Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

2011-05-09 Thread mgraves
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: Olivier oza_4...@yahoo.fr
 Date: Mon, May 09, 2011 7:10 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 Hi,
 
 I would be curious to play with an Android phone with Wifi-only capability.
 My plan is to install Bria on it and see if it could be used within a couple
 of WiFi access points, as a high-end wireless phone.
 
 Which handset would you recommend ?
 
 Regardshr--
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Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?

2011-01-09 Thread mgraves
Actually, all of the conference phones are known by the SoundStation
name and the desk phones are SoundPoint.


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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: Sat, January 08, 2011 11:16 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 On 01/08/2011 03:44 AM, Kevin P. Fleming wrote:
  On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote:
  We should also be very clear that the Siren codecs are supported on the
  Polycom SoundStation conference phones and the VVX-1500 Business Media
  Phones. These codecs are not supported in the SoundPoint desk phones.
  The SoundPoint series support the more basic G.722 codec in the
  IP335/450/550/560/650/670 models.
 
  The SoundPoint IP6000 and IP7000 conference phones (and maybe the 
  IP5000, I haven't checked) also support G.722.1 and G.722.1C.
 
 The IP6000 is actually model Polycom recommended for testing when we 
 implemented G.722.1.
 
 One of the annoying things about the Polycoms is trying to work out what 
 they can do. You have to search quite hard to find which codecs each 
 model supports.
 
 Steve
 
 
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Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?

2011-01-06 Thread mgraves
We should also be very clear that the Siren codecs are supported on the
Polycom SoundStation conference phones and the VVX-1500 Business Media
Phones. These codecs are not supported in the SoundPoint desk phones.
The SoundPoint series support the more basic G.722 codec in the
IP335/450/550/560/650/670 models.

Michael Graves
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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, January 05, 2011 6:09 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 On 01/06/2011 12:05 AM, Kevin P. Fleming wrote:
  On 01/05/2011 07:07 AM, Steve Underwood wrote:
  On 01/05/2011 03:29 PM, Bruce B wrote:
  Hi Everyone,
 
  1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
  2- Are these codecs only for Polycom units or are they universal
  across all other SIP phones that advertise the HD voice codec like
  Aastra?
  3- What is the main difference between the two and is it advisable to
  run these over the INTERnet (not INTRAnet)?
 
  The G.722 codec in * is G.722. The Siren7 codec in * is probably not
  Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a
  different code in the SDP and has some minor differences in the codec.
  The name G.722.1 may look similar to G.722, but the codecs bear no
  relation to each other. The Siren14 codec in * is probably not Siren14,
  but G.722.1C. G.722.1C is very similar to Siren14, but like
  Siren7/G.722.1 the SDP code is different, and there are minor
  differences in the codec.
 
  Asterisk actually supports both the Siren* and G.722.1* names in SDP 
  negotiations. I wasn't aware there were bitstream incompatibilities 
  between the Siren* and G.722.1* variants, even though the code may be 
  slightly different... so Asterisk uses a single codec module for both 
  variants.
 
 I am unclear how compatible or incompatible the bitstreams may be. What 
 I know (from implementing these codecs) is that the source code Polycom 
 provide licencees, as the basis for developing their own G.722.1 and 
 G.722.1C codecs, has several comments referring to things not being 
 quite the same as Siren7/Siren14. However, they don't hand out the 
 actual Siren7/Siren14 source code, so I don't know how much divergence 
 there is.
 
 Steve
 
 
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Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)

2011-01-04 Thread mgraves
IMHO G.722 beats Clarity By Polycom every time. 

I had an IP335 for review before they launched. The audio quality is the
same as the better models (IP450/550/650) only the user interface is
different. Very good speakerphone, too.

Review here: 

http://www.mgraves.org/2010/01/review-polycom-soundpoint-ip335-entry-level-hdvoice-ip-phone/

Michael Graves
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  Original Message 
 Subject: Re: [asterisk-users] VoIP PoE phones for restaurant (kitchen)
 From: Andy Graybeal andy.grayb...@casanueva.com
 Date: Tue, January 04, 2011 4:15 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
  The Polycom 321 has not been EOL'd and supports VLAN.  It is, however,
  lacking a 2nd ethernet port if you were to go that route.
 
  -M
 
 Thanks for the response Mark.  I see the 331 has two ports and the same 
 features as the 321.
 
 I'm wondering what phone would be best being used as an intercom in a 
 busy kitchen.  I asked this some months ago; but this time around I'm 
 writing it into this years budget.
 
 I see the 335 has HD Voice and the 321 has Clarity by Polycom.  Which 
 would be best in a noisy kitchen using the devices speaker phone?
 
 Should I seek another device for the kitchen all-together?
 
 -Andy
 
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Re: [asterisk-users] Pbx_för_Windows?_-_Email_f ound_ in_subject

2010-07-09 Thread mgraves
I echo the sentiment that you should just run Asterisk on some small
hardwarein an appliance like fashion. In fact, just yesterday I
posted an overview of hardware suitable for DIY appliances. I've used
many of the platforms mentioned.

http://www.mjgraves.com/2010/07/08/d-i-y-asterisk-appliances-a-question-of-scale/

Michael Graves
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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, July 09, 2010 8:17 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 Arjan Kroon | Mobillion wrote:
  Mayby Freepbx.
  http://www.freepbx.org/
 
 
 
 And,  as their page states,
 
 FreePBX is an easy to use GUI (graphical user interface) that controls 
 and manages Asterisk
 
 Doug
 
 -- 
 Ben Franklin quote:
 
 Those who would give up Essential Liberty to purchase a little Temporary 
 Safety, deserve neither Liberty nor Safety.
 
