Displaying 20 results from an estimated 2000 matches similar to: "Opinions please: Polycom IP 430 vs 330?"
2008 Feb 22
1
[VOIP-Users-Conference] Re: Opinions please: Polycom IP 430 vs 330?
Let me add another variable into the mix...what about the Linksys
SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a
deal at $80 street price.
Michael
On Fri, 22 Feb 2008 08:28:37 -0500, Matthew Brothers wrote:
>
>Michael Graves wrote:
>> I need to add a few phones to an existing installation. They have a
>> dozen IP430 at the moment. Does anyone feel that
2007 Aug 16
2
Seeking opinions: Polycom IP330 phones?
Does anyone online have an opinion on these? I've used 500/510/6001/601
models before. Need to know if these apparently lesser models can be
provisioned in the same way. Are end uers happy with them?
Thanks,
Michael
--
Michael Graves mgraves at pixelpower.com
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc.
2008 Jan 20
2
SIP <> GSM
I'd like to add a device to my Asterisk server to leverage my cellular
account. Does anyone on-list have experience with hardware gateways vs
using cah_bluetooth and an old cell phone?
I'm considering something like http://www.mobigater.com/index.php?p=5
Thanks,
Michael
--
Michael Graves
mgraves<at>mstvp.com
blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mjgraves at
2007 Nov 07
2
wifi
I'd like to survey those on-list who actually use wifi SIP handsets.
What type of wifi access point do you use? Are you happy with it?
I presently use some older Linksys WAP54G APs. I'd like to replace
these but in doing so I'd like to be moving in a VOIP friendly
direction. I've yet to find a handset that I'd buy in quantity, but my
last round of access points lasted >4
2007 Oct 17
2
sorta OT: Bounty for Click to Call plugin for IE
I'm in process of transitioning a number of offices to a hosted virtual
pbx from Junction Networks. It's a combination of OpenSER and Asterisk.
They have a nice click-to-call extension for Firefox, but I need the
equivalent for IE so that it can work with our CRM system. Junction
told me that they have a bounty on offer for this if someone's
interested in doing the work.
Would the
2008 Jan 14
0
Aastra Venture
Does anyone on-list have any experience with this system? I'd like to
speak with someone who's tried it out.
Thanks,
Michael
--
Michael Graves
mgraves<at>mstvp.com
blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mjgraves at pixelpower.onsip.com
skype mjgraves
fwd 54245
2008 Jun 04
1
G.722?
Which flavor of G.722 has been implemented in Asterisk? And starting
with what release version?
Thanks,
Michael
--
Michael Graves
mgraves<at>mstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mjgraves at pixelpower.onsip.com
skype mjgraves
54245 at fwd.pulver.com
2007 Nov 28
0
Re :Recommendations for 100 Wifi SIP phone
Hi,
Try http://gigaset.siemens.com/shc/1,1935,hq_en_0_11729_rArNrNrNrN,00.html
you will have good selections there.
Regards,
Vidura Senadeera
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Message-ID: <200711252201.lAPM1NUP005677 at mail871.megamailservers.com>
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2009 Mar 31
2
dynamic codec preferences
Has anyone here ever had the occasion to setup a system that would
dynamically alter it's codec preferences based on trafffic? That is,
presuming that the system is on a limited bandwidth connection is would
start to prefer a compressed codec as the call volume increased?
Perhaps shifting from G.711 to G.729?
Michael
--
Michael Graves
mgraves<at>mstvp.com
http://blog.mgraves.org
2008 Sep 28
1
G.722 between Eyebeam and a Polycom IP650
Hi All,
So I've been exploring the use of G.722 encoded wideband audio
recently. I have three different SIP devices that allow this: Eyebeam,
IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine
together. Calls pass between them in what the Polycom notes as "HD"
mode and the audio quality is certainly very good.
