.php /var/log/asterisk/cdr-csv/Master.csv
path is the location where your cdrmysql.php is located
second is the database already truncated for you
hope this helps
Regards.
Kyeyune Bob
VOIP Engineer
+256 774 702 258
On Wed, Sep 25, 2013 at 5:02 PM, Doug Lytle wrote:
> >> but
hello;
hopefully u can help me
i have asterisk vanilla installation 11 and i have also managed to install
webrtc2sip
how do i make asterisk to communicate with webrtc2sip c'se right now both
run independently
Regards.
Kyeyune Bob
Network & IT Engineer
+256 774 702 258
bob.kyey...@onesol
am also stuck with Alcatel lucent IP Touch 4018
any one connected them to Asterisk
thanks
Regards.
Kyeyune Bob
Network & IT Engineer
+256 774 702 258
bob.kyey...@onesolutions.ug
Integrated IT services from
Plot 57B Luthuli Avenue Bugolobi, Kampala
On Sun, Apr 28, 2013 at 11:56 PM, Ca
t timing source. I suspec that if I restart
the Asterisk service everything will come back up the way that it did last
time. However, I'm wondering if there would be a way to switch the Music On
Hold module back to using pthread timing without restarting the Asteris
Hello;
how do i embed and send the recorded file to email automagically
exten => _1XXX,3,MixMonitor(${CALLFILENAME}|b|/usr/sbin/wav2mp3
${CALLFILENAME} ${peeremail} ${EXTEN} ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} )
Regards.
Kyeyune Bob
Network & IT Engineer
+256 774 702 258
b
Did anything else change on her home network that could correlate to the time
this started flaking on you? (eg: a new router/gateway)
BB
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500,n,Hangup
This is working fine in an environment with many 330s, some 450s, some
335s and a 550 all running 3.3.1
Hope this helps you out.
Bob
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On Tue, Aug 2, 2011 at 4:39 PM, Lefteris Zafiris wrote:
> You can write a short makefile for just codec_ilbc module, build it and
> install it on your running asterisk system. You will have to install the
> asterisk18-devel package and get the asterisk source code either from
> a tar or from the s
Asterisk
from source?
Thanks,
Bob
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asterisk
I'm still using macro with Asterisk 1.8.5.0
On Fri, Jul 15, 2011 at 3:17 PM, Paul Belanger wrote:
> On 11-07-15 02:18 PM, Doug Lytle wrote:
>>
>> --[ UxBoD ]-- wrote:
>>>
>>> I back leveled to 1.8.3 and that works fine. What am I missing as
>>> app_macro has been installed okay?
>>
>> Macro was d
ads and DAHDI timing
modules and force MOH to use pthreads and force Page() to use DAHDI?
Bob
On Tue, Jun 28, 2011 at 12:03 AM, Faisal Hanif wrote:
> Call file are not suitable for you as asterisk process these files in serial
> mode (single threaded) and in case of large number of files p
nyone have insight into how we could accomplish this paging
feature or of anything that we may have missed?
I suspect we could get this all to work with the original Page()
application if there was a way to force MusicOnHold to use the pthread
timing module instead of the Dahdi timing module. I
On Fri, May 6, 2011 at 2:12 PM, Bob Beers wrote:
> On Fri, May 6, 2011 at 1:27 PM, Bob Beers wrote:
>> Hi Steven,
>>
>> Can you put the .spec file from dahdi-linux-kmod package up?
>
> Nevermind, I got it. :-)
>
> Looking at it now.
Not sure if this will work
On Fri, May 6, 2011 at 1:27 PM, Bob Beers wrote:
> Hi Steven,
>
> Can you put the .spec file from dahdi-linux-kmod package up?
Nevermind, I got it. :-)
Looking at it now.
Did you CC [Packager: Jason Parker
package to recognise the kmod PAE package as
> the right one for this kernel?
Hi Steven,
Can you put the .spec file from dahdi-linux-kmod package up?
I can get the srpm, but I'm stuck on a weak machine at the moment.
Maybe I can help you to mo
els concise"
"core show channels verbose"
>From my experience, they all "work" in 1.8, but do give different output.
--
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- Bob Beers
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ooked at SIP_HEADER() dialplan function?