 
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Re: [asterisk-users] Dual Atom mobo - call capacity

2010-06-10 Thread mgraves
Based on comments from Ward Mundy during a recent VUC call I'd expect
even a single CPU Atom system to handle that many phones in an office
application. Perhaps there may be merit in dual CPU in more of a call
center application.

http://www.voipusersconference.org/2010/nerd-vittles-incredible-pbx/

Michael Graves
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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 List asterisk-users@lists.digium.com
 
 
 


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Re: [asterisk-users] State of JACK support i9n Asterisk

2010-05-24 Thread mgraves
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
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  Original Message 
 Subject: [asterisk-users] State of JACK support i9n Asterisk
 From: Julien Claassen jul...@c-lab.de
 Date: Mon, May 24, 2010 8:51 am
 To: asterisk users mailinglist asterisk-users@lists.digium.com
 
 
 Hello everyone!
I haven't seen anything new about the JACK support in Asterisk and I was 
 wondering, if anyone has experience with a current release of Asterisk, JACK 
 and mISDN/googletalk etc. I'm thinking of installing a new version 
 (havingcurrently 1.60-beta9. But the excercise would be pointless, if it 
 doesn't help.
Kindly yours
  Julien
 
 
 Music was my first love and it will be my last (John Miles)
 
  FIND MY WEB-PROJECT AT: 
 http://ltsb.sourceforge.net
 the Linux TextBased Studio guide
 === AND MY PERSONAL PAGES AT: ===
 http://www.juliencoder.de
 
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Re: [asterisk-users] Softphones on thin clients...

2010-05-20 Thread mgraves
I've used HP Thin Clients as embedded hosts for Asterisk. The T5700
models that I have are 1 GHz CPUs, more recent models should be able to
run a soft phone without too much trouble. They all have local USB
ports, making USB headsets as good solution.

Another alternative might be to used a 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
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  Original Message 
 Subject: Re: [asterisk-users] Softphones on thin clients...
 From: Carlos Chavez cur...@telecomabmex.com
 Date: Thu, May 20, 2010 1:36 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
  Sent: Thursday, May 20, 2010 1:51 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Softphones on thin clients...
  
  
  On 20 May 2010, at 18:35, Carlos Chavez wrote:
 I am worried about conflicts when running 10 softphones on the same
   server since they will all try to use por 5060.
  
  And the fact most terminal services servers/clients still don't support
  audio input.. only output..
 
   Since the little box has a MIC jack I suppose that it should support
 audio input.  These boxes will be running Windows and using Eyebeam.
 
 -- 
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Carlos Chávez Prats
 Director de Tecnología
 +52-55-91169161 ext 2001hr-- 
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Re: [asterisk-users] SIP/WiFi handsets?

2009-09-23 Thread mgraves
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-users] SIP/WiFi handsets?
 From: Jason Baker jba...@glastender.com
 Date: Wed, September 23, 2009 10:02 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 Ken,
  I did lots of research on this for my VoIP deployment here where I work. We 
 have a huge manufacturing floor and all the supervisors have wifi phones. We 
 evetually settled on the Polycom Spectralink 8002. A nice rugged little phone 
 with great sound quality and some good features. We use a managed switch to 
 create seamless wifi coverage over all of our AP's. Provisioning the phone is 
 pretty easy, but no web browser if you were planning on using the phone to 
 travel with, some hotels require login for internet access.
  
  I also tried a clamshell wifi SIP phone by D-Link. This phone actually works 
 really well, but we had some minor issues with it so we went with all 
 Spectralink phones. But the D-Link phone would be good choice if you plan to 
 take your wifi phone on the road.
  
  I also tested the Linksys WIP330 which I thought was a terrible phone. Very 
 difficult to use.
  
  Good luck.
  
  
 http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html
  http://www.dlink.com/products/?pid=485
  http://www.voipsupply.com/linksys-wip330-na
Jason Baker
  IT Coordinator
   Glastender, Inc.
  5400 North Michigan Road
  Saginaw, Michigan 48604 USA
  Phone: 989.752.4275 ext. 228
  Fax: 989.752.4276
  www.glastender.com 
 
  
  Ken D'Ambrosio wrote:  Anyone know of any *portable* SIP/WiFi handsets? 
 Looking for a decent
  price:quality ratio, of possible. Keep seeing handsets for Vonage, etc.,
  in Best Buy and the like, but I imagine it's locked to Vonage, and can't
  be re-appropriated.
  
  Thanks!
  
  -Kenhr___
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[asterisk-users] Grandstream GXP-1200 G.722?

2009-07-08 Thread mgraves
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.

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[asterisk-users] Polycom wideband codecs?

2009-04-21 Thread mgraves
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
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Re: [asterisk-users] Polycom wideband codecs?

2009-04-21 Thread mgraves
In this case I had, in my hurry this morning, simply confused G.722.1C
and G.722.2. These are both low bitrate wide bandwidth codecs. 

They are also known by the Polycom marketechure nomenclature of Siren7
and Siren14. G.722.1 supporting 7 KHz passband, while G.722.1C support
14 KHz passband.

Michael Graves
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  Original Message 
 Subject: Re: [asterisk-users] Polycom wideband codecs?
 From: randulo spamsucks2...@gmail.com
 Date: Tue, April 21, 2009 1:40 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 On Tue, Apr 21, 2009 at 4:40 PM, Steve Underwood ste...@coppice.org wrote:
  Which Polycom supports G.722.2? I think they are only supporting G.722,
  G.722.1 and G.722.1C right now.
 
 Could someone enlighten me, what is the difference (the result part
 that matters, not the spec)?
 
 r
 
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[asterisk-users] A Suggestion To Asterisk Appliance Developers

2008-08-21 Thread mgraves
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
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c(713) 201-1262
sip:[EMAIL PROTECTED]
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Re: [asterisk-users] 5 min limitation on phone calls! how to!

