Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
Michael Graves
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Original Message
Subject: [asterisk-users] OT - Which Android handset with Wifi-only ?
From
Actually, all of the conference phones are known by the SoundStation
name and the desk phones are SoundPoint.
Michael Graves
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Original Message
Subject: Re: [asterisk-users
/670 models.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
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Original Message
Subject: Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD
voice codecs?
From: Steve Underwood ste...@coppice.org
Date: Wed
-soundpoint-ip335-entry-level-hdvoice-ip-phone/
Michael Graves
mgraves mstvp.com
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Original Message
Subject: Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
From: Andy Graybeal andy.grayb
-appliances-a-question-of-scale/
Michael Graves
mgraves mstvp.com
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Original Message
Subject: Re: [asterisk-users] Pbx_för_Windows?_-_Email_found_
in_subject
From: Doug Lytle supp...@drdos.info
Date: Fri
/
Michael Graves
mgraves mstvp.com
o(713) 861-4005
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Original Message
Subject: [asterisk-users] Dual Atom mobo - call capacity
From: Michelle Dupuis mdup...@ocg.ca
Date: Thu, June 10, 2010 7:19 pm
To: Asterisk Users
I am also very interested in support for Jack in recent release. I
envision a little project using Jack to route call audio into a digital
audio workstation (Neundo or Pro Tools) for real-time processing using
VST plug-ins.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
phone implemented as a web
plug-in or activex object. Tim Panton of PhoneFromHere.com has a great
Java soft phone object that we use to make G.722 calls to the ZipDX
conference bridge for the VoIP Users Conference every week.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
I had a good experience with that Polycom/Spectralink phone. Very rugged
as you say. The experience did highlight the weaknesses in consumer
Wifi AP, which reinforced my commitment to continue using DECT around my
office.
Michael
Original Message
Subject: Re:
.
Michael Graves
mgraves at mstvp.com
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c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
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FWD 54245
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Doing a little research before Friday's Voip Users Conference call with
Dan Behringer.
Are any of the newer Polycom wideband codecs implemented in v1.6?
Specifically, G.722.1 or G.722.2?
Thanks,
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
.
Michael Graves
mgraves at mstvp.com
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c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
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FWD 54245
Original Message
Subject: Re: [asterisk-users] Polycom wideband codecs?
From: randulo spamsucks2...@gmail.com
Date: Tue, April 21, 2009 1:40 pm
Yesterday I blogged a post about some ideas that I think will help
Asterisk appliances further penetrate SMB/SOHO sites in ways that are
not presently being addressed.
http://blog.mgraves.org/2008/08/20/a-suggestion-to-asterisk-appliance-developers/
Michael Graves
mgraves at mstvp.com
o(713) 861
You're not kidding. I would have to investigate cheaper routing.
Truncating my wife's calls would be met with harsh reaction at best.
Maybe painful, too.
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
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FWD 54245
Original Message
On Thu, 07 Aug 2008 00:28:29 -0500, Karl Fife wrote:
Example: Last week there was talk about Polycom's HDVoice
technology, and the term was being used interchangeably with G.722. In
fact there are important distinctions, but someone listening might
presume that the information was correct and
Dan York gave a security presentation at Astricon. I've heard the
recording he made of that session but it has yet to be published. He may
be available, as least as a representative of VOIPSA.
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
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Headset mic? Drive safe ;-)
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
Original Message
Subject: Re: [asterisk-users] The S word: Asterisk security
From: Fred Posner [EMAIL PROTECTED]
Date: Tue, July
The quad-band model is around $250 USD.
See Ebay auction here http://tinyurl.com/5tvoa9
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
Original Message
Subject: Re: [asterisk-users] Asterisk GSM Gateway
My employer has recently moved from a Checkpoint firewall to MS ISA, or
so I'm told. Does anyone have and advice on configuring this to pass SIP
to/from a hard phone inside the LAN? They have one Polycom IP430 that
they need to register with an external hosted provider.
Michael Graves
mgraves
What is the logic of them using SIP over TCP? Is this a broad industry
trend? Or just the latest attempt to get around SIP/NAT issues?
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
Original Message
Subject
or
Sangoma hardware is not a problem. And, after all, you get what you pay
for in most cases.
Michael Graves
mgraves at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
Original Message
Subject: Re: [asterisk-users] [Zaptel] Why no port to Windos
That Mitel phone looks interesting. I'm not familiar with others like
it, except for one from a Far East source that looked a little sketchy.
It's worth noting that Linksys sells a Wifi bridge adapter specifically
for their phones. It gets power from the phone and lets any of their
phones become
Original Message
From: [EMAIL PROTECTED] (Tony Mountifield)
Date: Fri, October 05, 2007 4:05 am
To: asterisk-users@lists.digium.com
In article [EMAIL PROTECTED],
C F [EMAIL PROTECTED] wrote:
If you want this to work nicely dont settle for anything else than a
channel
Sorry for being a little off topic, but I'mconsidering a few new phones
for my Asterisk installation. I have a mix of Polycom 500/600s and an
Aastra 480i CT. I'm considering adding a couple of Aastra 57i or 57i
CT.
