from:"Jessee J Holmes"

Re: [asterisk-users] Hot GXP-2000

2007-06-08 Thread Jessee J Holmes


Carlos,

We had this happen once here with a batch of phones received from  
Grandstream about a year ago now. Email Grandstream on it and they  
should know exactly what the problem is, I believe they ended up  
replacing the phones for us.



Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Jun 8, 2007, at 10:47 AM, Carlos Chavez wrote:


This is off topic for Asterisk but I need a suggestion.  I have a
customer (travel agency) that has recently begun complaining that  
their

GXP-2000 phones are getting very hot, they say that around mid day the
handset is so hot that it can burn your ear.  These phones are in
constant use and right now the weather in Mexico is hot.

	Anyone know why the handset would get so hot?  Only the phones  
assigned

to sales get this way and the others in the office do not have this
problem.  They are all connected to the same Linksys PoE switch for
power.

--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Where to find Polycom firmware with 330/320 support?

2007-06-06 Thread Jessee J Holmes

Version C is the latest available and for now is only available  
through your reseller or service provider.


Version C supports the new 320s and 330s. The bootrom got updated  
too, I think version 3.2.3b is the latest out there at this time.


Polycom normally only publicly lists the prior version of the  
firmware only (1.6.7 and 2.0.3b at this time)


Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Jun 6, 2007, at 8:29 AM, Fuermann, Jason Bryce wrote:

Looks like they haven't worked out all the links yet (they just  
redid their site).
http://www.polycom.com/common/documents/support/downloads/voice/ 
spip_ssip_sip_2_0_3b_sig.zip is one gen behind and can be  
downloaded without a reseller account.


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk- 
[EMAIL PROTECTED] On Behalf Of Mandeep Singh Bhabha

Sent: Wednesday, June 06, 2007 5:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Where to find Polycom firmware with  
330/320 support?


I just copied these files from given link.

On Tue, Jun 05, 2007 at 06:35:15PM -0600, Stephen Bosch wrote:

Fuermann, Jason Bryce wrote:
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/ 
soundpoint_ip330_320.html


This only works if you have a reseller account.

-Stephen-
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


--
With Regards,
Mandeep Singh Bhabha


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Microsoft's Move Into IP PBX Market

2007-05-17 Thread Jessee J Holmes


I want the "hot-dog stand" theme on my phone :)

On May 17, 2007, at 10:03 AM, Drew Gibson wrote:


George Pajari wrote:

From c|net News:
"On Monday,Microsoft and nine leading phone manufacturers--Asustek  
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung,  
Tatung, and Vitelix--announced the public beta program for  
Microsoft Office Communications Server 2007 and Microsoft Office  
Communicator 2007."


http://news.com.com/8301-10784_3-9719931-7.html? 
part=rss&subj=news&tag=2547-1_3-0-20



I'm told the main hardware requirements are
1.  Colour screens (well, _blue_ anyway)
2.  Support for the prompt "Error bridging call, (A)bort, (R)etry,  
(F)ail? _"


regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys - new info

2007-04-12 Thread Jessee J Holmes


Mike,

Got off the phone with Polycom on this  I have the same problem  
with my new 601 phone here (haven't seen the problem on the 650).


I'm trying to find answers and Polycom's only got one reported case  
of this (which I find bazaar, but whatever). The problem was  
resolved, the problem was the user was using 1.6.5 configuration  
files with the 2.x firmware. Once the user put the config files that  
are sent with the firmware, problem went away.


I cannot stress this enough, as minuet or insignificant a change may  
appear in the configuration files, or as similar as they look, use  
the ones provided with the firmware!


Now, I'm not sure if we've done this ourselves, but I'm having one of  
our support guys today looking at it and getting new configuration  
files into our phones. I'll see if that resolves the problem or not.  
I may not have an answer back until Friday or Monday, but if any of  
you guys experiencing this issue want to try as well, be my guess.  
Mentioning this now only because this is the information that came  
from Polycom support as a resolved problem.



Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Apr 12, 2007, at 10:37 AM, Mike wrote:

I found *something*.  I've gone into my CPU graph (on the phone, in  
status - diagnostic).  Two phones, one running 1.6.7 and one  
running 2.1.0, both on the same Hub, with the same general  
configuration (different SIP registration, and each using it's  
version-specific sip.cfg file).


The pre-2.x phone is running with CPU load approaching 0% (0%-7%).   
The 2.x phone has tons of spikes in the 100% range.


What could be causing this?  Where do I start looking?

Mike

From: [EMAIL PROTECTED] [mailto:asterisk- 
[EMAIL PROTECTED] On Behalf Of Jason Fuermann

Sent: Thursday, April 12, 2007 10:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 501 issue with latest  
firmware: sluggishkeys


also I've seen that not having the correct version of sip.cfg and  
phone1.cfg could cause weird problems. Make sure you are using the  
ones that came with the firmware.


Mike wrote:
Exactly.  It's a weird issue, and I can't imagine what the problem  
is,
except maybe for bad phones (but then again, why would the phones  
be only

bad with 2.x?)

UnlessI have bootrom 3.2.2.0019.  Is that what people running  
thelatest

have?

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of  
Jerry Jones

Sent: Thursday, April 12, 2007 00:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware:
sluggish keys

It has nothing to do with actually dialing. Even trying to press  
end call or

the speakerphone button does not work at times.

Have tried removing side cars etc, but definately seems to be a  
bug in the

2.x code stream.


On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote:



Jim King wrote:

I've seen an issue like this from time to time on 601s, even  
with the
latest firmware.  Not just the softkeys, but also the dial  
keys.  The

phones seem to "run slow" sometimes, failing to respond to a key
press right away but getting to it eventually.  It usually  
clears up

after a few seconds.
Also, I've noticed that the 601s sometimes ignore key presses
altogether, just as you describe.
I have not yet found a solution for this problem...


Try setting this in sip.cfg:  dialplan.impossibleMatchHandling="1"

I suspect it is either 0 or 2 now.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 330/320

2007-04-11 Thread Jessee J Holmes

Honestly, I can't remember and I've seen them ... but should be a bit  
better.


IP 301 phone - 4 line x 20 character monochrome display
IP 320/330 - 102 x 33 pixel graphical LCD

Not backlit from what I can see still (correct me if I'm wrong).

and heads up ... NO POWER SUPPLY'S ARE INCLUDED! (Just trying to  
prevent people from making a mistake, since this is the first time  
Polycom will be doing the Cisco approach to the power supply).



Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Apr 11, 2007, at 1:40 AM, Andrew Joakimsen wrote:


How is the screen compared to the other Polycom products?

On 4/9/07, Jessee J Holmes <[EMAIL PROTECTED]> wrote:

Mike,

I don't have much information, except they are due for shipment  
soon (mid to
end of April to distribution from Polycom). We've demoed a couple  
and I
personally believe they'll be a tough phone to find in stock for  
the first
few months their released. Demand on these from what I'm seeing  
right now is
very, very high. I think they are a great addition to the family  
and most

importantly  they have FULL DUPLEX SPEAKERPHONE! :)

550's are released products though.




Jessee Holmes

Atacomm / Ataractic Corporation

www.atacomm.com

V: 1-877-700-VOIP

[EMAIL PROTECTED]




Looking for voice over IP products?  Visit our VoIP store at
http://voipstore.atacomm.com/



On Apr 9, 2007, at 3:55 PM, Mike wrote:

Ah, thanks.  I didn't realize this.

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Darren
Nickerson
Sent: Monday, April 09, 2007 16:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 330/320

Mike [EMAIL PROTECTED] wrote:


How do you guys like the 330 and 320?

Mike,

As far as I am aware, neither of these handsets are presently  
shipping from
Polycom, so most people's experience will be limited to PDF  
brochures and
pretty pictures. On the face of it, this looks like a good  
alternative to
the IP301 since it adds native 802.3af PoE support. Not sure yet  
exactly

where the pricing will slot in, however.

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 330/320

2007-04-09 Thread Jessee J Holmes


Mike,

I don't have much information, except they are due for shipment soon  
(mid to end of April to distribution from Polycom). We've demoed a  
couple and I personally believe they'll be a tough phone to find in  
stock for the first few months their released. Demand on these from  
what I'm seeing right now is very, very high. I think they are a  
great addition to the family and most importantly  they have FULL  
DUPLEX SPEAKERPHONE! :)


550's are released products though.

Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Apr 9, 2007, at 3:55 PM, Mike wrote:


Ah, thanks.  I didn't realize this.

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Nickerson
Sent: Monday, April 09, 2007 16:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 330/320

Mike [EMAIL PROTECTED] wrote:


How do you guys like the 330 and 320?


Mike,

As far as I am aware, neither of these handsets are presently  
shipping from
Polycom, so most people's experience will be limited to PDF  
brochures and
pretty pictures. On the face of it, this looks like a good  
alternative to
the IP301 since it adds native 802.3af PoE support. Not sure yet  
exactly

where the pricing will slot in, however.

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] polycom repair

2007-04-05 Thread Jessee J Holmes


James Andrewartha,

We send these into Polycom for repair on occasion. It's will cost  
roughly $200~$300 direct with Polycom and will take 8~10 weeks for  
repair.
Obviously, most people opt not to do a repair and just buy a new  
phone instead. The only time it may make sense to repair an out of  
warranty phone would be if you have some sentimental attachment to  
your current phone or more realistically, you're not certain the  
newer phones will work with your PBX or SP. There may be other  
reasons, but I can't think of those right now. Keep in mind, IP 600's  
aren't available any longer from Polycom. The new replacement is the  
IP 601 phone or the IP 650.


Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Apr 4, 2007, at 11:36 PM, James Andrewartha wrote:


Hi all,

Has anyone had any experience getting Polycom phones repaired? The  
screen on
one of our IP600s got smashed, and I'm wondering if it's worth the  
effort to

get it repaired, or if it'd just be cheaper to buy a new phone.

Thanks,

--
James Andrewartha
Systems Administrator
Data Analysis Australia Pty Ltd
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Snom 320 password

2007-02-21 Thread Jessee J Holmes


Mike,

A few things you can try, default administrator password should   
by default. Maybe it just needs that entered.