2008-08-21 Thread mgraves
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
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c(713) 201-1262
sip:[EMAIL PROTECTED]
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  Original Message 
 Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
 From: Singer XJ Wang [EMAIL PROTECTED]
 Date: Thu, August 21, 2008 2:42 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 Phone Guy: NO PHONE FOR YOU!
 
 Karl Fife wrote:
  This has got to be one of the funniest threads ever to grace this forum.
  Sorry honey! ...CLICK.
  In my house, this might require a more 'diplomatic' approach :-)
  -Karl 
 
 
 
  On Thu, 21 Aug 2008 21:41:40 +0300, RoLaNd RoLaNd
  [EMAIL PROTECTED] said:

  i tried that before.. it didnt actually work! it the call kept on going
  well beyound the allowed test seconds...
  heres my extensions.conf:
 
 
  [sipura-line]
  exten = 301,1,Answer() ; Answer inbound calls
  exten = 301,2,Playback(silence/1)
  exten = 301,3,Background(simzy1) ; input an extension
  exten = 301,4,WaitExten(8)
  exten = 301,5,Dial(SIP/100,15) ; goes to operator
  exten = 301,4,Wait(8)
  include = spa
  exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
  exten = 301,n,Hangup()
 
 
 
 
  [spa]
  exten =_301,1,GoTo(sipura-line,${EXTEN},1)
  exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
  it will ring 3 times
  exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if
  line is busy or unavailable
  exten = _1XX,3,HangUp()
  exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds so
  it will ring 3 times
  exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if
  line is busy or unavailable
  exten = _2XX,3,HangUp()
  exten = _3XX,1,Dial(SIP/${EXTEN},15) ; each ring equals to 5 seconds so
  it will ring 3 times
  exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2 voicemail box if
  line is busy or unavailable
  exten = _3XX,3,HangUp()
  exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
  exten =_01,2,Set(TIMEOUT(absolute)=5)
  exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
  exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
  exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
  exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
  exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
  exten = 303,1,VoicemailMain ; voicemail box to be redirected to
 
 
 
 
  
  Date: Thu, 21 Aug 2008 20:26:48 +0300
  From: [EMAIL PROTECTED]
  To: asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] 5 min limitation on phone calls! how to!
 
  RoLaNd RoLaNd schrieb:

  Hello all!
   
  my last month's phone bill sky rocketed after i setup asterisk with
  softphones all over the house!
 
  could someone help me set up a limitation for my wife and kids not to be
  able to talk for more than 5 min at a time!
  or like 20 min per week! or whtever limitation i could set for this!
  
  Set(TIMEOUT(absolute)=seconds)
 
  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout
 
 
  Terve,
  Stefan
 
  -- 
  Last words of a stormchaser:
  Where is that rotation on the radar?!
 
 
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  http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE
  
 
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Re: [asterisk-users] Randulo: An open suggestion for the VOIP users Conference

2008-08-07 Thread mgraves
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 leave short-changed.  There
are other examples even from last week, one involving someone's claim
that there's not a way to pick up a phone and directly interface with a
voice recognition directory application without needing to press some
digits first.  As it turns out, it's easy if you know the trick.  

Id' be happy to put my money where my mouth is and kick off this
Friday's show with these examples  any others I'm not remembering at
this moment if you think it would be well received.  Perhaps others will
do the same.
What do you think?  

Thanks!
-Karl Fife

If you want to discuss this off-list, you can email me at
[EMAIL PROTECTED]

p.s.
As it turns out, HDVoice CAN use G.722, but it can also be overlain onto
other codec's such as use G.722.1 and even G.711µ [sic].  That's right,
you can have an HDVvoice call over the PSTN using G.711, using a
special companding overlay on top G.711.  As I understand it, the two
HDVoice compliant endpoints (Polycom, Cisco  others that license the
technology) have an in-band (but inaudible) handshake, and then begin
applying the proprietary companding overlay which extends the dynamic
range of the audio.  It sounds great even though the underlying codec is
not a wideband codec.  Certainly the sound is not as good as HDVoice
over a modern adaptive-transform codec like G.722 (1987) or even better
over G.722.1 (1999), but it's definitely a big improvement over the
Toll-Quality (Read: AM-Radio-Through-A-Pillow) that we're all used to,
and it is not dependent upon having a pure-IP connection involving ENUM,
DUNDI, or other non e.164 namespaces such as SIP URI's, ITAD Subscriber
Numbers etc.  In my opinion HDVoice is it's a brilliant transition
technology.  


Karl,

This is very interesting. Did you see that Polycom made G.722.1
available through a royalty free license earlier th
week?

http://www.polycom.com/usa/en/company/news_room/press_releases/2008/2008
0805.html

In tinkering with the three phones that I have (ip650/550  Siemens
S685IP) they all support only G.722. At least according to the
datasheets even the Polycom models don't handle G.722.1 as yet. 

Mind you I haven' gone so far as to use Wireshark to analyse the
traffic. Just measure the bandwith used across my router and note when
the phone indicates HD engaged on the line button.

Do you know if the companded processing you mention is implemented in
the Soundpoint models? Just from the sound of it it could improve S/N
ratio, be perhaps not frequency response. Still, it's good that they
can improve a call over the PSTN.

So, perhaps there's more to HDVoice than just G.722. Even so, G.722 is
all that I have experienced of HDVoice in their current Soundpoint IP
phones. I suspect that some of their technology is only deployed in
their larger conferencing systems, and not in the Soundpoint lineup.

Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves





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Re: [asterisk-users] The S word: Asterisk security

2008-07-01 Thread mgraves
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]
skype mjgraves
FWD 54245


  Original Message 
 Subject: Re: [asterisk-users] The S word: Asterisk security
 From: Kristian Kielhofner [EMAIL PROTECTED]
 Date: Tue, July 01, 2008 10:56 am
 To: Asterisk Users Mailing List - Non-Commercial
 Discussionasterisk-users@lists.digium.com
 
 
 On 7/1/08, randulo [EMAIL PROTECTED] wrote:
  Hi all,
 
   As I mentioned briefly in the SIP takeover thread, I'd like to try to
   talk about security this coming Friday. I realize it is a holiday in
   the USA, but do geeks ever take a day off, especially
   security-conscious geeks? Mark Spencer once said The Bug Tracker is
   never on vacation!.
 
   We will try to start this subject this Friday, but I have no
   experience at all with this. If you know anyone who is good in this
   area and would like to share their expertise and talk about security
   in the asterisk and voip contexts, I'd like to hear from them,
   especially next Friday July 4th.
 
   tia,
 
   Randy
 
 
 Randy,
 
   I'd love to participate as long as no one minds me calling in from
 the beach... :)
 
   I'm interested in developing my SIP DoS script (and any similar
 solutions).  While I'm reluctant to claim that it or anything like it
 could protect from a true DoS, it would offer some protection at the
 application level and that could make all the difference in some
 instances...
 
   As far as wider Asterisk/security issues I think J. Oquendo would be
 a great guest (hint, hint).
 
 -- 
 Kristian Kielhofner
 NOT sent from my iPhone or Blackberry
 
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Re: [asterisk-users] The S word: Asterisk security

2008-07-01 Thread mgraves
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 01, 2008 12:30 pm
 To: Asterisk Users Mailing List - Non-Commercial
 Discussionasterisk-users@lists.digium.com
 
 
 On Jul 1, 2008, at 11:29 AM, randulo wrote:
 
  Hi all,
 
  As I mentioned briefly in the SIP takeover thread, I'd like to try to
  talk about security this coming Friday. I realize it is a holiday in
  the USA, but do geeks ever take a day off, especially
  security-conscious geeks? Mark Spencer once said The Bug Tracker is
  never on vacation!.
 
  We will try to start this subject this Friday, but I have no
  experience at all with this. If you know anyone who is good in this
  area and would like to share their expertise and talk about security
  in the asterisk and voip contexts, I'd like to hear from them,
  especially next Friday July 4th.
 
  tia,
 
  Randy
 
 I love it. I'm celebrating the 4th with a 2000 mile motorcycle ride :)  
 I'll do my best to make it for the conference.
 
 
 Fred Posner
 www.voiptechchat.comhr___
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Re: [asterisk-users] Asterisk GSM Gateway Project

2008-06-23 Thread mgraves
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 Project
 From: Dinesh Nair [EMAIL PROTECTED]
 Date: Mon, June 23, 2008 1:03 am
 To: Asterisk Users Mailing List - Non-Commercial
 Discussionasterisk-users@lists.digium.com
 Cc: [EMAIL PROTECTED]
 
 
 On Sun, 22 Jun 2008 08:45:10 -0500, Michael Graves wrote:
 
  I have a small Portech GSM gateway. It works well. It's GSMSIP which
  seems to me a better solution than FXO/FXS type interfaces. They make
  gateways up to 32 port for E-1 interconnect.
 
 what did they cost, michael ?
 
 
 -- 
 Regards,   /\_/\   All dogs go to heaven.
 [EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
 +==oOO--(_)--OOo==+
 | for a in past present future; do|
 |   for b in clients employers associates relatives neighbours pets; do   |
 |   echo The opinions here in no way reflect the opinions of my $a $b.  |
 | done; done  |
 +=+


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[asterisk-users] SIP over M$ ISA

2008-06-09 Thread mgraves
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 at mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
FWD 54245




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Re: [asterisk-users] Microsoft Office Communications Server

2008-03-10 Thread 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: Re: [asterisk-users] Microsoft Office Communications Server
 From: Kristian Kielhofner [EMAIL PROTECTED]
 Date: Mon, March 10, 2008 5:18 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 On Sun, Mar 9, 2008 at 11:05 PM, Matt Riddell [EMAIL PROTECTED] wrote:
  -BEGIN PGP SIGNED MESSAGE-
   Hash: SHA1
 
   Has anyone done any integration with this?
 
   All I know so far is that it appears to use some non standard form of SIP.
 
   Any pointers?
 
   - --
   Kind Regards,
 
   Matt Riddell
   Director
 Matt,
   I believe OCS only supports SIP over TCP.  You'll either need to use
 Asterisk 1.6/trunk with SIP TCP or install SER/OpenSER as a UDP-TCP
 proxy.
 -- 
 Kristian Kielhofner
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Re: [asterisk-users] [Zaptel] Why no port to Windos?

2007-12-14 Thread mgraves
I'm a platform agnostic. I need to use a bit of everything in my daily
work. I will say that my current XP desktop has been very reliable. It's
not uncommon for it to stay up for a couple of weeks at at time without
a reboot. My linux and FreeBSD systems routinely go months untouched.

That said, consider the potential market size for people, the DIY sorts,
who would have Asterisk in their homes. Very small. Hence there's little
reason for someone to build hardware targeting that market and its
economic sensibilities.

Moving slightly up market into SMB space the cost of the Digium 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?
 From: Lee Jenkins [EMAIL PROTECTED]
 Date: Fri, December 14, 2007 9:51 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 Doug wrote:
  At 19:55 12/13/2007, Vincent wrote:
   Hello
   
   I was wondering why there doesn't seem to a Windows version of Zaptel,
   making the Digium and its clones unavailable for a Windows PBX.
   
   Is the Zaptel/Zapata combo too *nix-centric?
   
   Thanks.
  
  Windows is a half-baked, dying OS that in essence is
  a 32 bit extension and graphical shell, for a 16 bit
  patch to an 8 bit operating system, originally coded
  for a 4 bit microprocessor, written by a 2 bit
  company, that can't stand 1 bit of competition.
 
