our voip system has issues, cutting, breaking not clear voice IVR sound is breaking our voicemail is always full, need to be backed up all the time every 2 days. need to upload the VM daily to external server, and back up as well. the work has to be done via teamviwer
I have this PHP Files (Spanish Instructions) [url removed, login to view] I need put this code on my Elastix Server This is the file [url removed, login to view] I need help for test this code. I have a Grandstream gateway.
Need help to set up properly Elastix for my project. What im trying to achieve: I need to receive call from another external PBX and then terminate it to specified destination using SIP trunk configured on Elastix. This is super short term project to run our experiment but if successful we pay discuss long term partnership.
Necesito implementar un botón para realizar la llamada desde una pantalla hecha en JSP con java. El botón se debe conectar a un servidor Elastix / Asterick de donde salen las llamadas. Debo registrar quien realizo la llamada, la duración, el destinatario y si le contestaron; los botones que debe tener la pantalla son: 1 llamar, 2 Colgar.
I would like to hire someone who have knowledge in writing API or code for my custom agent CRM. We have design the Custom agent CRM in .Net looking the below API for elastix PBX 1.) Total active call and waiting call 2.) Number of agent login and call status 3.) DND/Break From agent panel.
I have: 1 Elastix Server 1 Grandstream Gateway IP with 1 Line FXO I need have any method for: - Upload a File with a lot of numbers (Maybi XLS, or CSV) - Upload a SoundFile for Dial to this numbers - Call automatizated system for all numbers. The process i need Call number by number and play this sound File. Considerations: The calls
Develop or install an automated and predective dial module to run on elastix. It is not a question of using the current basic module available. The application must transfer the calls answered to an attendant, operator or queue group. You can only play a message (voice broadcast) and generate reports with the results of the calls.
Need someone to make some change to A2Billing on an Elastix server to make possible a caller to call a member without having his member ID so make the system recognized the number that is called ( a member one) and then make the call through. As system already recognized a number the second times it calls, need to make it the first time and add this
ESPECIALISTA REDES DE COMUNICACIONES Y TELEFONÍA IP, Telefonía IP en Asterisk, freepbx, Elastix.. Conocimientos de Telefonía IP en Asterisk. PARA CONFIGURAR MODULOS DE NUESTRA SOLUCION DE TELEFONIA IP EN CLOUD BASADA EN ASTERISK. PRIMER MODULO A DESARROLLAR SISTEMA DE RECARGA DE MINUTOS PARA OPERADORES, ESTOS PAGAN PARA PODER RECARGAR DINERO CON
Looking for an experienced Linux Administrator w...experience. Mandatory skills: Centos7,6 Asterisk Optional Skills Expeience with 5 Softswitch Asterisk, Freeswitch, Kamailio,Opensips,vicidial,A2billing, Free pbx, elastix, PRI & PSTN card installation, call centre setup, Advanced IVR, Vmwire Esx, web-meetme ,Cloud solution using asterisk
Installation of GoIP SMS Manage Server on CentOS linux (Elastix 4). I alreary use an Elastix Predictive Dialer whith a GOIP8 as a GSM SIP Trunk. I whant to send an SMS message automatically through GOIP when the campagn ends to retry 2 times each celphone number without success.