Otherwise, if the phone is being used with Asterisk, there was a bug  
on an issue like this which may have since been resolved, but non-the- 
less is documented here: http://voipstore.atacomm.com/Support/KB/ 
ViewArticle.aspx/27934028032-1-10.htm


Secondly, if this doesn't work, I'd really suggest the simple  
routine, manufacturer reset followed by the latest stable firmware  
release upgrade.


Instructions for a factory reset can be found here: http:// 
voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-9.htm


Hope that helps,

Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Feb 21, 2007, at 4:52 PM, Mike Hammett wrote:

A client of mine has a Snom 320.  Usually when he comes in each  
morning, it is asking him for a password.  A power cycle brings it  
back to normal operation.  How do I troubleshoot this further?




--Mike







___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Grandstream GXW-4108 8 port FXO

2006-12-21 Thread Jessee J Holmes


Chris,

These devices are still very new to the market. Finding reviews on  
them may be tough still. From our experience its a good little device  
for the dollar; but, keep in mind, it's still a low cost gateway and  
that normally means don't expect too much.


We've sold few cases here and response on them have been both good  
and bad. Grandstream for a while stopped shipping any units they had  
due to firmware problems on the unit. Quality of this product is  
driven a lot off of the firmware alone unfortunately as well as the  
environment the units are being used in. However, supposedly  
Grandstream began shipping the unit again recently and released new  
firmware on many of their products which they have boasted to us as  
being "much better".


That being said, I can't say whether this unit works great, good, or  
poorly at this time. I would say it could work very well for a budget  
conscious,small office or home office setup.


I'd like to see some good user reviews on this unit from people that  
have it running in a live environment. We've only played with it here.


Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Dec 21, 2006, at 3:29 PM, cb wrote:

Has anyone used either the 8 port or 4 port FXO device from  
Grandstream? (GXW-4108 or 4104).


They seem to be the lowest cost multi port FXO devices that I can  
find, so I'm getting ready to buy the 8 port version. I just want  
to see if there are any opinions on the device before I commit to  
the purchase.


If people have not used the Grandstream, are there any issues with  
using similar devices (that is, FXO devices that connect to the  
Asterisk server via SIP over Ethernet).



I am looking to connect at least 8 PSTN lines, and as many as 12 or  
16 to Asterisk (Currently using Trixbox, but I'm also looking at  
either AsterixNow or just building from scratch on a bare linux  
box). Money is a major concern in my purchases, which is why I'm  
looking at the Grandstream (even used on ebay, I don't seem to be  
able to find 8-16 port FXO devices for less than the approx $50 per  
port the Grandstream will get me... plus it has a video input for a  
security camera which is just a plus to me as installing a web  
capable surveillance camera at the location is on my to do list).


-chris



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Audiocodes MP-114 noise

2006-11-08 Thread Jessee J Holmes

Jason,I think it's something only supported in the newer firmware. Get in contact with the place you bought the unit from, they should be able to get the latest firmware for you.MP118_SIP_F4.80A.034.004.cmp should work for MP114, this is what we used and all that we could get from Audiocodes, I think they should have named it MP11x though. :) Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 8, 2006, at 2:25 AM, Jason Kim wrote:Jessee,Thank you for your help.I downloaded firmware and sample configuration files.But the firmware was old version for MP118 and MP124.Where can i download recent one?Can i upload only ini file to changecountrycoefficient ?Regards,Jason.--- Jessee J Holmes <[EMAIL PROTECTED]> wrote: Jason,First, before you start reading, get to the latestfirmware from  Audiocodes (MP118_SIP_F4.80A.034.004.cmp), therehave been  significant echo improvements in this version.After many days of working with Audiocodes on thisproblem and much  time spent here by multiple technicians trying toreproduce and  resolve this issue; this morning, Atacomm receivedan email from  Audiocodes with a full explanation to this nowconfirmed issue with  all MP-11x units. Atacomm will immediately beginwork on a KB article  within our website that confirms this issue andoutlines the  manufacturer recommended steps to resolve thisproblem.Apparently, there have been some changes with theMP-11x's that can  negatively affect line noise and echo.  Below aresome steps which  can help to correct these problems:1. The new design did away with the Coefficent file. Audiocodes, now  instead, introduced a configurable parameter called countrycoefficient. This parameter can be adjustedto a specific  country based on known configurations.  For the mostpart this should  work.  70(USA) is the default value.  More can befound in the User’s  manual.2.  In just about every case, an FXO is added to aPre-existing PBX  or CO line, you can expect echo. This comes from thefact that delay  (IP Network) is being introduced, and what used tobe Side tone is  now delayed so much it is echo. Just about everydifference on the  line that can be heard between the pre fxo and postfxo installation  can be traced to echo, or line quality issues.3.  Going forward, Audiocodes would like to suggestthat when  installing the product do the following:A) Make sure the Line coming from the PBX or CO is aLoop Start line.  Ground start is not supported on the MP-11x seriesof gateways. (The  M1K FXO will in 5.0)B) Check that the Line can deliver for a 600 OhmImpedance line-52 to -24 V of Off Hook Voltage-15 to -6 V of  On Hook Voltage20 to 35 ma of loop current.If you know the line is not 600 Ohm, please gathermetrics on the  line, and the make and model of the PBX or switch itis attached too,  plus country of origin. If it is not from the USA,please look up the  country of origin and then find theCountryCoefficient to match this.  Load the .ini file to the board with this settingand reset.  Make  sure the Gateway has a firmware version of 4.60.035or higher or  4.80.030 or higher.C) Put the device on the network with Voice Volumeset to 0 and input  gain set to 0. Make calls, if there is no issue, youcan stop here.   However, Echo is still expected most of the time.D) The echo should be heard by the IP sideparticipant as their voice  is reflected back.  If this is the case, then whatneeds to be done  is to lower the voicevolume (IP—TEL). This way thespeaker’s  reflected voice will comeback low enough for theECAN to cancel it  out (-6 is usually recommended as the value to plugin here). A  little experimentation is needed as the loss for alllines will vary  based on length from the CO. Echo is usually takencare of in this  manner.E) The incoming speaker from the PSTN’s voice seemslow, set  InputGainLocation =1, and then slowly increment theInput Gain  Parameter(Tel?P) to adjust for this. In pastreleases (see the note  about loads above), the input gain was alwaysapplied prior to the  ECAN which had the effect of amplifying the returnedecho and noise  on the line causing crosstalk and clipping issues.This is no longer  the case.If the above does not resolve the issues, then youneed to go ahead  and collect DSP, Ethereal and Syslog traces alongwith the board.ini,  these are to be sent to your support agent, who willthen send these  to Audiocodes for their engineers to evaluate.  Thisshould not  happen often.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIPstore at http:// voipstore.atacomm.com/On Nov 3, 2006, at 12:14 AM, Jason Kim wrote:Jessee,I tried many combinations of "Voice Volume", "InputGain" and packetization time , but it's noisy steel.I'm using G.711A-law and packetization time is 2

Re: [asterisk-users] Audiocodes MP-114 noise

2006-11-07 Thread Jessee J Holmes

Jason,First, before you start reading, get to the latest firmware from Audiocodes (MP118_SIP_F4.80A.034.004.cmp), there have been significant echo improvements in this version.After many days of working with Audiocodes on this problem and much time spent here by multiple technicians trying to reproduce and resolve this issue; this morning, Atacomm received an email from Audiocodes with a full explanation to this now confirmed issue with all MP-11x units. Atacomm will immediately begin work on a KB article within our website that confirms this issue and outlines the manufacturer recommended steps to resolve this problem.Apparently, there have been some changes with the MP-11x's that can negatively affect line noise and echo.  Below are some steps which can help to correct these problems:1. The new design did away with the Coefficent file.  Audiocodes, now instead, introduced a configurable parameter called countrycoefficient. This parameter can be adjusted to a specific country based on known configurations.  For the most part this should work.  70(USA) is the default value.  More can be found in the User’s manual.2.  In just about every case, an FXO is added to a Pre-existing PBX or CO line, you can expect echo. This comes from the fact that delay (IP Network) is being introduced, and what used to be Side tone is now delayed so much it is echo. Just about every difference on the line that can be heard between the pre fxo and post fxo installation can be traced to echo, or line quality issues.3.  Going forward, Audiocodes would like to suggest that when installing the product do the following:A) Make sure the Line coming from the PBX or CO is a Loop Start line. Ground start is not supported on the MP-11x series of gateways. (The M1K FXO will in 5.0)B) Check that the Line can deliver for a 600 Ohm Impedance line-52 to -24 V of Off Hook Voltage-15 to -6 V of  On Hook Voltage20 to 35 ma of loop current.If you know the line is not 600 Ohm, please gather metrics on the line, and the make and model of the PBX or switch it is attached too, plus country of origin. If it is not from the USA, please look up the country of origin and then find the CountryCoefficient to match this. Load the .ini file to the board with this setting and reset.  Make sure the Gateway has a firmware version of 4.60.035 or higher or 4.80.030 or higher.C) Put the device on the network with Voice Volume set to 0 and input gain set to 0. Make calls, if there is no issue, you can stop here.  However, Echo is still expected most of the time.D) The echo should be heard by the IP side participant as their voice is reflected back.  If this is the case, then what needs to be done is to lower the voicevolume (IP—TEL). This way the speaker’s reflected voice will comeback low enough for the ECAN to cancel it out (-6 is usually recommended as the value to plug in here). A little experimentation is needed as the loss for all lines will vary based on length from the CO. Echo is usually taken care of in this manner.E) The incoming speaker from the PSTN’s voice seems low, set InputGainLocation =1, and then slowly increment the Input Gain Parameter(TelàIP) to adjust for this. In past releases (see the note about loads above), the input gain was always applied prior to the ECAN which had the effect of amplifying the returned echo and noise on the line causing crosstalk and clipping issues. This is no longer the case.If the above does not resolve the issues, then you need to go ahead and collect DSP, Ethereal and Syslog traces along with the board.ini, these are to be sent to your support agent, who will then send these to Audiocodes for their engineers to evaluate.  This should not happen often.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Nov 3, 2006, at 12:14 AM, Jason Kim wrote:Jessee,I tried many combinations of "Voice Volume", "InputGain" and packetization time , but it's noisy steel.I'm using G.711A-law and packetization time is 20ms.It can be impedance mismatch problem but i cannotadjust impedance of FXO port of MP-114.___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Audiocodes MP-114 noise

