We need a Survey Module for Asterisk or FreePBX. We require an IVR phone survey module that can be used to evaluate the quality of customer support calls and agents. At the end of the call an agent will ask the customer if they are prepared to answer a short survey and the call will be transferred to the survey module. The customer will be asked
Necessitamos integrar CRM versão 7 ou 6,5 com provedor pbx asterisk voip de modo que seja configurado para abrir tickets automaticos, leads automáticos de acordo com regras do provedor de pbx. O provedor de pbx possui regras de inserção de cpf ou cpnj para novos clientes e opções para chamar ramais das áreas comerciais, financeira, suporte, ...
ihave a asterisk ivr the ivr needs to get digets from calle and send them to an curl Press a student number and then # Enter the last test date in four digits Enter a questionnaire number later, press # Send data to CURL https: // topstudent .credit123 the ivr exten gets the response from the ws if response =1goto extn1, if response
hello, i am currently using a custom asterisk voicemail application. it was supposed to be mini-voicemail, but there are all kinds of extra options on it. so, i am looking for someone new to do it right. 1. user tutorial to setup voicemail 2. passcode entry 3. one user name 4. only one greeting, not several 5. greeting can be up to 3
Hi all, We need a plugin for Web, Outlook (plugin identifies the number phone on the website using the tel: ; callto protocol) which will work on the principle of 1. Clicking on the number selects the SIP account I have programmed on my IP phone, softphone, etc... 2. then dials the number to the Customer (+/- callback) The program - must select two
I need a script and or function that will trigger an external shell program be called, when any incoming calls come in. I then need the route to be handled as normal, regardless of the output of the externally called program. I need the external function to be passed the inbound DID, and the the caller id.
...machines and the option for sending faxes also. Let me know your experience, and your recommend program and server setup. Project to start ASAP. Skills: Amazon Web Services, Asterisk PBX See more: vicidial, goautodial, amazon ec2 setup, amazon aws setup, setup web server amazon aws, setup goautodial, need install vicidial, install amazon ec2, vicidial
It shold be a mo...notify the guy when new parts have been arrived to the shelves that are needed for reparing of some device where he is in charge. Include build free sip-sip calls using asterisk technology to call a manager, support team and guy who is responsible for suppliing parts if the part does not exist on a shelve or it is not working.
I need you to develop some software for me. I would like this software to be developed for Windows using Java. Configure Asterisk for video/audio conferencing over WebRTC and Mobile Apps. Our frontend include web app and mobile app will need to make video conference between patients and doctors.
We currently host an asterisk server which interfaces with a MySQL database, a script polls the database daily for customer payment reminders and makes outbound calls to contact the customer via cell phone and relay payment information; the customer can request limited information on a one-time basis via text at the end of the phone call. The customers
Hi, I need VoIP Business training - What i basically need to learn is Centos and Asterisk and how it works I also require trainer to teach me mysql and How to create Database on DBFORGE to access server and makes entries. The trainer must be in uk Thanks
...alternative to provide what I need below. I need solutions that make since and keep my PBX in place. Please don't recommend me changing to an entirely new system unless its Asterisk based and it can be done with a few changes. If you are a developer and can add to the system then that's a plus. Below is what I'm looking for: 1. EasyVPN or OpenVPN
We have several SNOM M25 and M65 Phones connected to a SNOM M700 Base. The Snom uses FreePBX Asterix 14.0.x (latest) as SIP-Gateway via PJSIP connected to SIP Gateway of German Telekom. All the Phones are properly connected and the Peers registered. Somehow, we can't manage to dial out or to be dialed in and we don't know why. Need an SIP Expert to deal with this and configure the system...
we have had a power cut and now out asterisk PBX wont come back online, this is the issues we get please see below. PDOException: SQLSTATE[HY000]: General error: 1030 Got error 134 from storage en gine in file /var/www/html/admin/libraries/BMO/Database
... p:1328 11. bootstrap_include_hooks() /var/www/html/admin/[url removed, login to view] 12. require_once() /etc/[url removed, login to view] 13. include_once() /var/lib/asterisk/bin/fwconsole:12 PDOException: SQLSTATE[HY000]: General error: 1030 Got error 134 from storage en
After making changes and upgrades on system, some issues appeared, that must be fixed. To do this You must have a very good knowledge of Asterisk PBX with WebRTC ! This is nothing for [url removed, login to view] must have time to get acquainted with the system before You start and be able to work a few hours during Eastern Standard Time on workdays. No Skype or similar