 Nice.
 
  Do you really want to reboot your telephone system
  3 times a day?
  
 
 I'm not a Windows basher as I make a good living from Windows based software, 
 but I couldn't see it either.
 
 My asterisk box was rebooted about 3 months ago when I made some changes 
 last. 
 It's running Asterisk, FirebirdSQL, 1 FastAGI server and a lot of natively 
 compiled AGI executables handling tech support, sales, caller id database 
 lookups, nag calling, etc, etc.
 
 
 I have to reboot my desktop xp box daily for it to run well.
 
 
 -- 
 Warm Regards,
 
 Lee
 
 If I don't see you around here, I'll see you around, hear?
 
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Re: [asterisk-users] Your favorite desktop wifi sip hardphone ?

2007-11-09 Thread mgraves
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 wifi device. The WPB54G. See http://tinyurl.com/2pn4v4

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
c(713) 201-1262



  Original Message 
 Subject: [asterisk-users] Your favorite desktop wifi sip hardphone ?
 From: Olivier [EMAIL PROTECTED]
 Date: Fri, November 09, 2007 1:55 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 Hi,
 
 Which is your favorite desktop wifi sip hardphone ?
 I'm looking for something like
 http://www.mitel.com/DocController?documentId=19401 which could be easily
 moved from one meeting room to another.
 
 (In this specific case, finding an electrical plug to power a large desktop
 phone is seen more relevant than finding an PoE Ethernet plug or using a
 mobile handset.)
 
 Which product would you recommend ?
 
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Re: [asterisk-users] Best config for 12 FXO system?

2007-10-05 Thread mgraves

  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 bank
 So why have Digium bothered to market a TDM2400P then? :-S
 Cheers
 Tony

Because they don't make channel banks? That's a question of economics
and techno-culture. If you're into boards (X100p, TDM400p, etc) then
making another board fits your worldview. If not, then perhaps a channel
bank makes more sense. 

Xorcom's USB connected device is kinda like a channel bank. I believe
that you can get 12 FXOs via that device and not need a t-1/E-12 card in
the Asterisk server at all. That could be the most cost effective
approach.

Michael


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[asterisk-users] Aastra phones?

2007-05-08 Thread mgraves
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 as to the real differences.

Thanks,

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
[EMAIL PROTECTED]
FWD 54245



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RE: [Asterisk-Users] Prices of g729 codec

2006-06-07 Thread mgraves
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 efforts. It would be really easy for them to say "no more" and it wouldn't really impact their business at all, except to reduce their headaches.

This will be especially true when they introduce their new hardware based transcoding engine. Why then should they continue to deal with the per stream softwaer codec licensing? If you want access to G.729a just buy the board...the license cost withn be buried in the price and they can afford to provide support to paying customers.

Again, we should be gratefull! It could very easily go away altogether.

Those of you constantly complaining...this is supposed to be a open source community...don't just demand a better licensing scheme...design and implement one. That can be your contibution to the project. I'm not a code jockey or I'd have a go myself.

In the interest of full disclosure, I have a small systems based upon Astlinux and a Soekris Net4801. I have 2 G.729a licenses on that box and I'd like to see Digium make the codec possibleusing the alternative C libraries that Kristian has used in Astlinux 0.4. I probably can't justify buying the hardware transcoder. And I definitely don't want them to withdraw the current codec offering.Michael GravesSr Product SpecialistPixel Power Inc[EMAIL PROTECTED]o(713) 861-4005o(800) 905-6412f(713) 864-8668c(713) 201-1262

 Original Message Subject: Re: [Asterisk-Users] Prices of g729 codecFrom: "Woodoo People .pGa!" [EMAIL PROTECTED]Date: Mon, June 05, 2006 10:15 amTo: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Talk to digium about this on [EMAIL PROTECTED], they might be able to  help you out there.  Zoa  Chris Mason (Lists) wrote:  I have no problem with paying Digium the $10 for G729 licenses,  everyone has to make money. It's the administration of the licenses  that sucks. I experiment with different hardware a lot, and make up  demo machines to install for customers with available hardware. I have  to put G729 licenses on them, usually $100 each time, and when I  insta
 ll the real hardware for the client, I can't transfer the  licenses. If I scrap that machine or change the interfaces, that's a  $100 loss. I believe when you buy a number of licenses, that should  determine how many instances you can use, regardless of how you want  to deploy them. In short, the method of enforcement is poor and leads to resentment  from customers. Surely Digium can construct a better system?i think, for those of us, who would like to transfer licences from one boxto other (i mean more than 1-2 or 10), we would have to buy a hardwarebase lock (of course, i don't care about, if the lock would contactdigium once a day or so) like usb, or a dumb pci ethernet card, soif we need we can move it to other. what do you think?(sadly there is no a 7day demo licence or anything to test) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
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RE: [Asterisk-Users] soekris hadware

2006-05-17 Thread mgraves
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: Re: [Asterisk-Users] soekris hadware
 From: olivier.taylor [EMAIL PROTECTED]
 Date: Wed, May 17, 2006 11:12 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
more kindly :
  
  http://www.astlinux.org/
  
  Olivier
  
  Christopher Snell a écrit : Google and voip-info.org will have answers to 
 all of your questions.
  
  
 On 5/17/06, Jonathan Gonzalez  [EMAIL PROTECTED] wrote: Hi group,
  
  i'm brand new and i would like to ask about soekris hardware. I read 
  along the web but i have some doubts that i think can be solved here.
  My question are the following:
  
  [...]

  
  
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RE: [Asterisk-Users] Compare to Skype

2006-04-30 Thread mgraves
  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 quality..  All about the codec in
 this case..


Errr...no...this is wrong. 