2006-11-03 Thread Jessee J Holmes

Jason,Funny enough, I'm working on this exact same problem right now with one of our customers and we're scheduling a conference call with Audiocodes today. I'll get back to you on what I found out. However, in the mean time, you may want to take it up with the place you bought the unit from, they can go through these same troubleshooting procedures (maybe something I forgot to cover), and will eventually take it up with AC themselves. A DSP recording may help determine where the problem lies within the device or possibly network; however, this needs to be done only in extreme measures. Contact your support agent (reseller or service provider), get them logs, config ini's, network diagrams, IP PBX information and configuration, whatever you can so they are fully aware of your network and they should be able to assist troubleshooting this issue more effectively than I can here.This part I can't really help you with on list, you're support agent would need to assist you from here on this matter; however, I'll let you know more information when I have it from Audiocodes when we get a resolution on our end for our customer. Maybe, hopefully, it will just be a quick tweak.I'll keep you informed what I find out on my end. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 3, 2006, at 12:14 AM, Jason Kim wrote:Jessee,I tried many combinations of "Voice Volume", "InputGain" and packetization time , but it's noisy steel.I'm using G.711A-law and packetization time is 20ms.It can be impedance mismatch problem but i cannotadjust impedance of FXO port of MP-114.--- Jessee J Holmes <[EMAIL PROTECTED]> wrote: Jason,There are a couple things we can try to fix yourproblem.Your firmware shouldn't be an issue, but latest I'vegot now is:  MP118_SIP_F4.80A.034.004.cmpLet's try some quick things first though:In your web interface, go to advanced config -channel settings /  voice settingsThere are some options here you can play with:"Voice Volume" (IP side of this thing) - by defaultthis should be  set at '1'. Try bringing this down slowly, I'd sayin increments of 5  (-4, then -9, and so on).Range on this option can be anywhere from -32 to+32, you really  shouldn't need to go beyond -15; but you're actualvolume on the  calls should still stay reasonable."Input Gain" (telco side) is another option you canslowly change as  well (set to 0 by default).There should also be spot where you can specify the"codername", you  could possibly try changing this to another codecsuch as G.729 or G. 711u-law (should be the same codec being used onyour Asterisk  system) try changing packet size from 20 to 40 or60. This may also  help.If none of this stuff helps, let me know. We canthen start getting  really technical.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIPstore at http:// voipstore.atacomm.com/On Nov 1, 2006, at 1:45 AM, Jason Kim wrote: Thank you Jessee,Firmware seems to be recent(4.80A.025.004).For 'noisy', I mean IP Phone <--> * <--> MP-114 side. Audio quality of MP-114 <--> PSTN <--> Analog phone is good.I think it can be power ground or gain problem.Any experience?Thanks,Jason--- Jessee J Holmes <[EMAIL PROTECTED]> wrote: Dear Jason,Please define better noisy? You talking echo issues? Is it on justyour side or on the called party's side as well?This start happening immediately, or was the boxworking before andthe problem just started?Also, a quick heads up, make sure before evenbeginning totroubleshoot an issue like this you do a factoryreset to the unitand get the latest available firmware on it. Usually that fixesannoying issues like this.Thanks,Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Oct 30, 2006, at 10:36 PM, Jason Kim wrote: It's noisy while talking.Any idea?Thanks in advance.Jason__  __Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com)___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users  ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asteri

Re: [asterisk-users] Polycom latest version

2006-11-02 Thread Jessee J Holmes

You're reseller or service provider.Or Polycom if you are VoIP certified with them.2.0.2 is the latest. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 2, 2006, at 1:38 PM, Mike wrote: Hi,   Where should I go to get the Polycom`s latest official (non-beta) version?  I am registered on the Polycom customer website but that doesn't seem accessible.   Regards,   Mike___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 601 Phone can not find TFTP server

2006-11-02 Thread Jessee J Holmes

This may help,http://voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-24.htmSounds like you don't have a configuration file on your TFTP server. After that message does it say anything about "Error loading (mac address).cfg!" and will then reboot?If so, get the MAC.cfg file in there. The article should explain this better. Not a guaranteed fix, but should help you locate and fix the problem.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 2, 2006, at 12:51 AM, Klaverstyn, David C wrote: Can someone please help me with a problem that I seem to have with this Polycom 601 phone.  It will not see my TFTP server and keeps saying “Could not contact boot server, using existing configuration”.  I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Snom or Cisco Phones?

2006-11-02 Thread Jessee J Holmes

CP,I've never heard complaints about the Snom 320 speakerphone hardware, nor do I think they sound bad from using the phones myself. I believe Snom did make a significant improvement to their speakerphone hardware not to long ago.Of course, there is never any guarantee on the "quality" of something, since I don't think something can please everyone. Maybe other users of this phone can post their feedback here as well and based upon that you can get a good idea of what your results should be. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 5:24 PM, cp wrote: Do the speakerphone’s work well on Snom 320’s? I have a Linksys 841 and could never get the speakerphone working well. -CP From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, November 01, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom or Cisco Phones? Writing this as a user of VoIP and not a reseller, (meaning off the record), we really love the Snom phones here as well, I wish the Snom 300's had a bit more functionality (like the Grandstream), and the Snom 320's and 360's were a little less confusing with their buttons (aka too many buttons on the keypad for some non-tehcy end users). But they have terrific functionality and great audio quality in most office environments, and are very easy to set up and install. Everyone seems to really love them. The Cisco phones are nice as well, but IF you decide to go with Cisco, READ what you are buying and what you are getting before just blindly buying it (in fact, do this anyways, it's common sense to do this before buying any product, anywhere). Cisco products normally don't come with half of the items you need, and unfortunately most resellers (and Cisco) don't make this too easy to read and understand. DO NOT buy refurbished Cisco if you want support, especially since there has been some bogus Cisco voice equipment shipping lately from some of the certified Cisco resellers/distributors. Network World had an article on this recently: http://www.networkworld.com/news/2006/102306counterfeit.html Cisco may have a great look to their phones and have the design very well thought out (not to mention the big Cisco name - which is good enough for some), but they are normally harder to install and configure and are VERY proprietary. If you buy Cisco, Cisco wants you to ONLY buy Cisco (for support and marketing reasons). Snom 320's are a great choice just because these phones mainly support everything the Snom 360's support (i.e. sidecars) Only main differences between these two models is that the Snom 360 has the larger LCD screen as well as newly added XML support. We have about 50 stations here, some management, some support, some sales and have pretty much decided as a company to completely use Snom phones for all of our employees. Keep in mind, each phone out there will have their specific pro's and con's, as well as quarks ... seems there is no real "perfect phone" out there yet. But Snom in my mind, is pretty dang close. This all of course is just personal opinion from past experience. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 8:35 AM, Tom Vile wrote: I love the Snom phones as well. The function keys are great and easy to use. On 10/31/06, mitcheloc <[EMAIL PROTECTED] > wrote:My vote is definitely for Snom, I've worked with Cisco phones for years, but the Snom is much better integrated, and the feature buttons can be retooled for any environment, making custom installs very easy. On 10/31/06, Conrad Wood <[EMAIL PROTECTED]> wrote: > On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote: > > Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia > > cars > > Not really. Both are very good phones. > > * My Clients prefer cisco because it looks more business-like. - The new > snom phones do look better though and the side car rules. > * The Cisco phone 'feels' very good in your hand, and the voicequality > is superb. (I'd say slightly better than that of the snom 360) > > * Technically, I find the snom phone more advanced and I can do more > cool stuff with it - Cisco doesn't seem to like giving features away in > SIP. > * Snom phones, for example, have freely programmable buttons that can > park/retrieve/transfer calls, show line status etc. I can't get that to > work with Cisco phones at all. > * Putting custom ringtones (a

Re: [asterisk-users] Audiocodes MP-114 noise

2006-11-01 Thread Jessee J Holmes

Jason,There are a couple things we can try to fix your problem.Your firmware shouldn't be an issue, but latest I've got now is: MP118_SIP_F4.80A.034.004.cmpLet's try some quick things first though:In your web interface, go to advanced config - channel settings / voice settingsThere are some options here you can play with:"Voice Volume" (IP side of this thing) - by default this should be set at '1'. Try bringing this down slowly, I'd say in increments of 5 (-4, then -9, and so on).Range on this option can be anywhere from -32 to +32, you really shouldn't need to go beyond -15; but you're actual volume on the calls should still stay reasonable."Input Gain" (telco side) is another option you can slowly change as well (set to 0 by default).There should also be spot where you can specify the "codername", you could possibly try changing this to another codec such as G.729 or G.711u-law (should be the same codec being used on your Asterisk system) try changing packet size from 20 to 40 or 60. This may also help.If none of this stuff helps, let me know. We can then start getting really technical.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 1:45 AM, Jason Kim wrote:Thank you Jessee,Firmware seems to be recent(4.80A.025.004).For 'noisy', I mean IP Phone <--> * <--> MP-114 side.Audio quality of MP-114 <--> PSTN <--> Analog phone isgood.I think it can be power ground or gain problem.Any experience?Thanks,Jason--- Jessee J Holmes <[EMAIL PROTECTED]> wrote: Dear Jason,Please define better noisy? You talking echo issues?Is it on just  your side or on the called party's side as well?This start happening immediately, or was the boxworking before and  the problem just started?Also, a quick heads up, make sure before evenbeginning to  troubleshoot an issue like this you do a factoryreset to the unit  and get the latest available firmware on it. Usuallythat fixes  annoying issues like this.Thanks,Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIPstore at http:// voipstore.atacomm.com/On Oct 30, 2006, at 10:36 PM, Jason Kim wrote: It's noisy while talking.Any idea?Thanks in advance.Jason  __  __Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com)___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users  ___ --Bandwidth and Colocation provided by Easynews.com--asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users  Low, Low, Low Rates! Check out Yahoo! Messenger's cheap PC-to-Phone call rates (http://voice.yahoo.com)___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Snom or Cisco Phones?

