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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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
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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> 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
> requ
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 experienc
This week marks one year since I first setup an Asterisk server in the
hopes of transitioning my home office to a total VoIP system. The
process has been an incredible learning experience. I've tried numerous
IP hard phones, eventually settling upon the Polycom IP600 as my
choice. I've also used mu
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.
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 has
f the Digium or
Sangoma hardware is not a problem. And, after all, you get what you pay
for in most cases.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
> Original Message
> Subject: Re: [asterisk-users] [Zaptel]
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 mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
> Original Message
> S
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 curious
> 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
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 a
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
mg
The quad-band model is around $250 USD.
See Ebay auction here http://tinyurl.com/5tvoa9
Michael Graves
mgraves 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 G
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 mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjg
Headset mic? Drive safe ;-)
Michael Graves
mgraves 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]&
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
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 mstvp.com
o(713) 861
That's the purely technological answer, which is completely correct.
There's a business side to it as well. Siemens is simply not in the
consumer electronics business in North America. They make this decision
consciously.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 20
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 mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245
> Ori
adset.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
FWD 54245
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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: [asterisk-use
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 mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
FWD 54245
> Original Message
> Subject: Re: [asterisk-users] Polycom wideband codecs?
> From: randulo
> Date: Tue, April 21, 2009 1:40 pm
> To: Asterisk Use
> 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
>
>
> Folks,
>
> I'm looking at buying an FXO card for my home Aster
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 PROTECTED]
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
> 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 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
f(713) 864-8668
c(713) 201-1262
> Original Message
> 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"
>
>
>
> That should not be a problem. M
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
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 direc
> 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"
>
>
> Has anyone tried out Hitachi IPC-5000
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
o(800)
> 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 provi
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 pr
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
f(713) 864-8668
c(713) 201-1262
> Original Message
> Subject: R
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 effor
Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
> Original Message
> Subject: [asterisk-users] OT - Which Android handset with Wifi-only ?
soft 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
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
e-pbx/
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
> Original Message
> Subject: [asterisk-users] Dual Atom mobo - call capacity
> From: Michelle Dupuis
> Date: Thu, June 10, 2010 7:19 pm
> To:
erisk-appliances-a-question-of-scale/
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
> Original Message
> Subject: Re: [asterisk-users] Pbx_för_Windows?_-_Email_found_
> in_subject
> From: Doug Lytle
> Date:
ew-polycom-soundpoint-ip335-entry-level-hdvoice-ip-phone/
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
> Original Message
> Subject: Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
> From: An
/670 models.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
> Original Message
> Subject: Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD
> voice codecs?
> From: Steve Underwood
> Date:
Actually, all of the conference phones are known by the "SoundStation"
name and the desk phones are "SoundPoint."
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
> Original Message
> Su
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