Hi Ali i need do option my the new system 1:- i need do system for the recharge card 2:- i need to do option for the ussd wich allow the employ to control all thing with easy 3:- i need add new culomn for the page which count all hit come from asterisk 4: - i need to do timer for every user which make me to control how many hour the user will used the service 5:- i need add option for the user page which allow us to stop or enable every user by check box 6:- i need add option which can control the time between the user like the max of the secund
Hi Ali i need do option my the new system 1:- i need do system for the recharge card 2:- i need to do option for the ussd wich allow the employ to control all thing with easy 3:- i need add new culomn for the page which count all hit come from asterisk 4: - i need to do timer for every user which make me to control how many hour the user will used the service 5:- i need add option for the user page which allow us to stop or enable every user by check box 6:- i need add option which can control the time between the user like the max of the secund
Hi Engineer, we like to invite members of the freelance to build a cloud phone platform with us. Engineer must have deep understanding on database, onboarding UI and linking of UI with asterisk system.
Hey everyone, I've been digging in to Asterisk for a while but I'm not a developer so I may not have all the concept right. I'm open to any suggestions that come from a professional developer. I'm looking to setup a multi-tenant asterisk server working on the Asterisk Realtime Architecture. Since its going to be used for multiple customers it must have a structure that is going to allow me to update it(as adding and editing extensions, IVRs, trunks or dial plans) real time without any downtime (not even seconds). I need to record calls by adding them to a DB so that an Administrator can listen to these records by a web-based gui in the future. I also need to voice mail service to be active. I want extensions to be able to listen to their private...
I have asterisk working with goip, I need to enable sip errors in asterisk and create a script to check sims are not budget is 25$ for this project, please do not bid high amounts I will not entertain such bid if you are expert of asterisk and know goip.
Must know, php, mysql, sms technology, voip, freeswitch, asterisk. ubuntu, python We are looking to retain a working for many projects. Our intent is to provide projects over years to come. You will have to log into our site and understand what is currently there in order to be able to edit, add , update.... Please inbox me for more info
I need to use Cloudflare as DNS load balancer by failover Or, let's say a round-robin load balancer using the Cloudflare API to add and remove nodes from a pool of IP addresses. I found similar code on Github but it's outdated and needs to be fixed. Additionally, I'll need the following functions. Sends a notification to Telegram or Prowl when any IP fails and when the app was executed Any additional advice to make it more useful is welcomed.
Hi , I need a Freepbx and asterisk spec to help me on some stuffs . I will take a test with a simple task then we go through main project.
Hello, We need help to setup Analog Voip Gateway with Asterisk, model: ~ Remote session is provided with TMATE, so be ready for it. Thanks in advance
I need to use cloudflare as DNS loadbalancer by failover, in which any specified port can be listened. Or, let's say a simple round-robin load balancer using the Cloudflare API to add and remove nodes from a pool of IP addresses. I found similar code on Github but it's outdated and needs to be fixed. Additionally, I'll need the following functions. Sends a notification to Telegram or Prowl when any IP fails and when the app was executed, also a mode which only sends important notifications must be included. Any additional advice to make it more useful is welcomed.
Backend software project for professionals, coded on for 15 years locally in South Africa, provides cloud file, SMS, fax, email client, cloud PBX, accounting system, and much more, need expert developers who can contribute significantly and at speed. Hurdles with old code, old technologies and many programmers who thought they can code who 'coded'. If you are 1. not an expert who is prepared to take on a challenge 2. not available 3. cannot code and read any php code at speed (no matter how bad the previous programmer was) 4. has the word 'impossible' in your vocabulary, please do not apply. We need someone who can read and rewrite existing code (often very old) in powerful ways. Prepared to negotiate substantial incentives or rates if coder successful on parts assigned...
asteriks management with phpagi libraries and outbound IVR ascertainable experience
We have a project within Asterisk IVR about 1000 lines of code containing: - Option selects - Webhook calls - Database lookups - Automated dial-out we need to change our telephony engine to Twilio and adjust automated dial out to use Twilio as well. Language considerations: PHP or Node.js Attached is the IVR script
Install vtiger connector to running Asterisk 11 and vtiger 7.2
This is to change a small software program into an ISO file for easy downloading by agents of a call center and make some adjustments to the existing software . Must use Teamviewer to connect to our Multi Disk Server.
We had to reinstall an asterisk server, and now the dahdi T1 cards are not refelcting that they are bing programmed for a T1. This work needs to be able to worked on now
i try to connect with asterisk throw pami but got some errors any help
Searching for somebody who can find out issues with asterisk and fix it. Thanks!
