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

Jeff LaCoursiere jeff at sunfone.com
Thu Jun 14 17:45:15 CDT 2012


On Thu, 2012-06-14 at 16:23 -0600, asterisk users wrote:

> 
> 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.
> 

Yealink and I think Aastra phones have OpenVPN built in.  We use Yealink
with layer 2 tunnels such that the phones have the same configuration,
network wise, wherever they happen to be plugged in.  No NAT issues
ever.

> - 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.
> 

Pretty much all SIP phones work that way.

> 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.
> 

Again all phones work that way.

> 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).
> 

Which SIP phone have you been using?  The ones we are familiar with -
Polycom, Linksys, Yealink, Snom, Aastra, Grandstream - all show you when
the network link is down, and all services return as soon as it comes
back up.  Even Linksys ATAs at least show you an LED of when the device
is registered, though you will just get dead air if you pick up 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?
> 

You could probably find 50 people to help you set such a system up on
this list (or more appropriately on -biz).

Cheers,

j





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