2006-11-01 Thread Jessee J Holmes

Writing this as a user of VoIP and not a reseller, (meaning off the record), we really love the Snom phones here as well, I wish the Snom 300's had a bit more functionality (like the Grandstream), and the Snom 320's and 360's were a little less confusing with their buttons (aka too many buttons on the keypad for some non-tehcy end users).But they have terrific functionality and great audio quality in most office environments, and are very easy to set up and install. Everyone seems to really love them.The Cisco phones are nice as well, but IF you decide to go with Cisco, READ what you are buying and what you are getting before just blindly buying it (in fact, do this anyways, it's common sense to do this before buying any product, anywhere). Cisco products normally don't come with half of the items you need, and unfortunately most resellers (and Cisco) don't make this too easy to read and understand. DO NOT buy refurbished Cisco if you want support, especially since there has been some bogus Cisco voice equipment shipping lately from some of the certified Cisco resellers/distributors. Network World had an article on this recently: http://www.networkworld.com/news/2006/102306counterfeit.htmlCisco may have a great look to their phones and have the design very well thought out (not to mention the big Cisco name - which is good enough for some), but they are normally harder to install and configure and are VERY proprietary. If you buy Cisco, Cisco wants you to ONLY buy Cisco (for support and marketing reasons).Snom 320's are a great choice just because these phones mainly support everything the Snom 360's support (i.e. sidecars) Only main differences between these two models is that the Snom 360 has the larger LCD screen as well as newly added XML support.We have about 50 stations here, some management, some support, some sales and have pretty much decided as a company to completely use Snom phones for all of our employees.Keep in mind, each phone out there will have their specific pro's and con's, as well as quarks ... seems there is no real "perfect phone" out there yet. But Snom in my mind, is pretty dang close.This all of course is just personal opinion from past experience.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 8:35 AM, Tom Vile wrote:I love the Snom phones as well.  The function keys are great and easy to use.On 10/31/06, mitcheloc <[EMAIL PROTECTED] > wrote:My vote is definitely for Snom, I've worked with Cisco phones foryears, but the Snom is much better integrated, and the feature buttons can be retooled for any environment, making custom installs very easy.On 10/31/06, Conrad Wood <[EMAIL PROTECTED]> wrote:> On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote: > > Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia> > cars>> Not really. Both are very good phones.>> * My Clients prefer cisco because it looks more business-like. - The new > snom phones do look better though and the side car rules.> * The Cisco phone 'feels' very good in your hand, and the voicequality> is superb. (I'd say slightly better than that of the snom 360)> > * Technically, I find the snom phone more advanced and I can do more> cool stuff with it - Cisco doesn't seem to like giving features away in> SIP.> * Snom phones, for example, have freely programmable buttons that can > park/retrieve/transfer calls, show line status etc. I can't get that to> work with Cisco phones at all.> * Putting custom ringtones (and choosing which ones to use) is a> no-brainer with snoms and real trouble with ciscos. > * On ciscos, I find the "upgrade" path from sccp to sip a totally> unnecessary annoyance. ___> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list> To UNSUBSCRIBE or update options visit:>   http://lists.digium.com/mailman/listinfo/asterisk-users >--Mitchel ConstantinSnap - A desktop user interface for Asteriskwww.snapanumber.com___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Audiocodes MP-114 noise

2006-10-31 Thread Jessee J Holmes

Dear Jason,Please define better noisy? You talking echo issues? Is it on just your side or on the called party's side as well?This start happening immediately, or was the box working before and the problem just started?Also, a quick heads up, make sure before even beginning to troubleshoot an issue like this you do a factory reset to the unit and get the latest available firmware on it. Usually that fixes annoying issues like this.Thanks, Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 30, 2006, at 10:36 PM, Jason Kim wrote:It's noisy while talking.Any idea?Thanks in advance.JasonCheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com)___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Audiocodes MP-20x

2006-10-23 Thread Jessee J Holmes

The reason not many people have this product, is because this product is not going to be available to the public at this time.Audiocodes will only provide this product (currently due to ship in December) as 1,000-piece minimum orders for the MP-202. The MP-201 will be available sometime quarter 1 2007 and then the mixed FXS/FXO 202 will follow.  MSRP is currently estimated at $99/unit.This unit is only to be sold to service providers and large installs per Audiocodes current VoIP direction they are moving.If you'd like more information on obtaining / testing this unit, you can contact me off list. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 22, 2006, at 5:47 PM, Andrew Joakimsen wrote:Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdfSeems like a good device, but I can't seem to find anyone actually using them... ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom boot error

2006-10-20 Thread Jessee J Holmes

The correct way to perform a factory reset on the Polycom phone is documented in one of our knowledge-base Articles, number: KB-001032http://voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-13.htmYou want to do a "factory format" to COMPLETELY erase everything on the phone.After this is done you can have the Polycom phone get the latest bootrom (important for this type of error) and firmware from your FTP or TFTP server set up at your office. If you don't have one, nows the time to learn (admin guide and normal IT knowledge), get one and use that to get the latest bootrom (3.2.2) and firmware (2.0.1 or 1.6.7 if you don't want to attempt version 2 yet) on your phones.That should resolve your problem.You can get the bootrom and firmware from your service provider, distributor, or reseller as long as they are Polycom certified.Hope that helps. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 19, 2006, at 10:26 PM, Neider, Clint wrote:  I am having the same issue as below.  Has this issue been  solved or does anyone know an answer?  This error recently began and we have multiple phones out of commission.  PLEASE HELP!! http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html  How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -Original Message-From: Dovid Bender [mailto:asteriskusers at dovid.net]Sent: Tuesday, August 15, 2006 11:16 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom upgrade issue  I believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine. - Original Message - From: Curt   Shaffer To: 'Asterisk Users Mailing List -   Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PMSubject: [asterisk-users] Polycom upgrade issue  OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. >From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log   0527180621|cfg  |4|00|Could not get all 512 bytes of the header. 0527181013|cfg  |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006   I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there?   Thanks!   Curt   Clint NeiderEmail Administrator[EMAIL PROTECTED]Alta Resources | IT Application Services | 120 N Commercial St | Neenah, WI 54956 | Office (920) 751-5800 x 7472 | This email message is intended only for the addressee(s) and contains information that may be confidential and/or copyright.  If you are not the intended recipient please notify the sender by reply email and immediately delete this email. Use, disclosure or reproduction of this email by anyone other than the intended recipient(s) is strictly prohibited. No representation is made that this email or any attachments are free of viruses. Virus scanning is recommended and is the responsibility of the recipient.   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom IP650

2006-10-18 Thread Jessee J Holmes

Dean,I don't think anyone does yet, Polycom is telling us December.In the past, they've been pretty good at keeping their word. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 18, 2006, at 8:43 AM, Dean Collins wrote: Does anyone have an actual delivery date on the new Polycom HD IP650’s? I’m getting sick of not having a backlit screen and thinking of upgrading.Cheers, Dean   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VoipSupply? [Semi-Urgent]

2006-10-16 Thread Jessee J Holmes

Dear VaibhaV,You can purchase this part from pretty much any certified Polycom reseller.For the IP 30x/50x you would want the Mfg Part Number 2200-07496-001For the IP 430/60x you would want the Mfg Part Number 2200-17492-101We among many other certified resellers sell this part.Being a reseller ourselves I can understand why VoIPSupply does this (as far as wanting the phone and the power supply shipped back whole), but I also understand your frustrations with this kind of setup.Additionally, being a Minnesota based company, we can understand how these kind of weather related conditions can affect quality of service and with such we offer our sincerest wishes that everyone at VoIPSupply stays warm and safe. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 16, 2006, at 3:25 PM, VaibhaV Sharma wrote:I don't think this is a problem because of the snow storm.I just got off the phone with them. The sales guy I used to deal with left afew months back and since then, its been a pain to get anything done withthem. People I have dealt with had no clue.I called them this morning for a problem to be told that a technical supportperson will call me back "within an hour". Then no one calls back for 5hours. So when I call them back, I am told "We don't do technical support onthe phone. I don't know who told you that".The lady who I was speaking with had no clue of what I was asking for. Shekept putting me on hold to ask someone for an answer.What was my question?Q. We purchased 25 polycom IP 601/501 from you a while back and one of them has a faulty power supply. How do I get a new one?A. Hold on Oh! You have to speak with RMA and not technical support. Go to our website / rma and submit an RMA.Q. Well, power supplies don't have serial numbers!A. Hold on. .. No you will have to obtain an RMA!Q. Well, what do I send to you? Can I speak with a technical support person?A. Hold on. .. Send us the power supply *and* the phone.Q. It will cost me the money for a power supply to ship the phone to you. Can you tell me somewhere else I can get just the power supply?A. If I had the answer I would have told you, sir.Gah!This is just one case. I am really disappointed with their service. I amworried about our technical support options for the polycom phones after thelast few expereinces with Voipsupply.--VaibhaVOn 10/14/06 10:36 AM, "Matt" <[EMAIL PROTECTED]> wrote: Contact them again... they have always been very good... I'm chockingthis up to the snow storm.On 10/13/06, Shaw Terwilliger <[EMAIL PROTECTED]> wrote: Matt wrote: Hi,Does anyone know what is going on with voipsupply? My sales guyhasn't been online in several days, their 800 number is fasy busy, asare their direct lines. And the canadian store website is down. Whatthe heck is going on? If you search the archives from a few months ago you'll find a fewunhappy voipsupply customers (including me). They never shipped what Iordered, didn't respond to any e-mail or calls. The president saw thelist traffic and sent me a long apology (stating his commitment toservice) and offered to send me an extra component that I had cancelledthe order for--free of charge--as a show of good will.It's been two or three months since that promise, and I never receivedthe part. He hasn't responded to my follow-up "did you really mean it?"e-mail either.--Shaw Terwilliger <[EMAIL PROTECTED]>SourceGear LLC ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom HDVoice