Skype uses ISAC from Global IP Sound. iLBC is something different see
http://www.globalipsound.com/solutions/solutions_Codecs.php

One of the reasons Skype sounds good is that its a closed system and so
can leverage a wideband codec. Instead of the normal 8khz sample rate
it uses 16khz. That makes for clearer sound. Since ISAC is a
proprietary relative of iLBC its jitter compensation is also very good.

My understanding is that Asterisk cannot presently use any wideband
codecs as it is hard coded to the 8khz sample rate at its core.
Adapting Asterisk to wideband capability has been discussed but will be
a huge amount of work. Further, only if you know that the calls will
stay wideband end-to-end will the benefits of wideband be apparent.
That means no PSTN segments.

Michael Graves
[EMAIL PROTECTED]


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RE: [Asterisk-Users] Compare to Skype

2006-04-30 Thread mgraves
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 processed the streams to bridge into the TDM/PSTN
world. It would be greatbut don't hold your breath.

For now there are Skype bridges like PSWG and Uplink that interface
Skype to SIP. These are simplistic but sometimes workable.

Does anyone here have experience with Uplink? I tried PSGW and gave up
eventually.

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: Re: [Asterisk-Users] Compare to Skype
 From: Ronald Wiplinger [EMAIL PROTECTED]
 Date: Sun, April 30, 2006 9:09 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 [EMAIL PROTECTED] wrote:
   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 quality..  All about the codec in
  this case..
  
 
 
  Errr...no...this is wrong. 
 
  Skype uses ISAC from Global IP Sound. iLBC is something different see
  http://www.globalipsound.com/solutions/solutions_Codecs.php
 
  One of the reasons Skype sounds good is that its a closed system and so
  can leverage a wideband codec. Instead of the normal 8khz sample rate
  it uses 16khz. That makes for clearer sound. Since ISAC is a
  proprietary relative of iLBC its jitter compensation is also very good.
 
  My understanding is that Asterisk cannot presently use any wideband
  codecs as it is hard coded to the 8khz sample rate at its core.
  Adapting Asterisk to wideband capability has been discussed but will be
  a huge amount of work. Further, only if you know that the calls will
  stay wideband end-to-end will the benefits of wideband be apparent.
  That means no PSTN segments.
 
  Michael Graves
  [EMAIL PROTECTED]
 

 
 Sadly to say, but users do not care about the why, they only care about 
 the quality! and they simple ask to fix it!
 
 I hope there is soon a solution, otherwise, we have to skip all our 
 effort and just use skype!
 And I would hate to see that. I just lost 20 US$ to Ebay - the newly 
 parent company of skype, for a not received parcel, but the rules says, 
 below 25 US$ there is no guarantee that you get anything
 
 
 bye
 
 Ronald Wiplinger
 
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RE: [Asterisk-Users] USB conference phone

2006-04-29 Thread mgraves
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]
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o(800) 905-6412
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  Original Message 
 Subject: RE: [Asterisk-Users] USB conference phone
 From: Kerry Garrison [EMAIL PROTECTED]
 Date: Wed, April 26, 2006 7:01 pm
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 
 
 it was a revewiers sample that I begged them to not make me send it back and 
 they let me keep it. 
   
  
  
 Kerry Garrison
 Publisher - http://GeekGazette.com - http://VOIPSpeak.net 
 (949) 502-7819 x200 - [EMAIL PROTECTED]
 http://www.techdatapros.com 
   
   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
 Sent: Wednesday, April 26, 2006 4:46 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] USB conference phone
 
  
  
  Lol  now the important question.Did you pay for it or was it a reviewers 
 sample J   
  
  
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry 
 Garrison
 Sent: Wednesday, 26 April 2006 7:23 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] USB conference phone   Yes, I have that device, 
 I wrote the review of it and have used it regularly ever since. I use it with 
 IDEFISK softphone for the most part but have tested it with Skype, X-Lite, 
 and SJPhone. I have had it since November and just love it.   Kerry Garrison
 Publisher - http://GeekGazette.com - http://VOIPSpeak.net 
  (949) 502-7819 x200 - [EMAIL PROTECTED]
 http://www.techdatapros.com 
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
 Sent: Wednesday, April 26, 2006 8:24 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] USB conference phone Kerry, do you actually own 
 one? Have you used it for long? What are you using it for?   (jim  
 personally I cant see the point of using your phone when I have a very good 
 quality headset and mic.). Dean   
  
  
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry 
 Garrison
 Sent: Wednesday, 26 April 2006 10:36 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] USB conference phone   This is an excellent USB 
 speakerphone 
 http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27   
  
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser
 Sent: Wednesday, April 26, 2006 6:26 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] USB conference phone I don't know about this 
 phone but I can tell you I have a vendor that will only talk to me via Skype 
 so I purchased this: 
 http://www.provantage.com/usb-internet-phone~220150620.htm   It operates nice 
 and has very good call quality.   
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
 Sent: Tuesday, April 25, 2006 8:22 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] USB conference phone Has anyone actually used these 
 USB speakerphones 
 http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
  Seems to get a pretty good review here  
 http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27   
   But looking for real world feedback. Cheers,   Dean   
  This e-mail and any attachments may contain confidential and privileged 
 information. If you are not the intended recipient, please notify the sender, 
 or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. 
 Any dissemination or use of this information by a person other than the 
 intended recipient is unauthorized and may be illegal. Unless otherwise 
 stated, opinions expressed in this e-mail are those of the author and are not 
 endorsed by the author's employer.  
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RE: [Asterisk-Users] Aasterisk large-scale deployment w/analog phones

2006-02-15 Thread mgraves
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]
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  Original Message 
 Subject: Re: [Asterisk-Users] Aasterisk large-scale deployment w/analog
 phones
 From: Dovid Bender [EMAIL PROTECTED]
 Date: Wed, February 15, 2006 3:58 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 I may be missing something here but why wouldnt ATA's
 work ? (other than cost).
 --- maka [EMAIL PROTECTED] wrote:
 
  hello,
  
  I am planning a fairly large hotel VoIP system,
  using analog phones. It will
  consist of about 100 analog phones, that must have
  access to a VoIP server.
  I am considering an option to use a couple of
  asterisk boxes, bundled with a
  total of four TDM2460E cards, and one TDM2451E card.
  
