Re: [asterisk-users] Digium IP Phones UNREACHABLE after registration

[Antony Stone]

On Thursday 12 April 2018 at 18:17:10, Hermann Wecke wrote:

> On Thu, Apr 12, 2018 at 11:43 AM, Antony Stone wrote:
> > 
> > Are you by any chance running fail2ban, without the IP address of this
> > location in a whitelist?
> 
> fail2ban: yes
> whitelist: yes

Have you checked:

a) the fail2ban logs
and
b) the firewall rules in place on the server immediately after the phones get 
kicked off

just to be sure this is not being caused by fail2ban somehow?

> > I'm wondering if some device is misconfigured and failing registration,
> > which get spicked up by fail2ban, and the network's public IP gets
> > blocked.
> 
> If I remove the Digium phones, after a while the other devices
> (brands) will register again.

What happens if you then add just one Digium phone?

Does it, and all the other brands, work as normal?

If yes, try adding one more, then one more - maybe you'll find one specific 
phone which is causing this problem.

If adding just one phone causes the problem immediately, try adding a different 
one instead (in case the first one you added is the one with the problem).

> >> The server remains operational and all other users/peers (not running
> >> Digium phones) are up and running.
> > 
> > You mean, all others at other locations, right?
> 
> Yes, other locations are OK. This location will be "clogged" (or
> "flooded"?) and unavailable.

Can you run wireshark or similar on the router / firewall at this location, to 
see what SIP traffic you get going back and forth to the server at the time the 
problem occurs?


Antony.

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Re: [asterisk-users] Digium IP Phones UNREACHABLE after registration

[Hermann Wecke]

On Thu, Apr 12, 2018 at 11:43 AM, Antony Stone
 wrote:
>> A few seconds after registration, the Digium phones will become
>> UNREACHABLE. Right after that, the entire VoIP network (where the
>> Digiums are located) will be also dropped - all other devices
>> (non-Digium) connected will be kicked from the asterisk box. There are
>> ObiHai, Yealink and Linksys at this location - all will be kicked.
>
> Are you by any chance running fail2ban, without the IP address of this
> location in a whitelist?

fail2ban: yes
whitelist: yes

> I'm wondering if some device is misconfigured and failing registration, which
> get spicked up by fail2ban, and the network's public IP gets blocked.

If I remove the Digium phones, after a while the other devices
(brands) will register again.

>> The server remains operational and all other users/peers (not running
>> Digium phones) are up and running.
>
> You mean, all others at other locations, right?

Yes, other locations are OK. This location will be "clogged" (or
"flooded"?) and unavailable.

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Re: [asterisk-users] Digium IP Phones UNREACHABLE after registration

[Administrator TOOTAI]

Hi Herrmann

Le 12/04/2018 à 17:22, Hermann Wecke a écrit :

I'm trying to solve a mystery for the last couple of days.

I have a mix of D70, D50 and D40 behind NAT. Server is in a
colocation, not a VPS.

For several years, everything was working fine, no issues. A few days
ago I started having problems at one particular site. NO CHANGES have
been made to this office network - same router, switch and internet
provider. No new equipment added or configuration changed (I only
upgraded the firmware and asterisk trying to solve the problem).

A few seconds after registration, the Digium phones will become
UNREACHABLE. Right after that, the entire VoIP network (where the
Digiums are located) will be also dropped - all other devices
(non-Digium) connected will be kicked from the asterisk box. There are
ObiHai, Yealink and Linksys at this location - all will be kicked.

The server remains operational and all other users/peers (not running
Digium phones) are up and running.


Don quite understand: above you say that all others phones are kicked 
too and here you say they are up and running ... Also, UNREACHABLE and 
kick are not the same for me.




Sip debug and tcpdump didn't show any relevant information to solve
the puzzle.


What is relevant: do you see SIP traffic coming from those 
phones/location *AFTER* they are UNREACHABLE/kicked?


Is a fail2ban involved ?