2006-10-13 Thread Jessee J Holmes

Actually, come to think of it, I don't know who will support it. Does Asterisk support G.722? From what I know it doesn't, is it included in the 1.4 beta? Will they support it? If Asterisk doesn't support it, then the phone won't do "HD" anyways. So then the questions comes to, what other PBX system or service provider will support this new "HD" standard?Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 13, 2006, at 11:14 AM, Forrest Beck wrote:Has anyone used the Polycom HDvoice phone yet?  I am curious if ituses a different codec.  Does it actually sound any better?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom HDVoice

2006-10-13 Thread Jessee J Holmes

I've played with one, but they are not available yet from Polycom (as in shipping to distributors). They are only demoing the phones at the moment. I'm sure if anyone has gone to any VoIP trade-show (like VON) they would have seen and used one of these, but under Polycom environments.It uses the G.722 codec, this is HDVoice according to Polycom. It's "supposed" to sound better. Under Polycom demo conditions, it does sound better. Polycom has samples of the sound of the phones on their website I believe so you can hear for yourself, but I'm still skeptical of what it really will sound like until we all get the phones in our hands and installed on our "messed up Asterisk environments" (joke). :)Hope that helps some. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 13, 2006, at 11:14 AM, Forrest Beck wrote:Has anyone used the Polycom HDvoice phone yet?  I am curious if ituses a different codec.  Does it actually sound any better?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list

2006-10-11 Thread Jessee J Holmes

Dean,Tough call ... I haven't played with an IP 500 in a long time now and all that I know is Polycom officially doesn't support them.I'm sure the 2.0.1 firmware wasn't designed to ever work with bootroms 2.xx. I'm sure the problem lies with either the phone not supporting it or the bootrom not accepting the firmware, but I'd be very weary of upgrading an IP 500 to a 3.xx bootrom.Maybe someone else has better experience with this that has some of the older phones. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 11, 2006, at 10:29 AM, Dean Collins wrote: Hi Jesse,4 x ip500’s  I’ve held off upgrading the bootrom past 2.62 as I understand this is a one way trip to 3.01 and above. As I’m a second hand hardware user I don’t have access to Polycom’s direct firmware and have been upgrading from freedomphone.net    Cheers, Dean   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, 11 October 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list  Dean,    Try obtaining the latest bootrom again, should be 3.2.2, we've seen this happen before for various odd reasons and Polycom's recommended fix is get the "non-engineering version" of the bootrom (don't ask please, just do it).     So download the bootrom again and attempt it once more, while you're at it, be safe and get the 2.0.1 firmware again as well. Let me know if this fixes you're problem. By the way, are you using IP 500's? or 501's? 500's may not take, I think we had that discussion in this list before.          Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/      On Oct 10, 2006, at 7:10 PM, Dean Collins wrote:I've had problems loading 2.01 onto 2 of my 4 polycom 500's     2 work great no probs, 2 I cant get it to upload without failing.        Cheers,     Dean         -Original Message-  From: [EMAIL PROTECTED] [mailto:asterisk-users-  [EMAIL PROTECTED]] On Behalf Of Eric "ManxPower" Wieling  Sent: Tuesday, 10 October 2006 5:56 PM  To: Asterisk Users Mailing List - Non-Commercial Discussion  Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users"   mailing   list     What I don't understand is why people MUST use the 2.0.x firmware.      ___  --Bandwidth and Colocation provided by Easynews.com --     asterisk-users mailing list  To UNSUBSCRIBE or update options visit:     http://lists.digium.com/mailman/listinfo/asterisk-users   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 2.01 sip issues

2006-10-11 Thread Jessee J Holmes

Dear Issac,Makes sense.We got asked about moving back to firmware 1.6.7 as well and the official answer from Polycom is "not a problem"! Put the firmware on your server and remove the 2.0 firmware from this server and when the phone reboots it will grab the 1.6.7 firmware and load it on the phone. Polycom says its as easy as that.We can surely get 1.6.7 on our ftp site. We haven't done this due to a major server system upgrade we've been working on for our website. The current server we have is a temporary location for this firmware, we have just the necessities on this server as this particular server isn't within our main cluster of servers (there isn't much bandwidth or power here until we move to the new server farm). I'll have one of our techs post this firmware in a little bit here on the temp ftp server.Good question on the .cfg files. I don't know ... 1.6.7 .cfg files won't work "correctly" with 2.0.1 firmware since the files changed. Not sure about the other way around, I would assume they'd work, but wouldn't recommend it as you may experience stability issues or glitches from the phone not knowing what to do with some the parameters in these newer files. It's always best to use the .cfg files given with the firmware on your phones.Hope that helps.As far 1.6.7 firmware supporting multiple presences (48 i think), maybe I was wrong on that; however, I remember reading the 2.0.1 firmware release notes and they mentioned that feature was fixed within the 2.0 firmware. Maybe they fixed it before that and just never documented it or maybe I misread it. If it works through in 1.6.7, great! Thanks Douglas. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 11, 2006, at 10:36 AM, Issac Simchayof wrote: Jessee, The reason for me upgrading to 2.01 is we wanted to add some 430’s to our system which from what I understand have a problem with 1.67, at this point we will just go with more 501’s instead. What is the procedure to go back to 1.67? Will you be adding 1.67 to your FTP site? currently you only have 2.01.Will the sip.cfg and phone.cfg from 2.01 work on 1.67? Thanks, Issac From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, October 11, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list Most want the 2.0.1 firmware for a few reasons: A) They have the latest; although, this is a poor reason, it's still a reason people download and use the latest firmware - remember here always, "If it's not broken, DON'T fix it!" B) They are hoping to fix a previous problem they've had in the past (i.e. stability issues - usually caused by other factors besides just firmware) C) They are told to. IF you are needing to talk to a support professional, especially Polycom, you NEED to upgrade to the latest firmware or they simply will not help you (most of the times anyways) These are a few I can think of anyways; and unfortunately, it's going to be a problem sooner or later. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 4:56 PM, Eric ManxPower Wieling wrote:What I don't understand is why people MUST use the 2.0.x firmware. Jessee J Holmes wrote: A few of our technical support staff here at Atacomm are currently working on this issue with Polycom, Digium and one of our customer's (who posted in here earlier this week). There are some major differences from the 1.x.x firmware and the 2.01 firmware. Obviously, many on here, voip-info.org, and all over the Internet have reported troubles where their presence feature that stops working after they upgrade their Polycom phones to firmware revision 2.0.1 when the phones are configured with an Asterisk system. A couple new things have been added to the .cfg files that MUST now be set in order for presence to work again with an Asterisk system. In the sip.cfg file: feature feature.1.name="presence" feature.1.enabled="1" In the phone[mac].cfg file:However, there are still confirmed problems with this setup (i.e. LEDs not working), which Polycom and ourselves are currently testing in our labs trying to fix. The reasoning for this is Digium doesn't seem to follow the "standards" for presence support and are currently working to change this functionality within Asterisk. Polycom designed their phones, and specifically their firmware, to work with the "standards" (more specifically, the Microsoft LCS - Live Communications Serv

Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list

2006-10-11 Thread Jessee J Holmes

Limit was increased in firmware 2.0.1.NOTE: a new Polycom Administrator's guide is now also available covering the 2.0.1 features. Re-obtain this manual if you haven't from your reseller or from Polycom direct if you're certified. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 11, 2006, at 12:01 AM, Douglas Garstang wrote:I think that limit was increased in 1.6.6 or 1.6.7.	-Original Message- 	From: C F [mailto:[EMAIL PROTECTED]] 	Sent: Tue 10/10/2006 6:57 PM 	To: Asterisk Users Mailing List - Non-Commercial Discussion 	Cc: 	Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list			On 10/10/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:	> What I don't understand is why people MUST use the 2.0.x firmware.		I believe it's because the limit of how many can be monitored at once,	someone correct me if I'm wrong.		>	> Jessee J Holmes wrote:	> > A few of our technical support staff here at Atacomm are currently	> > working on this issue with Polycom, Digium and one of our customer's	> > (who posted in here earlier this week).	> >	> > There are some major differences from the 1.x.x firmware and the 2.01	> > firmware. Obviously, many on here, voip-info.org, and all over the	> > Internet have reported troubles where their presence feature that stops	> > working after they upgrade their Polycom phones to firmware revision	> > 2.0.1 when the phones are configured with an Asterisk system. A couple	> > new things have been added to the .cfg files that MUST now be set in	> > order for presence to work again with an Asterisk system.	> >	> > In the sip.cfg file:	> >	> > feature feature.1.name="presence" feature.1.enabled="1"	> >	> > In the phone[mac].cfg file:	> >	> > 	> >	> >	> > However, there are still confirmed problems with this setup (i.e. LEDs	> > not working), which Polycom and ourselves are currently testing in our	> > labs trying to fix. The reasoning for this is Digium doesn't seem to	> > follow the "standards" for presence support and are currently working to	> > change this functionality within Asterisk. Polycom designed their	> > phones, and specifically their firmware, to work with the "standards"	> > (more specifically, the Microsoft LCS - Live Communications Server).	> >	> > This issue has been reported on multiple instances to Polycom, Digium,	> > and ourselves; but, no real resolution is completed yet. We'll continue	> > working on this issue within our labs and post an official answer when	> > one is available.	> >	> > Sorry for the bit of bad news, if anyone is willing to contribute	> > working / half working code, we'd be more than happy to look at it and	> > work with Polycom and Digium on getting this fixed for everyone AS	> ___	> --Bandwidth and Colocation provided by Easynews.com --	>	> asterisk-users mailing list	> To UNSUBSCRIBE or update options visit:	>    http://lists.digium.com/mailman/listinfo/asterisk-users	>	___	--Bandwidth and Colocation provided by Easynews.com --		asterisk-users mailing list	To UNSUBSCRIBE or update options visit:	   http://lists.digium.com/mailman/listinfo/asterisk-users	___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list