  Has anyone on this list done something similar? It
  would be great to hear
  some comments regarding a smilar setuyp/planning -
  Do you think is it better
  to distribute resources among multiple (more than
  two), lower-port-density
  asterisk servers? Or is it better to use a
  channelbank for that purpose?
  
  Cheers
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RE: [Asterisk-Users] Asterisk on laptop connected to POTS line

2006-02-02 Thread mgraves
  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:
 
   Anyone know of any equipment that I can use to connect a 
   laptop running asterisk to a POTS line (RJ11) ?
   
  Look at Xorcom's USB channel Bank.
 
 Which is a great product and you should all get one (and the fact that
 I'm a Xorcom employee has nothing to do with this recommendation), but
 sadly, still lacks FXO ports.

If Xorcom could offer something similar with 2-4 FXOs I'd just have to
buy at least one. Heck of an idea for a product, a quad FXO adapter
interfaced to Asterisk via local USB port. Wow!

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
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RE: [Asterisk-Users] Current viewpoints on the Sayson/Aastra 480i

2006-01-26 Thread mgraves
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 used.

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: [Asterisk-Users] Current viewpoints on the Sayson/Aastra 480i
 From: Mike Myers [EMAIL PROTECTED]
 Date: Thu, January 26, 2006 9:08 pm
 To: asterisk-users@lists.digium.com
 
 
 I have read the user reports on the Aastra 480i SIP phones, but the last 
 report is from October, and the advice was to stay away from these units.  I 
 was looking at the cisco 79XX series or the SPA-941's, but they don't have 
 backlit displays.  I also like POE support so I don't have to run multiple 
 wires to the phones and makes for a neater look.  I dislike the looks of the 
 SNOM series, so that's out.
 
 Have the bugs been worked out with the latest firmware, or are these still 
 phones to avoid.  
 
 thanks,
 Mike
 
 
 
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  With a free 1 GB, there's more in store with Yahoo! Mail.
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RE: [Asterisk-Users] Firewall/Embeded System/CF/etc

2006-01-23 Thread mgraves
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]
o(713) 861-4005
o(800) 905-6412
f(713) 864-8668
c(713) 201-1262



  Original Message 
 Subject: [Asterisk-Users] Firewall/Embeded System/CF/etc
 From: Manny A. Wise [EMAIL PROTECTED]
 Date: Mon, January 23, 2006 11:37 am
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 
 I am trying to build an silent non moving parts (fans,HD.etc) embedded
 system...Firewall/Asterisk/FXo/FXs/CF/etc
 
 Looking for anyone running asterisk with Coyote, IPcop, m0n0wal, Shorewall,
 etc in the same system/box!!!
 
 Offlist please...
 
 Thanks in advance!!
 
 Manny
 
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RE: [Asterisk-Users] Choosing an FXO card

2006-01-18 Thread mgraves
  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 card for my home Asterisk setup.  The question I 
 have is, should I got for a the Digium Wildcard FXO or settle for the 
 winmodem that Digium sell?  I have spoken to the UK distributor and they only 
 sell the Wildcard.  They said that the quality of the modem was questionable. 
  
 Does anyone have any experience of this?  Is the modem noticeably different 
 from the Wildcard in terms of quality?  Does it do things the modem doesn't?  
 Basically, it it worth putting out an extra £90 for the Wildcard?

Mike,

Save yourself the heartache and skip the local FXO if you can. I tried
several then just call forwarded the POTS line to an DID purchased from
an ITSP. Much better quality. No echo issues. No futzing about with
motherboards and IRQs.

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
f(713) 864-8668
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RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

2006-01-10 Thread mgraves
  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 out Hitachi IPC-5000 ?
 It looks nice and it's a bit exensive, but I would still like to hear
 how does it behave around Asterisk.
 

I used one for about 6 months. In general its well behaved with
Asterisk. It is a little tedious to get setup, but then you only need
to doit once. My issue with it were simple. The volume level was always
too low. Also, in order for a call to be sustained while moving between
access points they had to have the same SSID and be on the same
channel. That's less than ideal. Also, it useful range from the AP was
limited.

My assesment was that wireless SIP phone of any sort don't yet match a
decent cordless phone (Dect) plus an ATA.

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
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[Asterisk-Users] Skye gateway?

2005-09-23 Thread mgraves
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 direct approach? 

Some of my associates use Skype, which I'd prefer to avoid. However a
single gateway into my * server could be handy.

Thanks,

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
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RE: [Asterisk-Users] VoipJet Problems - anyone?

2005-08-18 Thread mgraves
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
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 I noticed their mysql server is down or can't connect to mysql server.
 I tried to download there cvs format price list.
 
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  Sent: Thu, 18 Aug 2005 16:04:30 -0400
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] VoipJet Problems - anyone?
 
  Hi,
  Does anyone know what is going on with voipjet?   This
  morning/afternoon they just seem to have gone down no word on
  their website.
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RE: [Asterisk-Users] will a firewall slow down asterisk?

2005-08-10 Thread mgraves

  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 be a problem.  My users conference using a voip line from an 
 ITSP so at any time there may be 4-8 calls passing over the firewall and 
 terminating in the MeetMe conference.  It works great.  I would recommend Pix 
 BTW.  Linksys would be my next rec.  But hey, they are both Cisco now...  8) 
   

I recommend m0n0wall (http://m0n0.ch/wall/)  which is a NetBSD based
firewall that includes traffic shaping. Easily managed via a web
interface. Runs on any decent PC with 2 or more NICs. Also on Soekris
or WRAP embedded platforms.

