On 01/23/2011 05:44 PM, Genes MailLists wrote:
> Any suggested readings for configuring the ATA for this case ?
It should be reasonably straightforward. Once you get the basic stuff
like IP information set up, you'll want to go to the Voice --> PSTN
Line in the web UI. You should only have to change a few settings:
Proxy and Registration
Proxy: {IP address or hostname of your Asterisk server}
Subscriber Information
Display Name: NO CALLER ID (or whatever you want to appear when
an incoming PSTN call provides no caller ID)
User ID: {device name in sip.conf}
Password: {password}
Dial Plans
Dial Plan 8: (S0<:{IP address or hostname of Asterisk server}>)
PSTN-To-VoIP Gateway Setup
PSTN Caller Default DP: 8
FXO Timer Values (sec)
PSTN Answer Delay: 3 (required for caller ID)
I believe that's it. The above dialplan will cause the SPA3102 to
forward PSTN calls to the Asterisk box as soon as it picks up. You'll
want your context for the device to Dial() whatever phones you want to
ring.
You'll find that the SPA3102 is an incredibly flexible device with a
ton of settings. Most or all of them are documented here:
http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf
Good luck!
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Ian Pilcher arequipeno@xxxxxxxxx
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