From http://www.dslreports.com/forum/r18692448-VERIFIED-guide-use-Innomedia-Gizmo-on-ViaTalk-
- Disconnect your WAN port from the gizmo
- Open your browser (I used IE - cannot verify if it will work with any other browser). Go to »192.168.251.1/Voice_adminPage.htm - the IP Address will verify if you changed the default settings in the gizmo. If so, use that address, but DO NOT FORGET the /Voice_adminPage.htm at the end.
- Login using your admin password. If you do not know the password, try one available at »gizmopasswords.blogspot.com
- Turn off provisioning if you have not done this already. To do this, go to NETWORK -> Provisioning setting. Uncheck the "Enable Provisioning" box and hit the "Save and Reboot" button. After rebooting, you may not be able to login to the admin account since the reboot will default to index.htm. Hit CANCEL, and go to »192.168.251.1/Voice_adminPage.htm to login.
- Go to VOIP -> SIP Proxy and enter these settings:
- SIP Proxy - Enter exactly what was sent in your email from ViaTalk. Remove any whitespaces at the end.
- Check "Use Outbound Proxy" box
- SIP Local Signaling Port: 5060
- SIP Domain:
- Registration Expiration time: 60
- Preferred Codecs List: PCMU/8000
- Choose "NONE" for all others.
- Click "Save" button.
- Go to VOIP -> User Account.
- Choose Line 1 from dropdown
- User ID: Your 11-digit phone number from email (starting with country code 1)
- Password: From email
- User Name: Same as User ID above
- Authentication ID:
- Use Hot Phone Number: NO
- Hot Phone Number:
- Use T38 FAX: Yes
- Click "Save" button
Check under the "information" to see if line 1 is registered
Note that when the port of your SR number to ViaTalk is complete, you'll need to use your SR number in place of your ViaTalk number in these configs.
6 comments:
I confirmed you do not need to put in "user name" and "authentication ID". As written you put your phone number in the "user id". IMPORTANT: make sure you include the 1 before your phone number! For example 12125551212. The voip light should go on within seconds.
ALSO IMPORTANT: I get a busy signal for the first phone call I make after changing the authentication. The second phone call works without a problem. So don't be fooled.
The BroadVoice service works, too.
Simply put the SIP server's IP address, and port 5060, in the old SIP proxy field, and put the phone number in the auth and user fields, and the password they supply in the password fields. BroadVoice provides all of the needed settings in the generic SIP information that they provide in their Account information screen. The only thing that isn't quite obvious is that the default SunRocket codec is also the BroadVoice codec (though, BroadVoice calls it something completely different). I have an Innomedia and have it working in this configuration rather well (and the gizmo registers faster with them, and the call quality is a bit better, too). However, this is temporary: when (if?) my numbers port, I will be setting up an Asterisk server with a slightly more complex configuration, if other companies go the way of SunRocket... I will be ready.
I wasn't ready this time, that's for sure.
Has anybody gotten the 2-line capability to work with the innomedia?
Hello,
I am walletless from dslreports - the one who originally wrote these instructions. I have also uploaded instructions for setting up the 2nd line on viatalk using innomedia gizmo on my blog: http://gandhi.wordpress.com/
Great information on Via Talk...
Post a Comment