Originally Posted by UAVirgin
SpaceBass, all your ranting about Asterix has pushed me over the edge. I'm going to go... take the plunge.
I have the PC, the x100p card and a freedigits account. My present Voip provider is Vonage but I plan to dump them for broadvoice.
Here's my questions. Using trixbox and the x100p card:
1). am I correct in thinking that I plug my Moto 5ghz wireless base station into the x100p to get dial tone?
2). can I connect to the Asterix server via my wifi UTS starcom F1000 phone and make an outgoing call on the pots line?
3). could I redirect an incoming pots line call through the asterix server to the UTS starcom wifi or sip phone?
3). what does the x100p card really do for me?
4). on the configuration side does it make sense to have my Asterix server in the DMZ or behind my firewall (linksys befzx41)?
That's the begging of my questions.
Its like a frickin' bat signal!
Glad you are taking the plunge....
1) This one took me a second, but I think I follow the question. Do your currently get dialtone from a rj11 (phone) jack on the Moto? If that is the case, then yes. The x100P is a card a lot like a modem that hooks into an POTS/PSTN line... so Vonage through the moto base station looks just like a PSTN line to the x100p.
2) yes, you can point your UTstarcom phone at Asterisk quite easily. If you are using it on the same line you'll have no problems. If you want to use it outside of your network you'll need to forward the approprate ports (more on that in a second) and use your public IP address
3) This is where it gets tricky because the answer to most any "can I do.." question is "yes"...meaning you can do just about anything you can think of. The short answer is yes, you can direct incoming calls to any phone. In that particular case, the UTstarcom is a client of the Trixbox server...its an "extension". So you just set it up like "when someone calls my house number ring extension 1001, 1002 and 1003...but if someone calls my other line (office) only ring 1001 and 1005 and my cell phone "(which is external to asterisk, but it can do that too).
4) a LOT of people put * in their DMZ. Frankly I think its a pretty good idea if you take some basic security measures. For instance, make sure to install the SSL (secure web) modification (yum -y install mod_ssl from the linux command line). I have slight concerns about someone using an exploit to gain access to the box and using it against the rest of your network or even for a dDOS attack or something...but Centos (linux) is pretty solid....
The other option is to forward the right ports. SIP- which is what BroadVoice and most of the others provide is a little harder to get through NAT than is IAX2. However, IAX2 is a little less popular with providers currently. That being said, I'm told Telasip will do IAX "trunks" if you ask.
For SIP:
UDP 5060 (standard SIP port)
UDP 10,000 - 20,000 (standard RTP ports)...you can greatly reduce that number by configuring the range in this file: /etc/asterisk/rtp.conf .... of course all of those files are accessable via the web interface so you dont have to use vi and the linux shell at all.
Also, from with in the web gui there is access to help chat, thats a worthwhile module to install...lots of sharp resources hang out there
I'll PM you my email as well... glad to help and really interested to hear what you think as someone with fresh eyes. I may be a little too wrapped up in this sometimes to know just how easy or hard it is to install and use.
-N