call pop-up on inbound call

[saigop]

Hi,

I have a GoAutoDial 3.0, and a inbound campaign with list uploaded. When I call from mobile number the values are not poping up by matching the caller-id.

Any customization has to be done in the dialplan level?

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

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 should also post: 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. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Note that "goautodial 3.0" is only the installer (useful information!) but the "Vicidial Version with Build" is required for free support on this site.

4) When you receive a call, your Carrier (Telco/VOIP Company) will send information which can be viewed in the Asterisk Command Line. Sometimes "sip debug" must be turned on to see enough detail to get the inbound caller id, and if that's true there are sometimes adjustments needed to allow Vicidial to see (and act upon) the caller id information. If you need help in getting to the asterisk CLI and/or turning on SIP debug, please let us know and we'll help.