[...]

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Daniel

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Re: [asterisk-users] Digium IP Phones UNREACHABLE after registration

[Antony Stone]

On Thursday 12 April 2018 at 17:22:39, Hermann Wecke wrote:

> I'm trying to solve a mystery for the last couple of days.
> 
> I have a mix of D70, D50 and D40 behind NAT. Server is in a
> colocation, not a VPS.
> 
> For several years, everything was working fine, no issues. A few days
> ago I started having problems at one particular site. NO CHANGES have
> been made to this office network - same router, switch and internet
> provider. No new equipment added or configuration changed (I only
> upgraded the firmware and asterisk trying to solve the problem).
> 
> A few seconds after registration, the Digium phones will become
> UNREACHABLE. Right after that, the entire VoIP network (where the
> Digiums are located) will be also dropped - all other devices
> (non-Digium) connected will be kicked from the asterisk box. There are
> ObiHai, Yealink and Linksys at this location - all will be kicked.

Are you by any chance running fail2ban, without the IP address of this 
location in a whitelist?

I'm wondering if some device is misconfigured and failing registration, which 
get spicked up by fail2ban, and the network's public IP gets blocked.

> The server remains operational and all other users/peers (not running
> Digium phones) are up and running.

You mean, all others at other locations, right?


Antony.

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[asterisk-users] Digium IP Phones UNREACHABLE after registration

[Hermann Wecke]

I'm trying to solve a mystery for the last couple of days.

I have a mix of D70, D50 and D40 behind NAT. Server is in a
colocation, not a VPS.

For several years, everything was working fine, no issues. A few days
ago I started having problems at one particular site. NO CHANGES have
been made to this office network - same router, switch and internet
provider. No new equipment added or configuration changed (I only
upgraded the firmware and asterisk trying to solve the problem).

A few seconds after registration, the Digium phones will become
UNREACHABLE. Right after that, the entire VoIP network (where the
Digiums are located) will be also dropped - all other devices
(non-Digium) connected will be kicked from the asterisk box. There are
ObiHai, Yealink and Linksys at this location - all will be kicked.

The server remains operational and all other users/peers (not running
Digium phones) are up and running.

Sip debug and tcpdump didn't show any relevant information to solve
the puzzle. I also replaced the router (twice, different models and
firmware versions), PoE switch and cable modem. No success. IP is
dynamic but the provider will only change it once a year. Qualify=yes
(or no) didn't fix. Removing the password (deny/permit IP) also
didn't.

Running Asterisk 13.20.0, firmware 2.6.2.

Any ideas where I should dig further?

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Re: [asterisk-users] Digium IP Phones D40

[Kevin P. Fleming]

On 06/22/2012 05:12 PM, bilal ghayyad wrote:

One of the problems I faced with Polycom is the voice volume and ring volume, 
it is low.

When it rings, even if it is maximum volume, still it is weak.
When I talk and I set the volume to the maximum, I still feel the voice volume 
is low and would if to increase it.


I have never, in over 7 years of using Polycom phones, heard anyone 
complain that the maximum volume was too low. Most devices of this type 
have their maximum volume controlled to meet guidelines set by 
government and industry recommendations (in order to avoid causing 
damage to users' ears), and the Digium phones are no exception.


If you are finding that the volume produced by common SIP phones is too 
low and you can't make it loud enough, I'd bet that the problem is not 
in the phones, but in your environment or your ears :-)


--
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org



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Re: [asterisk-users] Digium IP Phones D40

[Jeff LaCoursiere]

On Mon, 2012-06-25 at 10:45 -0500, Kevin P. Fleming wrote:
 On 06/22/2012 05:12 PM, bilal ghayyad wrote:
  One of the problems I faced with Polycom is the voice volume and ring 
  volume, it is low.
 
  When it rings, even if it is maximum volume, still it is weak.
  When I talk and I set the volume to the maximum, I still feel the voice 
  volume is low and would if to increase it.
 