However, things are not so easy with Eyebeam and the
2010 Dec 30
1
VUC; Friday December 31st - 2010: The Year in VoIP
On this weeks VUC call we will collectively be our own guests. That is,
we'd like to know what was the big issue that impacted YOU in 2010? All
opinions welcome.
Here are a few things to get you thinking in advance:
- Apple's Antenna-gate
- Asterisk 1.8 Launches
- Amazon EC2 as a DOS platform
- Cisco launched UMI video conference device
- More HDVoice capable phones
- Skype Outage
- VoIP
2009 Feb 08
0
Streaming meetings vs conference hardware
There simply aren't that many local Asterisk Users Groups. This is one
of the reasons that the VoIP Users Conference
(http://www.VoIPUsersConference.org) has been picking up members. Using
the Talkshoe conference bridge is certainly accessible, and the price
is right (free!)
I listened in on the streams from 25C3 over the Christmas break, and
fout it really enjoyable. It occurs to me that
2008 Nov 04
5
VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing
This Friday's edition of the weekly VoIP Users Conference call is all
about wideband audio (aka HD Voice) and conferencing. The guest for
this call is David Frankel, CEO of ZipDX a commercial service that
specializes in wideband conferencing. We expect an interesting call
touching on many aspects of VoIP going beyond the traditional phone
service, conference bridges, technical standards,
2009 Nov 24
1
DIDs > PBX > Multi-channel balanced audio output?
There came up an interesting question on last weeks VoIP Users Conf
call. http://vuc.me. An internet broadcaster wanted to be able to take
a number of "phone calls" and bring them into a traditional audio mixer
where each call was a separate balanced audio source to the mixer.
Let's say that he might need 8 simultaneous calls. Is there a way to
set this up using a single Asterisk
2011 Jan 27
0
OT: VoIP Users Conf Feb 4 with LifeSize
Hi All,
My appologies for the off-topic post, but I thought it would still be
of interest to this list.
If you've been reading our site at http://vuc.me you've no doubt seen
that we have the video call with LifeSize scheduled for Feb 4th. What?
You didn't know? Here's t
scoop:
http://www.voipusersconference.org/2011/hd-video-conferencing-with-lifes
ize/
We have a number of
2007 Oct 07
1
New Embedded Distro
Hi All,
A couple of weeks ago I noticed Askozia PBX, which is a new embedded
Asterisk & OS distro at http://askozia.com/pbx. This caught my
attention for two reasons; it uses v1.4 of Asterisk, and it uses the
m0n0wall development framework to build on FreeBSD with a PHP based
GUI. I've used m0n0wall for years, and FreeNAS also, which shares the
same OS/GUI framework.
I booted the latest
2007 Aug 15
0
slightly OT: Polycom SIP phones
How difficult is it to change the firmware load on a Polycom phone from
MGCP to SIP? I have a number of 500/600 phones and see some used phones
being offered with MGCP installed. I have the SIP firmware but have
never had to migrate between content loads. Any expected gotcha's?
Michael Graves
--
Michael Graves mgraves at pixelpower.com
Sr. Product Specialist
2009 Jul 08
0
Grandstream GXP-1200 & G.722?
Can anyone here have experience using G.722 on the Grandstream GXP-1200?
It's supposed to support the codec, but I wonder if the handset does it
justice?
The older BT-200 also supported the codec, but the handset was not good
enough. You could only hear the improved call quality using a headset.
Michael Graves
mgraves <at> mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgraves at
2009 Apr 21
4
Polycom wideband codecs?
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:mjgraves at mstvp.onsip.com
skype mjgraves
2010 Jul 09
0
Pbx_för_Windows?_-_Email_f ound_ in_subject
Hi all,
Many thanks for your replies!
Will tell my friend and see what he will be interested in.
Many thanks!
Christian
-----Ursprungligt meddelande-----
Fr?n: mgraves at mstvp.com
Till: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Skickat: 10-07-09 15:29
?mne: Re: [asterisk-users] Pbx_f?r_Windows?_-_Email_f ound_ in_subject
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