...and 1 day to deliver 2nd part please bid according to your time cost to complete the project. We will create initially milestone for the 1st part of the project. Once its concluded successfully we will create the 2nd milestone. 1st) Build, install and configure for simple tests the open source project Its a 1-2 hour task if your have experience with SIP and asterisk - Milestone Max is $100 and release will be on installation completion - time to deliver 1 day. Awarded freelancer should set this up on his server and we'll make a call to/from the telegram SIP gateway and if it works we will offer remote access to our server to do the installation on our server and instructions sent on how to install and configure. 2nd) Create a service like
Estoy buscando un panel web básico para asterisk, con las funciones básicas de. -Mostrar el número entrante y obtener el historial -Posibilidad de clasificar las llamadas por grupos, esto manualmente a una db. -Grabación de las llamadas y reproducción. -Lista negra, para rechazar esos números. No me digan FreePBX, eso lo podría usar para manejar asterisk a nivel profundo, yo necesito uno básico. Si fuera en phpagi, sería genial, sino ni modo en .net o java, al final en python.
Need configuration of the FreePBX to send SMS by gsm modem. My Freepbx works on raspberry pi 4 B with 16gb card, and has 1 modem connected with a sim card. There is 1 extension SIP trunk connected GSM to VOIP, and VOIP to GSM. Im connected to freepbx by zoiper app in my mobile phone. In short way: In my mobile phone im using "zoiper app" which is connected by SIP to Freepbx. I...send and receive sms from gsm sim card (modem is huawei e160 in raspberry with freepbx) using a zoiper or other voip app. Curently text mode in zopier doesn't allow me to send or receive gsm sms. [check images] All recived sms messages going to previously email address (spam inbox) in freepbx settings. IMPORTANT: Need skilled person to do the job, only serious performers and freelancers in a...
ASTERISK EXPERT to set up PBX We are looking for an Asterisk Expert to set up PBX. Interested freelancers can bid for the project.
1) Configurar un asterisk con un troncal. 2) Configurar una vez que se recibe la llamada la siguiente aplicación: 1. El cliente mete un código por teclado o deletrea el código alfanumerico *Si se deletrea el código alfanumérico se necesitará una conexión a la API de Google. 2. Chequeamos o en una base de datos local o del cliente (externa) (api configurable) 3. Nos devuelve un texto 4. Le preguntamos al llamante si quiere escucharlo (pulse 1) o quiere que se lo enviemos a su móvil por SMS (pulse 2). 4.1) Si pulsa 1 Reproducimos el texto con text to speech 4.2) Si pulsa 2. Si lo quiere con SMS le preguntamos móvil de destino. Enviamos SMS al móvil (para ello necesitamos poder configurar una API para...
Hi guys! I need an experimented guy with a heavy knowledge of Asterisk , SIP Protocol , PATTON E1 gateway, Issabel IPBX and RAD ASMI-52 modem for assistance on E1 trunking configuration between the PATTON gateway and my Issabel IPBX and E1 signalisation troubleshooting between my Patton and my ISP RAD ASMI-52 modem. Patton model is SN4171/1E15VHP.
Searching for an expert who can audit asterisk server and fix existing issues.
Preciso criar uma solução para que o gestor de PBXIP ISSABEL possa ter acesso ao retorno das chamadas realizadas pelo CALL CENTER de forma amigavel, ele devera acessar com login e senha e visualizar as campanhas com status de ATIVO e visualizar em lista em formato de excel o status de forma reduzida para NOME | NUMERO CLIENTE | BAIRRO | STATUS DO FORMULARIO INFORMADO pelo o operador, ele poderá baixar estas informacoes EM EXCEL e manipular os dados, poderá ter tambem a possbilidade de criar tambem graficos com base nos dados do STATUS informado por periodo.
Hello, We are looking for the expert who can work on the Asterisk server and telephony integration.
Hello ... i'm looking for someone who can develop or has already a IPTV/OTT middleware with Billing system., the application will need to be able : -customers management / STB, android, smart TV - info on STB/player where the customer can see when the subscription is out -posibility to send messages from server directly to screen -catch-up / recording feature -loadbalancing -EPG -VOD -LIVE TV -ADVERTSING OPTION this project is urgent, so here are the minimum requirements that i expect : 1) Must already run similar project which you can modify to fit my needs, then please provide me a demo to test and see how it runs, then we can discuss details of all the modifications. 2)if you never run such of project , don't bother bidding, i do not want to waste my time or your time. ...
We need help to install a Kamailio infrastructure with Asterisk, for few millions of calls per day. Solid experience required on Asterisk and Kamailio. Special Development Module for AS* and Kamailio. Development on the core code for AS*. Training. Database configuration. Monitoring script with Zabbix. GUI interface. Physical appointement in France (Paris-Toulouse) or Swiss (Geneva).