2006-10-11 Thread Jessee J Holmes

Dean,Try obtaining the latest bootrom again, should be 3.2.2, we've seen this happen before for various odd reasons and Polycom's recommended fix is get the "non-engineering version" of the bootrom (don't ask please, just do it).So download the bootrom again and attempt it once more, while you're at it, be safe and get the 2.0.1 firmware again as well. Let me know if this fixes you're problem. By the way, are you using IP 500's? or 501's? 500's may not take, I think we had that discussion in this list before. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 7:10 PM, Dean Collins wrote:I've had problems loading 2.01 onto 2 of my 4 polycom 500's2 work great no probs, 2 I cant get it to upload without failing.Cheers,Dean -Original Message-From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED]] On Behalf Of Eric "ManxPower" WielingSent: Tuesday, 10 October 2006 5:56 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing listWhat I don't understand is why people MUST use the 2.0.x firmware. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list

2006-10-11 Thread Jessee J Holmes

I'm not saying Microsoft is the standard (they usually aren't by FAR), but how Microsoft handles presence and interoperates with presence on various IP phones is what Polycom calls a "standard" (guess I should have quoted that word originally).I believe there is some RFC for presence out there that some people consider the "standard"; although, I'm not sure what this is... Saying the word standard to me is like saying that someone is "normal" . there is no such thing. It's normally just something that "most" people agree on as a standard. Anyways, some of us here at Atacomm are currently arguing with Polycom why they can't make their phones support BOTH "methods" of handling presence, we think that would be the simplest solution instead of just shutting out some of the systems like Asterisk. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 6:05 PM, Mike Clark wrote:Jessee J Holmes wrote: However, there are still confirmed problems with this setup (i.e. LEDsnot working), which Polycom and ourselves are currently testing in ourlabs trying to fix. The reasoning for this is Digium doesn't seem tofollow the "standards" for presence support and are currently working tochange this functionality within Asterisk. Polycom designed theirphones, and specifically their firmware, to work with the "standards"(more specifically, the Microsoft LCS - Live Communications Server). What are the specific standards to which you refer? To my knowledge,Microsoft LCS is *not* an industry adopted standard.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list

2006-10-11 Thread Jessee J Holmes

Most want the 2.0.1 firmware for a few reasons:A) They have the latest; although, this is a poor reason, it's still a reason people download and use the latest firmware - remember here always, "If it's not broken, DON'T fix it!"B) They are hoping to fix a previous problem they've had in the past (i.e. stability issues - usually caused by other factors besides just firmware)C) They are told to. IF you are needing to talk to a support professional, especially Polycom, you NEED to upgrade to the latest firmware or they simply will not help you (most of the times anyways)These are a few I can think of anyways; and unfortunately, it's going to be a problem sooner or later. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 4:56 PM, Eric ManxPower Wieling wrote:What I don't understand is why people MUST use the 2.0.x firmware.Jessee J Holmes wrote: A few of our technical support staff here at Atacomm are currently working on this issue with Polycom, Digium and one of our customer's (who posted in here earlier this week).There are some major differences from the 1.x.x firmware and the 2.01 firmware. Obviously, many on here, voip-info.org, and all over the Internet have reported troubles where their presence feature that stops working after they upgrade their Polycom phones to firmware revision 2.0.1 when the phones are configured with an Asterisk system. A couple new things have been added to the .cfg files that MUST now be set in order for presence to work again with an Asterisk system.In the sip.cfg file:feature feature.1.name="presence" feature.1.enabled="1"In the phone[mac].cfg file:However, there are still confirmed problems with this setup (i.e. LEDs not working), which Polycom and ourselves are currently testing in our labs trying to fix. The reasoning for this is Digium doesn't seem to follow the "standards" for presence support and are currently working to change this functionality within Asterisk. Polycom designed their phones, and specifically their firmware, to work with the "standards" (more specifically, the Microsoft LCS - Live Communications Server).This issue has been reported on multiple instances to Polycom, Digium, and ourselves; but, no real resolution is completed yet. We'll continue working on this issue within our labs and post an official answer when one is available.Sorry for the bit of bad news, if anyone is willing to contribute working / half working code, we'd be more than happy to look at it and work with Polycom and Digium on getting this fixed for everyone AS ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list

2006-10-10 Thread Jessee J Holmes

A few of our technical support staff here at Atacomm are currently working on this issue with Polycom, Digium and one of our customer's (who posted in here earlier this week).There are some major differences from the 1.x.x firmware and the 2.01 firmware. Obviously, many on here, voip-info.org, and all over the Internet have reported troubles where their presence feature that stops working after they upgrade their Polycom phones to firmware revision 2.0.1 when the phones are configured with an Asterisk system. A couple new things have been added to the .cfg files that MUST now be set in order for presence to work again with an Asterisk system.In the sip.cfg file:feature feature.1.name="presence" feature.1.enabled="1"In the phone[mac].cfg file:However, there are still confirmed problems with this setup (i.e. LEDs not working), which Polycom and ourselves are currently testing in our labs trying to fix. The reasoning for this is Digium doesn't seem to follow the "standards" for presence support and are currently working to change this functionality within Asterisk. Polycom designed their phones, and specifically their firmware, to work with the "standards" (more specifically, the Microsoft LCS - Live Communications Server).This issue has been reported on multiple instances to Polycom, Digium, and ourselves; but, no real resolution is completed yet. We'll continue working on this issue within our labs and post an official answer when one is available.Sorry for the bit of bad news, if anyone is willing to contribute working / half working code, we'd be more than happy to look at it and work with Polycom and Digium on getting this fixed for everyone ASAP.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 3:07 PM, Issac Simchayof wrote:Polycom 601 with Sip 2.01Anyone using Sip 2.01? I have upgraded my phones and now presence no longerfunctions. Buddy list shows all phones online but status does not change when someoneis on a call. Also blf does not function.I am using trixbox, 1.67 was working fine on the same box.Any ideas?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] problems with Polycom 500 boot up

2006-09-21 Thread Jessee J Holmes

Dear Dean,The difference between these two phones is simply that the IP 501 phone has extra memory on it which was added to the phone ONLY to provide HTTPS provisioning options. (this is what Polycom told us anyways)Funny thing is, not many people know how to use the HTTPS options because it's not well documented by the manufacturer supposedly. *shakes head* I've been working pretty closely with Polycom right now to get this FINALLY documented, hopefully well. We'll post it to our knowledge-bases when I get something put together on this topic (no idea when this will happen yet since I'm waiting on Polycom engineers to do some more in-lab testing). Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 7:33 PM, Dean Collins wrote: Speaking of which, can anyone tell me the differences between the IP500 and the IP501?   Cheers,Dean  From: Dean Collins  Sent: Wednesday, 20 September 2006 8:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up  Lol, does it work thought?   Cheers,Dean  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Forum Expansive Sent: Wednesday, 20 September 2006 7:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up  I just updated my 500 to the latest - 3.2.2. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Azfhasterisk Sent: Wednesday, 20 September 2006 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up  I am not positive but I thought that the 2.6.2 bootrom was the highest you could put on the ip500. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, September 20, 2006 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up  Get 3.2.2 from your reseller or installer. Bootrom 3.2.2 and firmware 2.0.1 is the latest available from Polycom. Your phone installer, service provider, or reseller should be able to provide you with this firmware.      Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/     On Sep 20, 2006, at 2:53 PM, Forum wrote:   I think that's the problem the bootROM version on the phone is 2.02 Apr 02  16.33. Does anyone have this version and the corresponding Sip?     -Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED]] On Behalf Of Eric  "ManxPower" Wieling  Sent: Wednesday, September 20, 2006 11:48 AM  To: Asterisk Users Mailing List - Non-Commercial Discussion  Subject: Re: [asterisk-users] problems with Polycom 500 boot up     Forum wrote:   Thanks for your response.           Unfortunately I still receive the same error - 'Error updating bootrom' -   no   matter what version of sip and the bootROM I upload to the ftp site. I   have   even used the latest release of the fimware - could I have somehow broke   the   phone with a corrupted flash. How do I do a full format when it can not  update the bootROM?      Check the EXISTING BootROM on the phone. You can't usually downgrade   versions.     Also check the password configured for the FTP user on the phone.  ___  --Bandwidth and Colocation provided by Easynews.com --     asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users     ___  --Bandwidth and Colocation provided by Easynews.com --     asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users    ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] problems with Polycom 500 boot up

2006-09-21 Thread Jessee J Holmes

Interesting, it shouldn't work according to Polycom, but I guess go with it :)I really apologize, I haven't worked with a IP 500 in such a long time and most people who mention the 500 really end up meaning the 501, I guess I assumed wrong. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 7:14 PM, Forum wrote: Like a charm From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dean Collins Sent: Wednesday, September 20, 2006 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] problems with Polycom 500 boot up  Lol, does it work thought?   Cheers,Dean  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Forum Expansive Sent: Wednesday, 20 September 2006 7:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up  I just updated my 500 to the latest - 3.2.2. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Azfhasterisk Sent: Wednesday, 20 September 2006 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] problems with Polycom 500 boot up  I am not positive but I thought that the 2.6.2 bootrom was the highest you could put on the ip500. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, September 20, 2006 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up  Get 3.2.2 from your reseller or installer. Bootrom 3.2.2 and firmware 2.0.1 is the latest available from Polycom. Your phone installer, service provider, or reseller should be able to provide you with this firmware.      Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/     On Sep 20, 2006, at 2:53 PM, Forum wrote:   I think that's the problem the bootROM version on the phone is 2.02 Apr 02  16.33. Does anyone have this version and the corresponding Sip?     -Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED]] On Behalf Of Eric  "ManxPower" Wieling  Sent: Wednesday, September 20, 2006 11:48 AM  To: Asterisk Users Mailing List - Non-Commercial Discussion  Subject: Re: [asterisk-users] problems with Polycom 500 boot up     Forum wrote:   Thanks for your response.           Unfortunately I still receive the same error - 'Error updating bootrom' -   no   matter what version of sip and the bootROM I upload to the ftp site. I   have   even used the latest release of the fimware - could I have somehow broke   the   phone with a corrupted flash. How do I do a full format when it can not  update the bootROM?      Check the EXISTING BootROM on the phone. You can't usually downgrade   versions.     Also check the password configured for the FTP user on the phone.  ___  --Bandwidth and Colocation provided by Easynews.com --     asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users     ___  --Bandwidth and Colocation provided by Easynews.com --     asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] problems with Polycom 500 boot up

