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Internet gibberish: How to get Gizmo SIP services running on your Asterisk@Home Server (v. 2.7) (Updated - 21 March 2006)

I have tried and tried and tried and finally succeeded (after succeeding once only to squash it because I wanted to do more). To properly set up a SIP connection to your Gizmo account so you can use Gizmo for outbound calls, you need to have the following info set:

General Settings
Outbound Caller ID: 1747XXXXXXX (your Gizmo SIP number if you wish)
Maximum Chanels: 1 (I haven't tested higher numbers)
Outgoing Dialing Rules (I didn't make any changes or fill any of the fields)
Dial Rules: blank
Dial rules wizards: default (pick one)
Outbound Dial Prefix: blank
Outgoing Settings
Trunk Name: gizmo (set this to what you want the trunk to be called)
PEER Details:
allow=ulaw&alaw&ilbc
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=proxy01.sipphone.com
fromuser=1747XXXXXXX
host=proxy01.sipphone.com
insecure=very
secret=your_password
type=friend
username=1747XXXXXXX

(for username and from user, replace with your Gizmo SIP number)

Incoming Settings (I've left all blank for now) - UPDATE - see below, not needed
Registration
Register String: 1747XXXXXXX:your_password@proxy01.sipphone.com

I then set up a dialing rule that requires you to dial 8 to get your Gizmo "line" and it works. So many of the instructions I found were half baked, didn't work, etc. You must especially have fromdomain and fromuser set for outbound calls to work. Sometimes the internet is filled with a lot of crap...

Later, I'm going to try to figure out how to set up incoming calls.
UPDATE: Wow... Inbound works fine as long as you have your routing properly set up - i.e. leave the above settings alone. If you are having problems, it is likely that your router is blocking certain incoming ports.
Continuing Notes (21 March 2006):
While I don't completely recommend it, you can put your Asterisk server as your/in your DMZ (if you have a single IP or are on the type of connection where you have no control over your IP, then that means you are probably using a SoHo router - e.g. Linksys, Netgear, D-Link, SMC, etc. - and you can place your machine "outside" your firewall to make sure the proper ports are open to you Asterisk server). Once you've figured out it works inside the DMZ or on a static IP with no firewalling, then you can slowly close down ports. The better way to approach this would really be to open up ports slowly, but this can be time consuming because for some reason, it can sometimes take some time for your incoming calls to be routed properly.
If you understand the following jargon, then the following port openings will work fine (if you're behind NAT, this means you need to port-forward the following ports to your Asterisk server):
TCP: 5060
UDP: 69, 5004, 5060, 5061, 5062, 5063, 4569, 5036, 8000, 9999-20001, 2727
The ones I bolded seem to have been part of the key for my setup - but, I'm doing quite a bit more than running my Gizmo inbound and outbound calls through this box.
Because my Asterisk server runs on a static IP, I simply had to open up my main firewall ports to that box.

Comments (2)

Bob Pogue:

Thanks Steven. Your solution has me up and running after I spent several hours sorting through other gibberish. I appreciate you sharing your experience through this Blog.

Mr.Suradet C.:

Thank you very much Steven. Your solution make me to success callout with Gizmo.

I have a headache for a week.
Thank you again. :)

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This page contains a single entry from the blog posted on March 16, 2006 12:30 PM.

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