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
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RE: [Asterisk-Users] iax.cc / sixtel are they legitimate?

2005-01-27 Thread mgraves
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 experience with VoicePulse Connect.

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: [Asterisk-Users] iax.cc / sixtel are they legitimate?
 From: Jon Gabrielson [EMAIL PROTECTED]
 Date: Thu, January 27, 2005 8:21 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 Does anyone have any experience with iax.cc/sixtel?
 Are they a legitimate company?  From their website
 it looks like you can get a private incoming 800 
 number for 30 cents/month plus 2 cents/minute.  
 Somehow that pricing seems a little cheap for a 
 DID number.  I assume there has to be some minimum
 usage or something.  Any info as far as actual costs 
 and/or voice quality would be appreciated.
 
 
 Thanks,
 
 
 Jon.
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[Asterisk-Users] FXOs

2004-08-27 Thread mgraves
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 echo problems
that make it unusable. Sipura has not acknowledged the problem ( at
least to me) although several in the user community make refernce to
new firmware that might address the issue, real soon now.

I see a lot of activity recently on-list about the TDM-400. Of course,
mentions on-list are more than likely the result of people having
problems. We don't hear about people who have no issues with a product.

So, the nature of my inquiry is to explore how many people out here have
good/great experiences with the various small FXO adapters? While the
TDM-400 is my next possible purchase I'd also like to hear about
devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc.
With so many products being offered I would hope that we have some
collective experience with each one.

Thanks,
Michael



Michael Graves
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Pixel Power Inc
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[Asterisk-Users] DISA and notransfer/reinvite?

2004-07-29 Thread mgraves
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. 

A support engineer calls in to a DID that I have from VoicePulse
Connect. They match the password and dial out to the UK...also via VPC.
This I know to work. However, if I allow SIP reinvite then the call, in
theory, leaves my server completely for the rest of its duration. Or
does it? Is this sensible? Are there issues that I should be concerned
about in such an arrangement?

Thanks,

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



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RE: [Asterisk-Users] Options for 3+ FXO ports

2004-03-08 Thread mgraves

  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
 incoming FXO lines.  Based on my reading of the mailing list archives
 and
 what I read on the Digium web site, I don't think that this can be
 achieved
 with X100P cards as it doesn't seem to be practical to have more than
 3
 X100P cards at the time (IRQs is the main problem I think).  At four
 FXO
 ports, I can't quite justify something like a full T1 on a monthly
 basis.
 But, since I've never ordered one, I'm thinking that it's a all or
 nothing
 deal (i.e.: I can't order a T1 and pay for only 4 voice channels). 
 It would
 seem that the best I can do is a 3x8 configuration according to the
 information on this page
  
  
  The Mediatrix 1204 FXO gateway sort of works with asterisk in the US,
 but
  have had a few issues with CallerID being recognized when the four
 pstn
  lines have different ring cadence (at least I believe its a cadence
 issue).
  
  Other then that, there seems to be a significant market opportunity
 for
  anyone that can produce a reliable 4-port device.
  
  A goggle search for channel banks turned up www.nextag.com with
 samples such
  as this:
  Adtran TA750 Chassis w/psu ac adapter $900
  Adtran Quad FXO card: $230
  Adtran TA750 w/12 FXS ports: $1250
  Adtran TA750 w/FT1, 20 FXS: $1900
  
  However, I don't consider those prices cost effective for four ports
 after
  adding in the digium T1 card, etc.
  
  I'm interested in a 4-port as well if anyone knows of something.
 
 See: http://www.welltech.com.tw/
 
 No yet tested, but waiting my first samples next week. Have callerid
 and 
 reversal polarity detection!. Prices seems to be half of Mediatrix.
 
 Jorge

I have been eagerly awaiting word on these devices! They would not give me one on 
evaluation, and charge a 20% restocking so I have not tried them myself. Please post 
your findings :-)

Michael Graves

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[Asterisk-Users] IAX image for SNOM 200?

2004-03-03 Thread mgraves
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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RE: [Asterisk-Users] Web based UA

2004-02-25 Thread mgraves
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


  Original Message 
 Subject: Re: [Asterisk-Users] Web based UA
 From: Jonathan Moore [EMAIL PROTECTED]
 Date: Wed, February 25, 2004 11:16 am
 To: [EMAIL PROTECTED]
 
 I think xten is supposed to have an active X control version of their
 softphone that would probably do what you are talking about.
 
 
 On Wed, 25 Feb
 2004, Michael Graves wrote:
 
  Hello All,
  
  Does anyone here have any experience with web based soft clients for
 *?
  I'm thinking about putting a page up on our corp web server that
 would
  let staff in the field connect to our in-house phone system via the
  internet. This could help staff making overseas calls while on
 trips,
  without demanding that they use a particular laptop/soft phone. They
  could use an PC on a broadband connection.
  
  Thanks,
  
  Michael
  
  --
  Michael Graves   [EMAIL PROTECTED]
  Sr. Product Specialist  www.pixelpower.com
  Pixel Power Inc. [EMAIL PROTECTED]
  
  It is dangerous to be correct about matters when the established 
  authories are wrong. - Voltaire
   
  ** Tag(s) inserted by Bandit Tagger98 -
 http://www.gbar.dtu.dk/~c918704
  
  
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 -- 
 Jonathan Moore
 Technology Coordinator
 Winfield Public Schools
 Office 316-221-5100
 Fax 316-221-0508
 
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[Asterisk-Users] Pingtel Phones?

2004-02-15 Thread mgraves
 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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