 I have never, in over 7 years of using Polycom phones, heard anyone 
 complain that the maximum volume was too low. Most devices of this type 
 have their maximum volume controlled to meet guidelines set by 
 government and industry recommendations (in order to avoid causing 
 damage to users' ears), and the Digium phones are no exception.
 
 If you are finding that the volume produced by common SIP phones is too 
 low and you can't make it loud enough, I'd bet that the problem is not 
 in the phones, but in your environment or your ears :-)
 

Actually we get that complaint a lot too (Polycom ring volume).  We
typically install in hotel environments, and in their back office the
environment can be noisy, as well as in their restaurants.

I imagine in a typical office environment this wouldn't be an issue...

j



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Re: [asterisk-users] Digium IP Phones D40

[Steven Howes]

On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote:
 Actually we get that complaint a lot too (Polycom ring volume).  We
 typically install in hotel environments, and in their back office the
 environment can be noisy, as well as in their restaurants.
 
 I imagine in a typical office environment this wouldn't be an issue...

The old Cisco/Linksys SPA9xx series used to be ear-bleedinly loud. The newer 
SPA5xx series have been seriously toned down. I'm guessing it's all 'safety' 
related :S

Steve
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Re: [asterisk-users] Digium IP Phones D40

[Bryant Zimmerman]

We have the ringer volume issue with some customer environments as well. We 
use Grandstream phones in a lot of installs so we just upload a custom 
ringtone with the db pushed up on it a bit. 
We are testing the Digium phones and have concerns if we will be able to 
use them for the high noise env customers. Polycom phones do have the same 
ring volume issue for these customers.
No issues in general office env. 

Bryant


 From: Steven Howes steve-li...@geekinter.net
Sent: Monday, June 25, 2012 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Digium IP Phones D40

On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote:
 Actually we get that complaint a lot too (Polycom ring volume). We
 typically install in hotel environments, and in their back office the
 environment can be noisy, as well as in their restaurants.
 
 I imagine in a typical office environment this wouldn't be an issue...

The old Cisco/Linksys SPA9xx series used to be ear-bleedinly loud. The 
newer SPA5xx series have been seriously toned down. I'm guessing it's all 
'safety' related :S

Steve
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Re: [asterisk-users] Digium IP Phones D40

[Tim Nelson]

- Original Message - 
 We have the ringer volume issue with some customer environments as
 well. We use Grandstream phones in a lot of installs so we just
 upload a custom ringtone with the db pushed up on it a bit.
 We are testing the Digium phones and have concerns if we will be able
 to use them for the high noise env customers. Polycom phones do have
 the same ring volume issue for these customers.
 No issues in general office env.


For everyone complaining about the Polycom's lack of volume, are you simply 
hitting the volume buttons, or are you also aware of the myriad of adjustments 
available in the Polycom XML provisioning configs? We have a local educational 
customer that experienced volume problems with Polycom due to noisy classroom 
environments, and with a few tweaks to volume and gain in the XML configs 
pushed via TFTP, the phones were ear-splittingly loud, both ringers and 
handset.

--Tim

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Re: [asterisk-users] Digium IP Phones D40

[bilal ghayyad]

Dears;

One of the problems I faced with Polycom is the voice volume and ring volume, 
it is low.

When it rings, even if it is maximum volume, still it is weak.
When I talk and I set the volume to the maximum, I still feel the voice volume 
is low and would if to increase it.

The volume is really very important (before we talk about features and 
provisioning), so if the Difium IP Phones have a good voice volume and ring 
volume (not the same problem that the Polycom has) then I am surely into 
selecting Digium.

Looking to hear from you.
Regards
Bilal

-

  Hi All;
 
  Any one used Digium IP Phones D40?
 
  I need to know if they are stable with good voice
 quality? Comparing to Polycom 330, which is better? Let us
 talk frankly although I know that we have to support
 Digium.
 
