I think you would only need a headset if you need privacy. That is the
catch-22 in healthcare environments -- HIPAA would prevent Dr. Smith calling
nurse Jones to ask the temperature of patient Susan White because nurse
Jones could be with another patient.
Support for the Vocera server to
I've worked with these before. They are designed to run a whole hospital
shift, so there should be no worries regarding the battery.
I'm not aware of the server having any kind of SIP support -- I think you
would need to have a PRI trunk to another PBX. The last time I talked to
them they had
Along these lines, are there any services that send an e-mail/SMS when there
is a tornado/thunderstorm warning or watch in one or more ZIP codes?
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann
Please take note of their posting:
https://aws.amazon.com/security/
which discusses the issue and what they're doing to improve response.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred
...
On Apr 20, 2010, at 6:18 PM, Frank Bulk wrote:
Please take note of their posting:
https://aws.amazon.com/security/
which discusses the issue and what they're doing to improve response.
Frank
If only they wrote the truth...
When we find misuse, we take action quickly and shut it down
Patrick
On Sat, Oct 10, 2009 at 06:57, Frank Bulk frnk...@iname.com wrote:
There are two commercial vendors that come to mind, namely DiVitas and
Agito.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
There are two commercial vendors that come to mind, namely DiVitas and
Agito.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick
Sent: Friday, October 09, 2009 8:15 PM
To: Asterisk Users Mailing
Depending on the latency, wrapping the UDP stream into a TCP-based tunnel
can be good -- if the VPN tunnel occasionally drops a packet, the tunnel
will re-transmit the UDP packet. Of course, if the (one-way) latency is too
high, the re-transmitted payload will arrive outside the jitter buffer and
Resending is not a waste if the re-transmitted packet can arrive within the
jitter buffer window. Practically speaking, though, since UDP packets are
generally not retransmitted (unless it's within some kind of TCP-based
tunnel), it's a moot point.
Frank
-Original Message-
From:
Note to those Americans scratching their heads over this: nano-BTS systems
are not so unusual in the Netherlands, unlike the USA.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent:
If the users' understanding of jitter is technically correct, and they're
complaining about quality issues (due to jitter or packet loss), then
lowering the jitter buffer isn't going to help.
An ADSL link, depending on the sync rate, can have 40+ msec of latency
between the DSL modem and DSLAM.
There's a bit of oversimplification going on here -- it's not a ...
database. Different CNAM providers have different databases which are
populated from many sources. Most of the data probably matches, but not all
of it.
If the Calling Name is incorrect, the person who received the call will
, and with the
exception of this issue, all is working well. But, my client is really
unhappy that their callerID NAME isn't showing up.
On Tue, Jul 7, 2009 at 3:42 PM, Frank Bulk frnk...@iname.com wrote:
There's a bit of oversimplification going on here -- it's not a ...
database. Different CNAM
Intersting. Vitelity is charging for something that they might already be
getting paid for. Of course, updating a name for a number takes time, and
so that's probably why they can justify charging the customer something.
Most times when you sign up you specify how you want the directory listing
For the CNAM vendors who pride themselves on completeness/coverage, don't
you think that they have some interest in getting data from the likes of
Teliax? Maybe they wouldn't pay for it, but ITSPs have to realize that to
retain certain customers that they have to their customers numbers
How does that work?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Tuesday, July 07, 2009 8:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
/downloads/elastix/Elastix_HA_Cluster.pdf
- Original Message -
From: Frank Bulk frnk...@iname.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 10, 2009 4:22:31 PM GMT -05:00 US/Canada Eastern
Subject: Re: [asterisk-users
Have you tried TCPMP 0.72RC1? It's free and plays lots of formats,
including a WAV file with a codec type that our voicemail system generates
but that Windows Media Player does not.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
It's not clear where the HA comes in. Can you explain?
