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 (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... 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 "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. 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) I see you tried to post some version information, but let's be clear: The name of the software here is "Vicidial". Goautodial is Merely an Installer for Vicidial. Asterisk is the underlying telephone server. But Vicidial is what makes the magic happen and you left off your Vicidial version with build entirely. lol.
4) This post is in the Vicibox Server Install and Demo board. It belongs in the Support Board. I'll move it.
5) UDP port range 10000 to 25000 should also be forwarded. However: The best method is to get the Vicidial server its own IP address that's public directly on the internet without a router and without NAT. The challenge is that if the other side of the call (agent phone?) is also behind a router, now there is NAT twice. SIP protocol doesn't handle double NAT at all. So many routers have a SIP algorithm to "help out" SIP calls. Some work, others "not so much" and some aren't active. Different on every router. But if you forward UDP port range 10k to 25k, you may get lucky.