I have cdr from asterisk ( my sql db) I need to choose several columns ( filter them accurding to the info inside) Than to send excel sheet on spesific time to my email. I can add many reports as i want. It should be a tool for windows.
Create a service like We need: 1) Our clients calls from Telegram app to our Telegram account should go to our Asterisk PBX via SIP 2) Be able to from our Asterisk PBX call our clients Telegram accounts. 3) Handle 50 concurrent calls Open source example:P roject Here you can see how the bot works Libraries used: Current TG2SIP version can handle about 100 simultaneous active calls with opus@48k codec on ordinary hardware. 50% milestone on delivery of working project connected to our PBX for tests 50% milestone on delivery of source
Want to convert avaya 1608-l to sip so that it can be used on asterisk pbx
Create a service like option to use their open source project so that our clients can call our asterisk PBX from their telegram accounts. Here you can see how the bot works Libraries used: Supported call types are: SIP->TG call by username SIP->TG call by phone number (if callee is registered in telegram) TG->SIP to fixed SIP URI (can be set up in TG2SIP settings). Current TG2SIP version can handle about 100 simultaneous active calls with opus@48k codec on ordinary hardware. 50% milestone on delivery of working project connected
Can anyone make avaya 1608I work on Asterisk pbx
Searching for an expert who can audit asterisk configuration and can make the PBX secure from outside.
We are looking to develop an app (very much like UBER, but for other services), it must include the back-office website to administer all request and integrate to our current telco and VOIP network (Asterisk, and possibly Twillio for audio and video calls), appointment and project management platform. It must be able to track minutes, clients information, user information via a database.
we need asterisk configure in our local PC for telecalling and tele marketing purpose scope of work install & configure asterisk (essabella) basic setup trunk inbound and out bound voice clip sending moudule setup
Hi, we are a small startup from Poland. We are looking for someone who will help us with Asterisk integration. We have our own application based on Laravel Framework. We want to improve our app about dialer features most calling, receiving calls and recording calls. Do you have any experience of that type of work which could help us?
need to enable call recording on asterisk service for 20 seconds to match available recording so the can take further step
Dear Sir or Madam, I am searching for an expert in VoIP and Asterisk or alternative program. What ist he problem/task? I am going to create an android app like „Marcophono“, which you can find in the german playstore. With this app Users can do prank calls. It works like this: 1)User A choose a scenario in the App. 2)In this scenario User A can see different Audio Messages. 3)User A calls User B with the App. 4)If user B answer the call, User A can press different buttons in the App like „Hello my friend“, „How are you“ etc. User B hear that messages directly live, but not the voice from User A. Important: User B did not have this app installed on his phone. Thats the point I need your experience. I think the owner of „Marcophono“ did ...
i need a php script which connects to ami asterisk (this is already done) then it needs to set dynamic agents to be remove or added to/from a specific queue and also we need a shell script which logs off all agents from a specific queue
i need a php script which connects to ami asterisk (this is already done) then it needs to set dynamic agents to be remove or added to a specific queue then we need some jquery tool which polls from ami and displayes if user is logged in to queue or logged off. and also we need a shell script which logs off all agents from a specific queue
We are looking for some Asterisk work on a number of items. We are not using freepbx or other GUI. The current tasks consist of the following: The configuration is two SIP trunks from Twillio. One trunk works the other does not. Need to troubleshoot. All calls coming into the default extension of either of the trunks must simul ring to all extensions existing now and any that are added in the future. On one trunk this is configured for existing extensions only. Needs to work on newly added extensions as well as second trunk. Notification of all missed calls to default extension must be sent to a specified mobile number via text. Any voice mail message left to that extension must also be sent via text and email. Notification of all missed calls to other extensions must be s...
Task - Enable DTMF ( press 1 to connect) if pressed then forward to an external number and if no response disconnect the call.
Asterisk Freepbx Admin - Zoho phone bridge Integration
Hi Ram i have two asterisk conmected via IAX, when call is NO ANSWER or BUSY or CANCEL im not getting good Sip cause but 503, exemple client send call he get ringing 180 than after timeout (no answer) he get 503 instead of 408, the gateway is sending correct sip cause to 1st asterisk but always congestion(503) except when call Answered I think something wrong in or or in my My Architecture is CLIENT --SIP--> ASTERISK --IAX--> ASTERISK --SIP--> GW
Ura com reconhecimento de voz IA para asterisk. Text-to-speech e reconhecimento de voz.
When making a outbound call from Skype for Business softphone unable to hear the ringing on SFB app. But the call rings at the other end and when answer the call works fine. Current setup is when calling from SFB has a trunk connecting to Asterisk freepbx and the SIP trunk to the provider is configured in Asterisk. it would be great to have someone who has knowledge in both Asterisk and SFB server.
