Page 1 of 1

broadcast dial to customers with a pre-recorded message

PostPosted: Wed Sep 29, 2010 6:47 pm
by richmac
hi all,

Is anybody know?

how to set broadcast dial to customers with a pre-recorded message..

Please help..

Im using vicibox redux 3.0.3

Thanks,

PostPosted: Wed Sep 29, 2010 7:25 pm
by mflorell
Read the ViciDial Manager Manual

PostPosted: Wed Sep 29, 2010 9:47 pm
by williamconley
when you post, please post your entire configuration including (but not limited to) your installation method and OS with kernel or version, vicidial version and build, asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)
________

you seek the functionality of "Survey" which is in the "Detail" menu of "Campaigns" (on each campaign, there is a "Detail" button at the top which will reveal the "Survey" section).

Read the manual for the use of this feature and you will have what you want.

Please do not use it inside the US, however, without a good lawyer looking over what you are doing first. :wink:

how to transfer calls to an agent

PostPosted: Sat Oct 02, 2010 11:23 pm
by richmac
hi matt,

Thanks for the reply. I can now broadcast dial to customers with a pre-recorded message.

But the problem is how to transfer calls to an available agent?

The settings of my Campaign->Survey is default and I login in to my eyebeam using sip extension.

On the customers end I try to press 1 but nothings happened.

Is their any other configuration to be made?or what is the next step to transfer to available agent?

Vicibox Redux 3.0.3 | Vicidial 2.4-280 Build 100912-0842 | Asterisk 1.4.27 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation



Thanks..

PostPosted: Sun Oct 03, 2010 9:36 am
by williamconley
if you press one and nothing happens we need to verify that the DTMF is being passed from your carrier back to you (some carriers will block this capacity to stop press one campaigns).

use sip debug in the asterisk CLI to see if this information is being transmitted to you.

also helpful if you post (just for this carrier, of course) your "account entry" (which may have a dtmfmode entry in it) and your dial plan entry.

it may be that you are not accepting the dtmf or that they are not sending it, or that they are not sending it in a way you can accept. (so you can set your dtmfmode in the account entry to request it in a different manner). this can also be affected by the codec in use.

for more information, you can also google dtmfmode (this is an asterisk function, there is a lot of information available on many sites, in-depth, on the topic of dtmfmode)[code][/code]