Greetings: Thank you for taking a moment at my post. I have a dedicated Linux box with Ubuntu [url removed, login to view] installed on it. Have also installed Asterisk but have not configured it. I would be looking for someone to setup the SIP information and install a soft phone and get it to dial out. Furthermore, to protect my PBX server I would want whatever possible
We are a small manufacturing organisation with 3 different locations in different states in India. We also have a sales team of around 10~15 employees with another 10~15 employees at the backend. Current telephony solution is OLD legacy Land line phones in conjugation with mobile numbers. The problems that we face: 1. Sales employees leave the organisation
Hi Asterisker, I have an asterisk job for ya. i have a newly built Asterisk Now that migrated configs from another Asterisk. currently can't receive calls with the same exact setting i used to receive calls on with previous and have some error msgs on it need ur help fix it all and make it operational.
- Asterisk Setup and config as per requirement. - SIP server for Voice, IMs, Video calls with codec setting, disable other things. - Config to use FULL server resources ( Ram config/ Network config ) - Load balance high volume calls. ( currently we will use only 1 server, but needs to be modular for load balancing ) - Dial plan configuration
Hello, I need that if customer hangup from their softphone or if me providing termination to someone.. When their user hangup call ,, ...hangup from their softphone or if me providing termination to someone.. When their user hangup call ,, call will be still active in our both servers. I need to do it from asterisk. Let me know if anyone can do that..
We are seeking an expert asterisk / FreePBX developer to connect our phone system to our customer database and helpdesk. Use Case: We have customers who need support that call to our customer center. We want to identify their called ID and automatically retrieve a support ticket connected to their phone number. The ticket will appear on the screen
...second number to the 102 extension. * 1 internal PBX (Asterisk) that has 2 more endpoints with extensions 200 and 201. They should be accessible to the FS (FreeSWITCH) endpoints and the FS extensions (100-102) should be accessible to the Asterisk endpoints. Endpoints connected to Asterisk should be able to make outside calls too via the external
Necesitamos desarrollar 3 paneles en PHP para 3 perfiles. Estos paneles deben llevar implementados Web Rtc y serán integrados posteriormente en Asterisk.
Hi, Looking for some one who can teach installation and customization of Asterisk freeswitch a2billing kamailio all in standalone server and connecting of writing application
...call USEING by GSM MOdem per modem have 1GSM SIM CARD i have totall 40 MODEM so MODEM connected to USB HUB and HUB connect to pi usb port PI Have install asterisk now if any call come to asterisk server calls have to out from GSM modem to mobile GSM modem work as like a GSM GAteway ====================== need a GUI to dial USSD command for check balance
Estamos necesitando un sistema dialer de IVR que el mismo pueda: Llamar de forma simultanea a una lista de numeros sin limites de canales. El software tiene que soportar 5000 canales simultaneos. Debe quedar registrado la hora del comienzo del llamado, la hora de la finalizacion Crear encuestas utilizando audios nivel profesional, para realizar las preguntas a los usuarios y grabar sus respu...
1- To configure the open source Linphone Flexisip to interact with my voip server using TLS protocol. 2- Customize the app to interact with my freeswitch/asterisk server to do registration , logo, color, configuration, etc customization - show me how to do this
Hello, I have Freepbx 13 asterisk 11 a Cisco Unified IP Phone 8851 with BEKEM V2. phone is powered by PoE. When the KEM is connected to the phone, the phone displays an error "The BEKEM has been detected but is disabled by the administrator" and the KEM remains switched off. Everything else works fine