Does anyone here have experience with the 480i CT and the newer 57i CT?
I'm
For all the noise about this noone has mentioned one important thing. We should be gratefull that we have access to G.729a in Asterisk, whatever the mechanics of the licensing. It's obvious that its a pain in the [EMAIL PROTECTED] for Digium who absolutely not making ANY on it money for their
Also...an article I wrote earlier this year
http://www.tomsnetworking.com/2006/01/13/how_to_asterisk_pbx/
Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
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c(713) 201-1262
Original Message
Subject:
Original Message
Skype uses iLBC codec, which has great jitter compensation. IIRC, the
newer SIP channels of * are supposed to have the same capabilities, but
I have not tested. I really do not like Skype (prefer FWD), but I must
say, over satellite, etc, they provide
What would be ideal is the introduction of an open source wideband codec
implementation. Then you could see it adopted into SIP end points and
used with SER realtively quickly. Sadly, an Asterisk implmentation
would lag a little behind due to the amount of work required in an
implementation that
I bought one of these. It's a great device. So good that I gave it to my
boss to use with Skype. It's far better than the speakerphone in the
Alcatel phone on his desk. We've used it with Skype and Gizmo.
Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
I would've expected a channel bank or two to be the definitive solution.
With a T-1 connection or two back to the * server.
Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
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c(713) 201-1262
Original Message
Original Message
Subject: Re: [Asterisk-Users] Asterisk on laptop connected to POTS line
From: Tzafrir Cohen [EMAIL PROTECTED]
Date: Thu, February 02, 2006 9:15 am
To: asterisk-users@lists.digium.com
On Thu, Feb 02, 2006 at 09:13:29AM -0500, Alexander Lopez wrote:
I've had the 480 CTi for about 6 months. It's a great phone. It
displaced a Polycom IP600 on my desk. The Polycom phone was also a
great phone, but the 480 CTi is a bit easier to configure use...also
the cordless handsets are a big improvement over the Hitachi WIP5000
wifi phones that I have also
Manny,
You really need to try Astlinux. See www.astlinux.org. It does pretty
much what you desire.
Also see my recent article about Aslinux embedded on a Soekris Net4801
(http://www.tomsnetworking.com/Sections-article153.php)
Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL
Original Message
Subject: [Asterisk-Users] Choosing an FXO card
From: Mike Hemstock [EMAIL PROTECTED]
Date: Sun, January 15, 2006 6:18 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Folks,
I'm looking at buying an FXO
Original Message
Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *?
From: Asterisk-User [EMAIL PROTECTED]
Date: Tue, January 10, 2006 5:23 am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Has anyone tried
Hi All,
Has anyone on-list tried using a USB style adapter like the VTA-1000 to
provide some form of gateway from Skype into *? If so, how well did it
work? Would I need to combine the VTA-1000, which appears to be
USB-to-RJ11 and a standard FXO port?
Would it be possible to provide a more
No problems with US calls. No calls going through to UK though. My
account login on the website worked this morning.
Michael
Original Message
Subject: RE: [Asterisk-Users] VoipJet Problems - anyone?
From: Innocent Evil [EMAIL PROTECTED]
Date: Thu, August 18, 2005 3:32 pm
Original Message
Subject: RE: [Asterisk-Users] will a firewall slow down asterisk?
From: Wiley Siler [EMAIL PROTECTED]
Date: Wed, August 10, 2005 11:04 am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
That should not
I have one DID and make outgoing calls through Sixtel. Their support has
been good with the minor issues I've seen. The're very * aware. I also
use Voipjet for outbound and Clearpath for an 800 number.
All provide IAX termination. All three companies have been great
compared to my prior
Hi All,
I'd really like to see a show of hands with regard to people's
experience with FXO interfaces. I own a few X100p cards and have had
nothing but problems with them.
I also took part in Sipura's beta program, for the SPA-3000. While it
can be an improvement over the X100p, it presently
Hello,
I've just set up DISA on my * server. I'm using it to avoid cellular
overseas calling charges from support staff in the field at our
customer sites. Support staff often spend hours on the phone to our UK
factory. However, I'm not sure about the implications of reinvite in
this arrangement.
Original Message
Subject: Re: [Asterisk-Users] Options for 3+ FXO ports
From: Jorge Mendoza [EMAIL PROTECTED]
Date: Mon, March 08, 2004 8:53 am
To: [EMAIL PROTECTED]
Rich Adamson wrote:
I'm looking into implementing an * solution and I'm expecting to
require 4
Earlier on I read that there is an IAX image for the SNOM 200. Is this true? Does it
work? Where might I get this?
Michael
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You may be right here. I was thinking of an ActiveX plug-in. I don't expect them to
use public internet kiosks so they should be able to use the ActiveX approach. I was
hoping that something IAX based could be found as it would make the connectivity
easier and open port risk reduced.
Michael
Hello All,
Does anyone here have any experience with pingtel Xpressa hard phones? I am
considering buying a couple. Already have Snom200s, but want something with better CTI
and full duplex speakerphone.
Michael
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