2006-09-20 Thread Jessee J Holmes

Thank you pointing that out, you're right. Thought they were working with the IP 501 phone not the 500. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 5:46 PM, Azfhasterisk wrote: I am not positive but I thought that the 2.6.2 bootrom was the highest you could put on the ip500. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, September 20, 2006 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Get 3.2.2 from your reseller or installer. Bootrom 3.2.2 and firmware 2.0.1 is the latest available from Polycom. Your phone installer, service provider, or reseller should be able to provide you with this firmware. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 2:53 PM, Forum wrote:I think that's the problem the bootROM version on the phone is 2.02 Apr 02 16.33. Does anyone have this version and the corresponding Sip? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Eric "ManxPower" Wieling Sent: Wednesday, September 20, 2006 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Forum wrote: Thanks for your response. Unfortunately I still receive the same error - 'Error updating bootrom' - no matter what version of sip and the bootROM I upload to the ftp site. I have even used the latest release of the fimware - could I have somehow broke the phone with a corrupted flash. How do I do a full format when it can not update the bootROM? Check the EXISTING BootROM on the phone. You can't usually downgrade versions. Also check the password configured for the FTP user on the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] problems with Polycom 500 boot up

2006-09-20 Thread Jessee J Holmes

Get 3.2.2 from your reseller or installer. Bootrom 3.2.2 and firmware 2.0.1 is the latest available from Polycom. Your phone installer, service provider, or reseller should be able to provide you with this firmware. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 2:53 PM, Forum wrote:I think that's the problem the bootROM version on the phone is 2.02 Apr 0216.33. Does anyone have this version and the corresponding Sip?-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Eric"ManxPower" WielingSent: Wednesday, September 20, 2006 11:48 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] problems with Polycom 500 boot upForum wrote: Thanks for your response.Unfortunately I still receive the same error - 'Error updating bootrom' - no matter what version of sip and the bootROM I upload to the ftp site. I have even used the latest release of the fimware - could I have somehow broke the phone with a corrupted flash. How do I do a full format when it can notupdate the bootROM? Check the EXISTING BootROM on the phone.  You can't usually downgrade versions.Also check the password configured for the FTP user on the phone.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] problems with Polycom 500 boot up

2006-09-20 Thread Jessee J Holmes

Dear Steve,This is very puzzling, it should take.bootrom.ld is the file on your FTP server right? Does your FTP server show the phone pinging it in the FTP application logs?Which bootrom are you trying to load. The latest should be 3.2.2 I believe.Let me know and I'll see what I can find you. You shouldn't have a corrupted bootrom chip, but I won't rule it out just yet. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 20, 2006, at 12:18 PM, Forum wrote: Thanks for your response. Unfortunately I still receive the same error – ‘Error updating bootrom’ – no matter what version of sip and the bootROM I upload to the ftp site. I have even used the latest release of the fimware – could I have somehow broke the phone with a corrupted flash. How do I do a full format when it can not update the bootROM? Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Friday, September 15, 2006 7:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with Polycom 500 boot up Dear Steve, The phone may be looking for it's specific configuration files (not phone1.cfg, but instead 0004Fcfg {or [mac].cfg}). In our past experience, if the phone was ever formatted (fully formatted), the phone will request this from the FTP server specified. Of course confirm your phone's login to your FTP server is correct, confirm the phone is logging in and grabbing the files (should be able to be done through your FTP program's interface). Also, as odd as this sounds, check your firewall on your network. In the past, we've ran into some weird things happening where the firewall will let some Polycom phones through, but not all. So confirm your Polycom phone is talking to your FTP client (again your log files can tell you this). For further information, I suggest looking at one of our knowledgeable articles on this topic: http://voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-24.htm Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 14, 2006, at 4:03 PM, Forum wrote:I have a Polycom 500 that I am having issues with provisioning via an ftp server. I have a bunch of 301’s that find the server and configure without an issue. For some reason the 500 gives me an error that it ‘could not contact boot server’ and will reboot continuously. I also get the error ‘Error updating Bootrom’. I am using Bootrom 3.2.1. What files do I need on the ftp server ? – I have sip.Id, bootrom.Id, sip.ver, phone1.cfg and sip.cfg. Any help would be appreciated! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom default handset volume

2006-09-19 Thread Jessee J Holmes

I think the following should resolve your problem, if I'm understanding it correctly:In the sip.cfg file, look at setting the following:1= remember last setting, 0=return to defaultThis will make the phone remember your setting and will not reset the setting every time you need to make a new call. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 19, 2006, at 10:23 AM, Damon Estep wrote: I had read a post somewhere that there is an XML parameter for the Polycom config files for default handset volume, but I can not locate it again. Anyone know what it is? I want to set the default handset volume higher on some phones, despite the ADA hearing aid warning in the admin manual J ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] problems with Polycom 500 boot up

2006-09-15 Thread Jessee J Holmes

Dear Steve,The phone may be looking for it's specific configuration files (not phone1.cfg, but instead 0004Fcfg {or [mac].cfg}). In our past experience, if the phone was ever formatted (fully formatted), the phone will request this from the FTP server specified. Of course confirm your phone's login to your FTP server is correct, confirm the phone is logging in and grabbing the files (should be able to be done through your FTP program's interface).Also, as odd as this sounds, check your firewall on your network. In the past, we've ran into some weird things happening where the firewall will let some Polycom phones through, but not all. So confirm your Polycom phone is talking to your FTP client (again your log files can tell you this).For further information, I suggest looking at one of our knowledgeable articles on this topic: http://voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-24.htm Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 14, 2006, at 4:03 PM, Forum wrote: I have a Polycom 500 that I am having issues with provisioning via an ftp server. I have a bunch of 301’s that find the server and configure without an issue. For some reason the 500 gives me an error that it ‘could not contact boot server’ and will reboot continuously.  I also get the error ‘Error updating Bootrom’. I am using Bootrom 3.2.1. What files do I need on the ftp server ? – I have sip.Id, bootrom.Id, sip.ver, phone1.cfg and sip.cfg.  Any help would be appreciated! Steve   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Grandstream Budgetone phones don't show

2006-09-11 Thread Jessee J Holmes

Ricardo,From what I know its a physical limitation of the display Grandstream chose on that phone, Grandstream recommends purchasing the GXP-2000 phone instead if you're looking for this feature.Grandstream has no plans from what I am aware of of making this change to the BudgetTone series phones.You are more than welcome to inquire directly from Grandstream though, this is just from what I know from dealing with them in the past. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 11, 2006, at 11:47 AM, Ricardo Carvalho wrote:I guess this functionality will be in the future added to new firmware releases don't you people think so?Ricardo.Doug Lytle wrote: These phones aren't capable of alphanumeric entries, only numeric.Doug Tom Vile wrote: They only do numeric callerid.  ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Jessee J Holmes

Also keep in mind that as of right now, the latest bootrom and firmware available from Polycom (and thus your reseller) are Bootrom 3.2.2 and Firmware 2.0.1The 2.0.1 firmware is new as of a day or two and include some enhancements for buddy lists and shared presence as well as newly added secured TLS support (according to Polycom). Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 7, 2006, at 1:19 PM, Douglas Garstang wrote: That process is worse than pulling teeth!   -Original Message-From: Jessee J Holmes   [mailto:[EMAIL PROTECTED]]Sent: Thursday, September 07, 2006   11:25 AMTo: Asterisk Users Mailing List - Non-Commercial   DiscussionSubject: Re: [asterisk-users] Polycom new firmware and   bootromAll authorized Polycom resellers will have access   to this firmware and are required to provide this firmware to you. Contact the   reseller you purchased the Polycom phone from.  Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:  Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom.On 9/7/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:Polycom   are analy retentive about giving out software updates.>   -Original Message-> From: Nathan Alberti [mailto:[EMAIL PROTECTED]]> Sent:   Thursday, September 07, 2006 10:25 AM > To: Asterisk Users Mailing   List - Non-Commercial Discussion> Subject: Re: [asterisk-users]   Polycom new firmware and bootrom>>>>>   Stupid question where did you find it ? >>>   Looked at their site downloads and under the extranet site but   could> only see old versions.>>   Nathan.>> On 07/09/2006, at 10:21 AM, Chris Dos   wrote:>> > Well, it seems that Polycom has release new   firmware 2.0.1 and> > bootrom 3.2.2.> > I've proceded   to upgrade all my ip430 phones because they were> >   essentially> > broken with the original firmware. >   >> > All the phones boot up fine now, grab their   files.  They> just won't> > talk to the>   > asterisk server any more.   I just figured out that I need   to hard> > code the sip > > server and tell it to talk   udp only.  After this, the> phones worked> >   again.> >> > Any idea on what I need to configure to   fix the phones so> they will> > know which > >   server to talk to and only talk to it via udp?> >>   > Chris> >> >   ___> > --Bandwidth   and Colocation provided by Easynews.com   --> >> > asterisk-users mailing list> > To   UNSUBSCRIBE or update options visit:>   >   http://lists.digium.com/mailman/listinfo/asterisk-users>>   ___> --Bandwidth and   Colocation provided by Easynews.com   -->> asterisk-users mailing list> To UNSUBSCRIBE or   update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users   >___--Bandwidth   and Colocation provided by Easynews.com   --asterisk-users mailing listTo UNSUBSCRIBE or update options   visit:    http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Jessee J Holmes

