[asterisk-users] Digium IP Phones - Teleworker Capability?

[asterisk users]

On Thu, Jun 14, 2012 at 4:05 PM, Kevin P. Fleming <kpfleming at digium.com> wrote:
> On 06/14/2012 04:57 PM, asterisk users wrote:
>>
>> We couldn't see anything about this on the Digium site, but maybe
>> someone here can comment?
>>
>> Do the new Digium phones provide good "teleworker" functionality?
>
>
> Yes, I believe they do :-)
>
>
>> The benchmark we're comparing against is the capabilities of Mitel
>> 3300 IP systems  with Mitel 5330 IP phones (running their proprietary
>> MINET protocol), specifically:
>>
>> a. A Mitel phone can be easily configured for teleworker mode (select
>> TW mode and the IP of the gateway server).  The phone reboots and it
>> is ready to be used (once the Mitel border gateway is set to recognize
>> the unit's ID, based on its MAC address, printed on the label on the
>> back of the phone).  If the phone gets reallocated back to a directly
>> connected office environment, a simple reset procedure brings it back.
>
>
> Digium phones can do something similar, and in an upcoming firmware release,
> there will even be features available to make this happen on a fairly
> automatic basis.
>
>
>> b. You can plug in the phone virtually anywhere. It has a built-in
>> tunnelling mechanism providing end-to-end encryption and is very
>> tolerant of the network configuration, routers, NAT, etc.
>
>
> Digium phones speak SIP and RTP to the server, just like pretty much any
> other SIP phone. They employ many modern NAT traversal techniques and should
> work in most network situations. They don't currently provide encryption for
> signaling and media, though.
>
>
>> c. If the link between the phone and the gateway goes down, the phone
>> will restore itself gracefully and automatically once the network
>> function resumes.  Absolutely hassle-free to the user.
>
>
> I don't understand this; SIP phones don't require this at all. The phone is
> an intelligent device on its own. If there is no network connectivity to the
> server, then calls cannot be placed or received, but once connectivity is
> restored, operation would be back to normal.
>
>
>> d. Users can be configured to have hot-desk functionality.  The phone
>> has a default extension assigned, but the user can be set up so that
>> they can "log in" to their normal office extension number from
>> wherever they are.  Their office phone is automatically logged-out and
>> goes to its default extension when you log in to a teleworker phone
>> (you don't have to log out from it first).  Your phone buttons,
>> display settings, voicemail WMI and access, (everything) move to this
>> new phone, and you can work from your home office, on the road, etc.,
>> and inbound and outbound calls work just like you were there in the
>> office (callerid, etc).
>
>
> Yes, this is supported.
>
>
>> These four features would be a big selling point for us to consider
>> moving our organization from Mitel to Digium/Asterisk/Switchvox.
>>
>> How much of this can be done with Asterisk/Switchvox and, say, the
>> Digium D70 phone with dynamic button display?
>
>
> Most of it, I think. Give them a try!
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
> --


This is pretty good news, overall. To comment on Kevin's points:

- The end-to-end encryption is important to us, because
client-ID-sensitive information is part of our environment.  Something
like built-in OpenVPN would work for us, if that were an option.

- Being fault-tolerant (of less than perfect DSL and rural-wireless
connections - if the boss is at his cabin, for instance) and being
very user-friendly about it is really important to end users.  Minet
has a heart-beat mechanism so that if the connection goes down between
the phone and the switch, the display shows it.  Of course, calls get
diverted to voicemail during that period.

If something is not working in the network, the user is informed about
it, and when it is fixed, everything continues, including button DSS
status updates, voicemail WMI, etc.

On typical SIP phones, everything looks normal until you go to use it,
then there is no dialtone, or you just get dead-air on the handset).

Our users are pretty demanding, and want a utility-grade solution that
will always work - for them.

- > Most of it, I think. Give them a try!

Is there a detailed application note in the Digium wiki (or anywhere
else for that matter) about these implementing features under
Asterisk/Switchvox?

Thanks!