Hi there,
We are replacing an 'old' IVR system - 4 port Dialogic hardware card connected to BT (our UK PSTN landline provider) with a PSTN system called FeatureLine (which is essentially just a hosted remote PBX). Calls come into our Dialogic card, where the server is running an IVR program called VoiceGuide.
As we have about 20 or so servers collocated in a data centre, we think it is about time to replace our 'old' system and go VOIP.
Our 'old' system answers the calls and prompts the caller to press 1, 2, 3 or 4 after which the Dialogic card puts the call on hold and sends the call to the appropriate extension by announced transfer. When the called extension is picked up, a .wav file is played to the callee with the name of the caller. This is done by a database script that looks at the Caller ID and if it exists in the database it plays a specific .wav file, e.g. fredflintstone.wav, which we record for each of our clients and which simply says the client's name.
So when we answer a call we hear something like "Fred Flinstone - press 1 to accept, any other key to send to voicemail". Thus, most of the time we know who is calling and can decide if we wish to accept the call or not.
The IVR system is connected to a SQL database and we have a website through which we can control it from anywhere. We can also change the destination and send calls to an external landline or mobiles - all controlled from the website. The whole process is started when an .EXE is run when the call comes in and the CLI parameters are passed to the IVR system and database.
We now have the 3CX IVR working very well i.e. - press 1 for sales and it transfers the caller to our office landline or we can pick the call up VOIP. We can change the configuration very easily through the 3CX web interface and we are very happy with this as it is, but would still like to emulate part or all of the Caller ID facility we had with our old system.
Usually, people look at the Caller ID and if they want to talk to that person they pick up the phone, if not they let it ring until it goes to voicemail. This works fine in the UK for both PSTN and VOIP, but we work abroad sometimes where the PSTN telephone network does not always forward Caller ID. However the Internet latency is long which means VOIP from the UK is not that good, so we use PSTN when abroad.
1) Does anyone know if the can be a variable (say $divertnumber) could be set in any those fields?
2) Can a .wav file be played to the callee when the callee picks up the phone?
There are a few things more that I would like to do - too many to list at the moment.
Sorry that the above is so long!!
Regards - Colin
