Saya membutuhkan yg paham dan pengalaman utk melakukan setting/configurasi asterisk saat ini sdh terpasang akan tetapi masih mengkonsumsi cpu server boros sekali rencanaakan dipakai utk call center dg jumlah agent 100 orang.
Tengo un servidor Asterisk Instalado y en el mismo tengo Vicidial instalado. Tengo un problema al momento de interconectar mi Asterisk con un Softswitch y es que el puerto de comunicación de mi Softswitch es el 6060 no acepta peticiones por el 5060 que es el puerto estándar de Asterisk. La cuestión es que he intentado establecer la conexión y no he
Hola, Tenemos un servidor freepbx queremos que quede habilitado para ser usado con ciertas caracteristicas mas informacion al privado. Hi, we have a server freepbx we want it enabled to be user with some configuration more information at private
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.
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)
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.
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.
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
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. Ther...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.
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
I have a Raspberry Pi and have installed (initially) the Asterisk + FreePBX Per this documentation: [log masuk untuk melihat URL] 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
...un presupuesto para lo siguiente: En la empresa tenemos 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
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
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 -
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 [log masuk untuk melihat URL] 2) аптайм приложения 99.99%
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
...Resource Tech Appraisers Then make the document into a fillable form for Mac / PC, Internet. It will be sent to people to fill it all out and submit it to send back. Red Asterisk are required fields - All fields should support Alphanumeric characters. Under Additional equipment and Notes should support max characters for a long message, if any.
I am looking for someone to quickly put together a recording demo using either Asterisk or Freeswitch where the user is able to do the following: 1. Login into a basic site (will provide access to a DO server) 2. Have the option to start/stop a recording (simple button should be present for the user to click) 3. Once the recording is stopped it should
...system integrated to 1 sip trunk, VoIP phone and softphones A fully secured system that is only accessible to admin and users registered in PBX Full documentation of how to in Freepbx eg: add new PBX user, configure softphone, setting to be used for softphone, VoIP phone config and settings Auto-attendant with MOH for inbound Leave a VM by pressing a number
i need to integrate 8 port dinstar with my own CRM
...horarios de la centralita. 3. Solucionar problemas con el asterisk que no muestra debug en el cli 4. No puedo conectar las extensiones en los teléfonos ip. 5. Configurar gateway Dinstar de 8 Puertos y configurar las llamadas de cada uno de los puertos a una extension específica. 6. Actualizar asterisk de la version 13 a la 16 ...
Hello, My freebpx is not connecting to asterisk and i need a expert asap to help me with it, will be via anydesk,
We are looking for someone who can setup an asterisk server for us. It must be setup with correct security. Furthermore, you must show how we setup new agents and DID numbers with queues and opening hours. We do not need an interface; it can be directly in the conf files. A ”phone” must also be made, like this image, where it is possible to Answer
I need someone to develop a custom Asterisk - FreePBX module/application, this must have; - I want to make a keypad recording and audio player on call, for example (I call someone and want to play an audio stored on database and if user inputs #123 on keypad I will get that output)
Estoy buscando alguien que me ayude a terminar de configurar el Freepbx que tengo con una troncal SIP. El servidor tiene que conectarse a 2 interfaces. La red local y la red de la troncal SIP (tiene configuradas las rutas necesarias). Las llamadas luego de cierto tiempo se caen y hay que reiniciar los servicios.
Hello Viktor, we haven't talked for a while. I just show a project where you integrated Asterisk + MS Teams. We've got a client who wants to use MS Teams with a trunk from our Asterisk. We're not sure if it is worth the effort. So, can you please let us know the price for that integration. It would be enought with a step-by-step guide and support from
Hi, we have new installation of freepbx and asterisk. We created extensions and used sip app to connect. One phone can call another phone, but both parties cannot hear each other. Error is "Couldn't negotiate stream 0:audio-0:audio:sendrecv"
we already have a running and functioning Asterisk system installed. we just need someone to look at the code and find the reason why some calls are getting dropped (2-3 %). and some other simple issues.
This is a new Asterisk 18 installation from scratch, preferred a Docker/Podman container. For this project, you need to set up a new IAX trunk with one of our partners and set up the dialplan as per the specifications received. Basic calls routing from incoming to less than 10 VoIP SIP phones.
...milestones: Milestone 1 Setup fully working Asterisk 18 as Docker container with ODBC MySQL remote connection: Dockerfile Asterisk config files Milestone 2 Migrate existing Asterisk application (dialplans, extensions, trunks, bash scripts containing Asterisk commands etc) from Asterisk 11.7.0 to Asterisk 18 container from Milestone 1 with MySQL connec...