I read in a previous post that NAT isn't supported with this phone, I would have to disagree (it could be that this feature has been fixed with the latest firmware). I have many clients who have a dedicated PBX at cari.net and have phones at their office or at multiple offices all behind a NAT firewall and connect to the dedicated server running Asterisk.

Just an FYI.

please somebody helpme to configure a aastra 9112i with a linksys spa9000, how change the parameters for this ip phone, i also i have linksys spa400 configuring to the spa9000 please helpme send to sadomask@gmail.com

During a recent setup with ~85 users using firmware 1.4.1, users were complaining that the audio quality was tinny.
I found the following settings dramatically improved the user's perception of the audio quality:
handset tx gain: -5
handset sidetone gain: -10

I second the comment that the Aastra support is great. They *understand* the concept of support.

Most of you probably think this phone only supports 3 lines. Well this phone actually supports 9 lines by programming 6 of the 7 extra buttons on the right side. This is the cheapest 9 line phone on the market.

However I must agree that this phone has a few "MINOR" glitches.
1: The feedback sound to make the phone live is too high;
2: The phone does not seem to support reinvites, or I just don't know how to enable it;
3: The phone does not work through NAT (but for an office PBX that makes no difference);
4: You need to restart the phone when you change the programmable buttons.
5: Can take about 1-2 minutes to get phone back online if you don't setup TFTP config files.

The Support is great from Aastra though, they love taking in the feature requests and try to implement them as quickly as they can. I must say that I have better support from Aastra than I ever got from Linksys (who uses prewritten messages to send off support answers and can not get it right.)

Mustardman, sorry it took me so long to reply...been swamped
To answer your question...On the GXP2000, when a BLF is showing busy state, the key will dial that number when pressed. On the GXP, when the BLF is showing ringing state, pressing the button will send **<ext number>. In asterisk, in your dial plan all you have to do is define an extension like this: exten => _**.,1,Pickup(${EXTEN:2}) and it actually works quite nicely. This is documented in the Grandstream section of this wiki..What would be nice is the ability to control what is dialed in this state, but I can define dial plans specific to each phone manufacturer.

Alan,

Regarding the feature to be able to pickup on a line like the GXP2000. Is that not BLA which Asterisk does NOT support. How does the GXP2000 do it if asterisk does not support it? Please explain in more detail.

Alan,

The BLF without speedial was fixed in v1.3.1. I just tested it and it works just like it should. Now all we need is for asterisk to support BLA so that we can use that added feature in these phones

Yep, found the gain settings on google..
and yes the sidetone gain does indeed change the level of handset feedback, but even with it set to -10, there is still this 'other' echo. I say echo because it seems to be at a different time to the sidetone.
So in summary, sadly i'm still not happy with this phone. Maybe its a hardware thing, in which case its never going to be fixable. We'll see.
On the plus side, i have discovered the snom360, initial trials are good, although its 50% more expensive, feature wise its got it all, and whats more.. theres no echo in the handset :o)

1-Configured phone to use additonal lines beyond line 3. Lines can be configured, but when selection of the line is made on one of the programmable buttons or an incoming call to one of those lines, it does not light to indicate it's in use, ringing, on hold etc...even though it can be used and is functional (but not user friendly for the masses). Configured to interact with Asterisk PBX.
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2-When a button is configured for BLF it works great at a basic level, however is useless for any other function. Would be nice to see it function as a speed dial to the same number it is monitoring and additionally be able to pick up on that line via a feature code...similar to what latest firmware on the GXP-2000 has done.
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3 - Anyone else notice this BUG? If you are speaking to someone or listening on speakerphone and have the person on mute and then put them on hold, when you return to the call, the line is still muted (mute still flashes when you return to the call on that line via handset. However, when you try and unmute, you can't though, the light goes off to indicate unmuted, the mic is still muted. The work around was to go back to speakerphone and then back to handset.
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4 - Pressing a DND programmed button...does anyone know if the light next to the programmed button is supposed to indicate status? Currently all that is indicated is a small easily missed icon on the display.

Tech support (via the link on Aastra's site) shared with me these settings:

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As of 1.3 there are a number of configurable parameters dealing with voice quality that can be used in a station’s config file (ie MACADDRESS.cfg).

headset tx gain:
headset sidetone gain:
handset tx gain:
handset sidetone gain:
handsfree tx gain:

All adjustable between -10 and 10. However most users report that the following setting provide the best quality:

handset tx gain: 10

handset sidetone gain: 0

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As a side note, their tech support is awesome!