responses only come to the sender and not to each member in the group. Additional projects will be discussed with the awarded developer.

to fix dial plan to do the following : 1) dial the trunk and ignore the first congestion from trunk because when we get congestion the call keeps flowing to the trunk so we want to ignore the congestion I get and keep the call when I put 2 dial command it works but I want to ignore that can we ?

Hello everyone, There is a FLUTTER project for which designs are already prepared. Please find the zip file attached with the designs. Need to only integrate this VoIP SDK: Here are some features we want from that SDK: -- Voice Call; -- Call Transfer; -- Conference Room; -- Record Calls; -- Call on Hold; -- Call History; -- Contact List; -- Settings: User information; SIP settings; -- Authentication with license Final output will be the Android and iOS project submitted to the store with the SDK integrated for all these screens. Source code will also be shared for analysis with the chosen candidates. TERMS: 100% Payment will be funded to Escrow, but will only be released once project is complete and approved by the

I need to integrate CyberArk and Asterisk -Issabel for password vault.

Hello, Few call center employees need to work from home, so VPN server + 3 clients need to set up, to be used by existing X-lite application. Input data: - All is working on Windows. - Before 'Remote VoIP' software was used before and partially congifured on clientagents side, but now VPN server gone, - PBX is Asterisk-based; also Yealink, windows server 2012, - Call center agents use X-lite old version as free SIP-phone tool. - DynDNS address of the head office, as head office IP address is dynamic.

I'm starting to use Accredible to generate the certificates for VoIP School. I want to create custom certificates and badges for the school. The current certificate and badge (I think they are really ugly) i did it myself. The badges will have the title Verified (Asterisk/FreeSwitch/SIP) professional I'm uploading the current certificate and one badge I did (as I said terrible) and the VoIP School Logo

we are looking for freelancer who is able to install and configure Asterisk with FreePBX and USB dongles for voip termination (mobile). If you have the required know how please do not hesitate to contact me and say which hardware requirement should we have and the duration do you need to finish the project and the price to do this.

Hi, I want to have an asterisk installed on the digital ocean with a customized caller id.

Hi Marcelo B., I noticed your profile and would like to offer you my project. I saw you have already done a few integrations for Linphone with asterisk and moved away from Flexisip. I am looking to create a very stable system and dont want to waste time using the wrong software and then having to do it again. Let me know a good time to talk. my whatsup [Removed by Freelancer.com Admin]

we are looking to build system where we can generate script based calls which is going loop back to our server . also we need to maintain ASR and ACD as per predefind critrea

Hello, Few call center employees need to work from home, so VPN server + 3 clients need to set up, to be used by existing X-lite application. Input data: - All is working on Windows. - Before 'Remote VoIP' software was used before and partially congifured on clientagents side, but now VPN server gone, - PBX is Asterisk-based; also Yealink, windows server 2012, - Call center agents use X-lite old version as free SIP-phone tool. - DynDNS address of the head office, as head office IP address is dynamic.

Hello, Few call center employees need to work from home, so VPN server + 3 clients need to set up, to be used by existing X-lite application. Input data: - All is working on Windows. - Before 'Remote VoIP' software was used before and partially congifured on clientagents side, but now VPN server gone, - PBX is Asterisk-based; also Yealink, windows server 2012, - Call center agents use X-lite old version as free SIP-phone tool. - DyDNS address of the head office, as head office IP address is dynamic. Time to finish project : 1 day, 2 max