I agree, Polycom should make this publicly available; but unfortunately, I've seen worse policies out there *cough* Cisco *cough*.The reseller shouldn't give you any hassle about it  and if they do, or if you can't reach them for whatever reason (A.K.A. no email replies or phones being answered), that's violation of the contract they had to sign to be Polycom Authorized in the first place, and Polycom will take immediate action to rectify the situation if that's the case. I'd suggest finding another place to purchase from if your current reseller is giving you troubles with getting the firmware for you.The firmware is probably about a 50 MB download (i think) and can be downloaded via HTTP or FTP. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 7, 2006, at 1:19 PM, Douglas Garstang wrote: That process is worse than pulling teeth!   -----Original Message-From: Jessee J Holmes   [mailto:[EMAIL PROTECTED]]Sent: Thursday, September 07, 2006   11:25 AMTo: Asterisk Users Mailing List - Non-Commercial   DiscussionSubject: Re: [asterisk-users] Polycom new firmware and   bootromAll authorized Polycom resellers will have access   to this firmware and are required to provide this firmware to you. Contact the   reseller you purchased the Polycom phone from.  Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:  Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom.On 9/7/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:Polycom   are analy retentive about giving out software updates.>   -Original Message-> From: Nathan Alberti [mailto:[EMAIL PROTECTED]]> Sent:   Thursday, September 07, 2006 10:25 AM > To: Asterisk Users Mailing   List - Non-Commercial Discussion> Subject: Re: [asterisk-users]   Polycom new firmware and bootrom>>>>>   Stupid question where did you find it ? >>>   Looked at their site downloads and under the extranet site but   could> only see old versions.>>   Nathan.>> On 07/09/2006, at 10:21 AM, Chris Dos   wrote:>> > Well, it seems that Polycom has release new   firmware 2.0.1 and> > bootrom 3.2.2.> > I've proceded   to upgrade all my ip430 phones because they were> >   essentially> > broken with the original firmware. >   >> > All the phones boot up fine now, grab their   files.  They> just won't> > talk to the>   > asterisk server any more.   I just figured out that I need   to hard> > code the sip > > server and tell it to talk   udp only.  After this, the> phones worked> >   again.> >> > Any idea on what I need to configure to   fix the phones so> they will> > know which > >   server to talk to and only talk to it via udp?> >>   > Chris> >> >   ___> > --Bandwidth   and Colocation provided by Easynews.com   --> >> > asterisk-users mailing list> > To   UNSUBSCRIBE or update options visit:>   >   http://lists.digium.com/mailman/listinfo/asterisk-users>>   ___> --Bandwidth and   Colocation provided by Easynews.com   -->> asterisk-users mailing list> To UNSUBSCRIBE or   update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users   >___--Bandwidth   and Colocation provided by Easynews.com   --asterisk-users mailing listTo UNSUBSCRIBE or update options   visit:    http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom new firmware and bootrom

2006-09-07 Thread Jessee J Holmes

All authorized Polycom resellers will have access to this firmware and are required to provide this firmware to you. Contact the reseller you purchased the Polycom phone from. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote:Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom. On 9/7/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: Polycom are analy retentive about giving out software updates.> -Original Message-> From: Nathan Alberti [mailto:[EMAIL PROTECTED]]> Sent: Thursday, September 07, 2006 10:25 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: Re: [asterisk-users] Polycom new firmware and bootrom> Stupid question where did you find it ? >>> Looked at their site downloads and under the extranet site but could> only see old versions.>> Nathan.>> On 07/09/2006, at 10:21 AM, Chris Dos wrote:> > > Well, it seems that Polycom has release new firmware 2.0.1 and> > bootrom 3.2.2.> > I've proceded to upgrade all my ip430 phones because they were> > essentially> > broken with the original firmware. > >> > All the phones boot up fine now, grab their files.  They> just won't> > talk to the> > asterisk server any more.   I just figured out that I need to hard> > code the sip > > server and tell it to talk udp only.  After this, the> phones worked> > again.> >> > Any idea on what I need to configure to fix the phones so> they will> > know which > > server to talk to and only talk to it via udp?> >> > Chris> >> > ___> > --Bandwidth and Colocation provided by Easynews.com --> >> > asterisk-users mailing list> > To UNSUBSCRIBE or update options visit:> > http://lists.digium.com/mailman/listinfo/asterisk-users>> ___> --Bandwidth and Colocation provided by Easynews.com -- >> asterisk-users mailing list> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users >___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Budgetones - multiple phones losing IP addressduring day

2006-09-06 Thread Jessee J Holmes

So the phones aren't loosing the IP address any longer?Just confirming what you meant by "It's running now". Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 6, 2006, at 11:02 AM, Harden, Bob wrote: Its running now tethereal -f "host 12.20.121.2 and icmp"Capturing on eth0  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, September 06, 2006 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Budgetones - multiple phones losing IP addressduring day  Garth,    This may be a silly question, but are you running the latest firmware on the phones from Grandstream? If not, try upgrading one phone to see if it helps solve the problem.      Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/   On Sep 6, 2006, at 3:21 AM, Garth van Sittert wrote:Hi All     I have a site with 50 Budgetone 102's and about 5 snom phones.     At random intervals during the day about 20 or 30 of the Budgetones lose their connection to the network all at the same time.  It happens about once a day.  The Snom phones are fine and never get disconnected.  I can't ping the Budgetones IP's and the way to fix them is to simply unplug and reconnect.  Nothing interesting shows up in the logs even in debug mode except for the 'Peer XXX is now unreachable' repeated for each extension.     I haven't had much experience with the Budgetones.  Does anyone have any idea what could be causing this?     Thanks  Garth     ___  --Bandwidth and Colocation provided by Easynews.com --     asterisk-users mailing list  To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-users     ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Budgetones - multiple phones losing IP address during day

2006-09-06 Thread Jessee J Holmes

Garth,This may be a silly question, but are you running the latest firmware on the phones from Grandstream? If not, try upgrading one phone to see if it helps solve the problem.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 6, 2006, at 3:21 AM, Garth van Sittert wrote:Hi AllI have a site with 50 Budgetone 102's and about 5 snom phones.At random intervals during the day about 20 or 30 of the Budgetones lose their connection to the network all at the same time.  It happens about once a day.  The Snom phones are fine and never get disconnected.  I can't ping the Budgetones IP's and the way to fix them is to simply unplug and reconnect.  Nothing interesting shows up in the logs even in debug mode except for the 'Peer XXX is now unreachable' repeated for each extension.I haven't had much experience with the Budgetones.  Does anyone have any idea what could be causing this?ThanksGarth___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Can budgetone 101 display name part of cid?

2006-08-15 Thread Jessee J Holmes

Doug,That is correct you can only display the number on the BudgetTone 101, 102, and 200.If you wish to display the name as well, you will need to upgrade to the GXP-2000 phone. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 15, 2006, at 11:19 AM, Doug Lytle wrote:Guus Houtzager wrote: Hi,I've been trying to get this to work, but I'm not havinf much luck. So is it even possible to get a budgetone 101 to show the text bit of the    I'm not sure about the 101, but the Budgetone 100 is only capable of numeric data.Doug-- Ben Franklin quote:"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 1.6.7 firmware?

2006-08-08 Thread Jessee J Holmes

Dean,You need to put these files on a FTP server which the Polycom phone can access and log into.  Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 8, 2006, at 4:51 PM, Dean Collins wrote:Now does anyone have some firmware update instructions, I've forgottonhow to do this.Cheers,Dean -Original Message-From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED]] On Behalf Of Matt FlorellSent: Tuesday, 8 August 2006 5:14 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom 1.6.7 firmware?http://www.freedomphones.net/polycom/files/Firmware has just been updated on this site to include 1.6.7MATT---On 8/8/06, Dean Collins <[EMAIL PROTECTED]> wrote: http://www.freedomphones.net/polycom/files/I haven't updated firmware in ages but does this help?Cheers,Dean -Original Message-From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Louis-David MitterrandSent: Tuesday, 8 August 2006 12:37 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Polycom 1.6.7 firmware?Hello,I am looking for the latest 1.6.7 Polycom firmware?Is it available somewhere?Thanks,___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] polycom headset question

2006-08-08 Thread Jessee J Holmes

Dean,Which headset are you using? It may be an amplification issue from the headset side. Does this happen with only one phone with this headset or all phones?I know Polycom normally works pretty well with Plantronics headsets, Polaris to be specific since the amplification is built in on these headsets. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 8, 2006, at 12:10 PM, Dean Collins wrote: Does anyone know if there is a way of making the headset louder on the polycom 500’s? The handset volume works fine but I just find the headset a little low even on the highest setting.  Cheers,Dean   ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom 1.6.7 firmware?

2006-08-08 Thread Jessee J Holmes

You need to obtain this firmware from your certified installer or reseller. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 8, 2006, at 11:37 AM, Louis-David Mitterrand wrote:Hello,I am looking for the latest 1.6.7 Polycom firmware? Is it available somewhere?Thanks,___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] polycom soundstation 501 crash

2006-08-02 Thread Jessee J Holmes

At first, you would have to get these from your service provider or your reseller. They should have them available. I wish I could find a sample of one of the .cfg files, but I can't seem to locate it at this moment; however, here is a starting sample from Polycom.File Name: .cfgObviously, rename the file to the MAC address of the phone and change the text within the file to match up with your phone and preferred settings.If I can find a full working sample, I'll send it. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 2, 2006, at 9:28 AM, Stas Khromoy wrote:not to sound like an idiotbut where do i get the files ?these guys ?http://www.polycom.com/resource_center/0,1454,pw-6812-12612,FF.htmlSoundPoint IP/SoundStation IP SIP Software 1.6.7SoundPoint IP/SoundStation IP BootROM 3.2.1 Original Message  Subject: Re:[asterisk-users] polycom soundstation 501 crashFrom: Jessee J Holmes <[EMAIL PROTECTED]>To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>Date: 8/2/2006 10:20 AM This means the phone is attempting to load this configuration file and cannot find it from your boot server. The phone at this point must have a boot server with these files. Put this file on a FTP server and point the phone to that server to pickup and download this file.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Aug 2, 2006, at 8:50 AM, Stas Khromoy wrote: hey folkshope some one came across this problemone of our polycom's just crashedafter reboot it comes up with this errorerror loading 0004f204fcc.cfg___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users   ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] polycom soundstation 501 crash

2006-08-02 Thread Jessee J Holmes

This means the phone is attempting to load this configuration file and cannot find it from your boot server. The phone at this point must have a boot server with these files. Put this file on a FTP server and point the phone to that server to pickup and download this file. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 2, 2006, at 8:50 AM, Stas Khromoy wrote:hey folkshope some one came across this problemone of our polycom's just crashedafter reboot it comes up with this errorerror loading 0004f204fcc.cfg___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

50 matches

Mail list logo