 I don't think these topics about comparing A to B work very
 well. For 
 me, it comes down to what has worked well in the past. 
 With that in 
 mind, it will be hard for us to give up Polycom phones.
 
 I had the ability to test the Digium phones while at Digium,
 and they 
 are rugged phones, they look professional too. 
 However, I only tested 
 with them for a month or so.
 
 Now the Digium phones have a tighter integration out of the
 box with 
 Asterisk / Switchvox however that is not an added feature
 since we 
 already have provisioning modules for Polycom phones. If you
 have never 
 mass deployed Polycom phones, it does require some
 work.  You need to 
 get your hands dirty but Polycom has a lot of documentation
 about the 
 process.
 
 With Digium, they take this point of pain away from
 you.  A ship the 
 phones with a tight integration with asterisk / Switchvox.
 There are 
 some other features about visual voicemail and JS
 applications, however 
 I don't require them so not a feature I am interested in.
 
 So, to answer your questions, compare Polycom to Digium. For
 me, the 
 winner is Polycom because their phones have been around for
 years. 
 Digium's only months. And because we have Polycom phones at
 95% of the 
 sites we manage, adding another vendor into the mix for use
 to support 
 does not make sense at this time.

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

[Kevin P. Fleming]

On 06/14/2012 05:23 PM, asterisk users wrote:

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


Not yet, I don't believe.

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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

[asterisk users]

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?

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.

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.

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.

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).

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?

Thanks for all comments!

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

[Kevin P. Fleming]

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: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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

[asterisk users]

On Thu, Jun 14, 2012 at 4:05 PM, Kevin P. Fleming kpflem...@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: kflem...@digium.com | SIP: kpflem...@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!

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

[Jeff LaCoursiere]

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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[asterisk-users] Digium IP Phones D40

[bilal ghayyad]

Hi All;

Any one used Digium IP Phones D40? 

I need to know if they are stable with good voice quality? Comparing to Polycom 
330, which is better? Let us talk frankly although I know that we have to 
support Digium.

Regards
Bilal

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Re: [asterisk-users] Digium IP Phones D40

[Carlos Alvarez]

On Mon, Jun 11, 2012 at 9:58 AM, bilal ghayyad bilmar...@yahoo.com wrote:

 I need to know if they are stable with good voice quality? Comparing to
 Polycom 330, which is better? Let us talk frankly although I know that we
 have to support Digium.


Voice quality is great.  I would choose the Digium phones over a Polycom
every time, that's an easy choice.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Digium IP Phones D40

[Paul Belanger]

On 12-06-11 12:58 PM, bilal ghayyad wrote:

Hi All;

Any one used Digium IP Phones D40?

I need to know if they are stable with good voice quality? Comparing to Polycom 
330, which is better? Let us talk frankly although I know that we have to 
support Digium.

I don't think these topics about comparing A to B work very well. For 
me, it comes down to what has worked well in the past.  With that in 
mind, it will be hard for us to give up Polycom phones.


I had the ability to test the Digium phones while at Digium, and they 
are rugged phones, they look professional too.  However, I only tested 
with them for a month or so.


Now the Digium phones have a tighter integration out of the box with 
Asterisk / Switchvox however that is not an added feature since we 
already have provisioning modules for Polycom phones. If you have never 
mass deployed Polycom phones, it does require some work.  You need to 
get your hands dirty but Polycom has a lot of documentation about the 
process.


With Digium, they take this point of pain away from you.  A ship the 
phones with a tight integration with asterisk / Switchvox. There are 
some other features about visual voicemail and JS applications, however 
I don't require them so not a feature I am interested in.


So, to answer your questions, compare Polycom to Digium. For me, the 
winner is Polycom because their phones have been around for years. 
Digium's only months. And because we have Polycom phones at 95% of the 
sites we manage, adding another vendor into the mix for use to support 
does not make sense at this time.