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw
Sent: Wednesday, June 10, 2009 8:19 AM
To: Asterisk Users Mailing List -
It's been a few years ago, but Network Computing had tests results showing
that VoIP over a VPN was measurably better than outside a VPN. Why?
Because the latency was low enough that lost UDP packets (within the VPN
tunnel) could be re-transmitted before the jitter buffer had expired. Since
most
If people don't mind taking turns talking, it will work. It's just going
to be like talking on a CB. Reminds me of talking to my grandparents in the
Europe as a child in the early 80's.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
In a SOHO environment I would agree with you, but not if your coverage area
needs to be tens of thousands of square feet. Deploying a complete overlay
wireless infrastructure doesn't make sense and is another infrastructure to
manage and maintain.
Frank
-Original Message-
From:
Two of the wireless carriers have a Centrex-like solution:
http://www.networkcomputing.com/channels/wireless/showArticle.jhtml?articleI
D=202200832pgno=5
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Look it up at www.localcallingguide.com. The name on record there may be
the LECs name -- then you need to find a SIP provider or DID handler that
has a business relationship with that LEC. For example, in the state of
Iowa Vonage obtained (at least some of) it's numbering resources from
It all has to do with interconnection agreements with the ILEC and if the
reseller has numbering resources in the requested area. Looks like
BroadVoice does have all those elements taken care of.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Just google 2950 and rate-limit and you'll see that it's possible to do so
with the EI immagebut in 1 Mbps increments.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday,
Not in the form factor that you would expect.
Can I ask why? Most modern VoFi phones support WPA2.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Wednesday, February 11, 2009
BRI is not even supported on Nortel's rural market softswitch, the CS-1500.
BRI is a dead horse..
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Higgins
Sent: Tuesday, January 27, 2009 11:50 AM
At least for this telco, ISN and BRI was a money-loser. We've spent more
time trouble-shooting those connections (on behalf of the customers) than we
ever made in monthly or per-minute charges.
Frank
From: asterisk-users-boun...@lists.digium.com
Tim:
Are you referring to the older-style cable telephony where they had an
analog carrier on the cable plant, or PacketCable VoIP?
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent:
Is it possible to list multiple hosts, separating them with ampersands?
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rob Hillis
Sent: Sunday, January 11, 2009 4:32 PM
To: Asterisk Users Mailing List -
fromhost=acme.com/sip.acme.com
2009/1/6 Frank Bulk frnk...@iname.com
I tried that before, but I just tried it again. Unfortunately, the same
thing:
No user '5551236049' in SIP users list
Found peer 'ACME' for '5551236049' from 172.16.10.40:5060
[ACME]
host=172.16.10.40
username=username
You're the miracle worker! Thanks!
Frank
From: Andres [mailto:and...@telesip.net]
Sent: Tuesday, January 06, 2009 11:19 AM
To: Frank Bulk
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work
unless I add
password
CLIM 7
CPBY 0
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Bulk -
iName.com
Sent: Monday, January 05, 2009 6:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not
work unless I add insecure=very to my Outgoing settings, but I don't
want to do that. I do want to authenticate. Outgoing (Asterisk PBX to
Class 5 switch) calls do authenticate and work.
The Nortel CS 1500 I'm using as
?
Keep in mind the domain is part of the digest authentication process and
is a factor in the encoding of the nonce.
Frank Bulk - iName.com wrote:
The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not
work unless I add insecure=very to my Outgoing settings, but I don't
] Incoming side of SIP trunk does not work
unless I add insecure=very
Frank Bulk - iName.com wrote:
The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not
work unless I add insecure=very to my Outgoing settings, but I don't
want to do that. I do want to authenticate. Outgoing
it by IP address instead of hostname as reverse DNS may not be
resolving. e.g. host=123.123.123.123
On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk frnk...@iname.com wrote:
This is what I have in my configuration now:
[ACME]
host=sip.acme.com
username=username
secret=password
type=friend
I've done a SIP
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