--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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Re: [asterisk-users] Digium IP Phones

[Kevin P. Fleming]

On 05/11/2012 11:08 PM, Danny Dias wrote:

Does the D40 will support the option to develope apps? As i could see on
videos only the D70 has the apps button, and also, the lcd screen is
smaller. Right?


All of the Digium IP phones will support user-developed applications 
once the SDK has been released. The D40 and D50 have the same size 
screen, but it is smaller than the one on the D70. The D40 and D50 do 
not have a hard 'Apps' button, but they do have an on-screen softkey for 
access to applications.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Digium IP Phones

[Danny Dias]

Does the D40 will support the option to develope apps? As i could see on
videos only the D70 has the apps button, and also, the lcd screen is
smaller. Right?

Enviado desde mi Samsung Galaxy S II
El 10/05/2012 12:44, Kevin P. Fleming kpflem...@digium.com escribió:

 On 05/09/2012 08:38 PM, Danny Dias wrote:

 Hello,

 Im looking to buy a digium phone D70 unit just for testing on lab; to
 really understand the phone and features.

 I cant find any website with opinions; any here? Are they really
 valuable to the price? (D70 quite expensive)

 Does the SDK for building apps is usable? Can you build powerfull apps?
 Examples?


 The phone app SDK has not been released yet, it's still under development.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org

 --
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Re: [asterisk-users] Digium IP Phones

[Dennis Dryden]

Hello,
I've been using the Digium D40's for a few weeks now and I think they
are good for the price. There are a few UI problems but I hope/expect
they will be resolved in a firmware update or two.

Haven't looked at the SDK yet.

Thanks,
Dennis



On Thu, May 10, 2012 at 2:38 AM, Danny Dias ing.diasda...@gmail.com wrote:
 Hello,

 Im looking to buy a digium phone D70 unit just for testing on lab; to really
 understand the phone and features.

 I cant find any website with opinions; any here? Are they really valuable to
 the price? (D70 quite expensive)

 Does the SDK for building apps is usable? Can you build powerfull apps?
 Examples?

 Many thanks


 --
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Re: [asterisk-users] Digium IP Phones

[Arthur Stanfield]

We've just had one of each delivered for us to play with in our lab (Literally 
an hour ago!). Not had chance to play with them yet, But initial thoughts are 
they look good. Build quality seems fine for the price. I'll form more of an 
opinion when i get chance to play with them properly tomorrow. 

I don't think the SDK is available yet (I've not been able to find it on the 
digium site). I'm itching to get my hands on it though! My first thought when 
seeing the D70 and looking at the screen for the speed dial keys was I hope we 
can use this screen in for the apps, It's perfect for a tetris clone. :)

Cheers,
AJ.

- Original Message -
From: Danny Dias ing.diasda...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, 10 May, 2012 2:38:02 AM
Subject: [asterisk-users] Digium IP Phones




Hello, 

Im looking to buy a digium phone D70 unit just for testing on lab; to really 
understand the phone and features. 

I cant find any website with opinions; any here? Are they really valuable to 
the price? (D70 quite expensive) 

Does the SDK for building apps is usable? Can you build powerfull apps? 
Examples? 

Many thanks 
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Re: [asterisk-users] Digium IP Phones

[Kevin P. Fleming]

On 05/09/2012 08:38 PM, Danny Dias wrote:

Hello,

Im looking to buy a digium phone D70 unit just for testing on lab; to
really understand the phone and features.

I cant find any website with opinions; any here? Are they really
valuable to the price? (D70 quite expensive)

Does the SDK for building apps is usable? Can you build powerfull apps?
Examples?


The phone app SDK has not been released yet, it's still under development.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] Digium IP Phones

[Danny Dias]

Hello,

Im looking to buy a digium phone D70 unit just for testing on lab; to
really understand the phone and features.

I cant find any website with opinions; any here? Are they really valuable
to the price? (D70 quite expensive)

Does the SDK for building apps is usable? Can you build powerfull apps?
Examples?

Many thanks
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