<https://wiki.asterisk.org/wiki/display/AST/Function_SIP_HEADER>
Maybe you can parse Reason header in 4xx or 5xx response?
HTH,
-Bob
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iki/display/AST/Application_DISA>
cheers,
--
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uld also work.
Can you show your sip.conf and a trace of the call with 'sip set debug on'
that shows the a=crypto: line in the INVITE SDP? It doesn't happen for me.
I am able to send INVITEs with or without the a=crypto: line by setting
e
On Thu, Jan 20, 2011 at 12:03 PM, Danny Nicholas wrote:
> Putting the "--" in front of it might make it go away.
If I am not mistaken it should be exactly
two dashes followed by a space on a line alone
to indicate the end of the mail content.
But not all mail readers will honor
er, (hyphen,
pipe, comma, etc.)
I can't really tell what you are trying to achieve, but if CALLINGout
is the value
of a variable, say X, and you want just the first 6 characters, you
could use (maybe):
exten => s,n,Set(newVAR=${X:0:6})
HTH,
-Bob
--
The linewrapping by gmail of the patch file makes it difficult to read.
So, I added it as an attachment for any interested readers.
--
-Bob
--- asterisk-1.8.0-beta2.orig/channels/chan_sip.c 2010-07-26 15:59:03.0 -0400
+++ asterisk-1.8.0-beta2/channels/chan_sip.c 2010-11-05 12:18
On Fri, Nov 5, 2010 at 10:58 AM, Bob Beers wrote:
> Hi list,
>
> My need is to append a site specific parameter to the
> Contact: header on all INVITEs exiting * via a SIP trunk.
> I'd like it to look something like this:
>
> Contact:
>
> where SITE-ID=us.h
ing contact header
parameters from arriving requests/registrations, but nothing about
creating any such parameters.
Thanks for any help/hints,
--
-Bob Beers
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On Wed, Nov 3, 2010 at 4:32 PM, Danny Nicholas wrote:
> TAM, Bob! Guess I've got to go through now and "unquote" my literals...
Hi Danny,
Glad that helped. But on second thought, maybe the better fix is to
remove the double quotes in the Gotoif()'s, like thi
exten => s,n(ok),Set(BALCOUNT=0)
>
>
>
> -- Executing [...@tb-account-balance:7] GotoIf("SIP/134-",
> "0?tb-account-balance,s,reset_bc") in new stack
>
> [Nov 3 14:23:02] WARNING[20937]: ast_expr
> Thanks guys. A lot of info here :-)
>
> I am wondering if anyone followed this and it was working for them:
>
> http://scribblej.com/svn/
>
> ???
Hello Bruce
We successfully deployed it and now saving thousands on commercial ASR
ports. It seems users are rather happy with it. The recognition se
On Sun, 2010-05-02 at 09:52 -0400, Dan Journo wrote:
> Hi Bob,
>
> Thanks for that. Is there any way I can make the task run in the
> background and free up the console? Also so that I can disconnect my
> ssh session without losing the task.
>
> Thanks
> Dan
Matthieu
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote:
> How can i log a continuous ping test to a file and include the date
> and time of each ping?
Try this:
#!/bin/sh
for (( ; ; ))
do
NOW=$(date +"%T %m/%d/%Y")
PING=$(ping -qc 1 example.com)
echo $NOW: $PING >> pinger.log
done
exit 0
On 4/6/2010 10:31 AM, Steve Edwards wrote:
> On Tue, 6 Apr 2010, bob gailer wrote:
>
>
>> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
>> the other fails:
>>
>> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09ee
Apr 2010, bob gailer wrote:
>
>
>> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
>> the other fails:
>>
>> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
>> "IAX2/InterOffice/210,300,tr") in ne
e-7578'
== Everyone is busy/congested at this time (1:0/0/1)
The only difference I am aware of is that one server has a public IP
address, the other is behind a NAT.
The trunk from the server with the public address works fine.
I added nat=yes to the other's peer details - did no
On Mon, Mar 8, 2010 at 7:08 PM, Mike wrote:
> This seems like a basic thing to set up, so I have no doubt many people have
> done this. Anyone care to point me in the right direction?
In our config files, we have:
softkey1 type: speeddial
softkey1 label: "Voice Mail"
softkey1 value: *97
This se
On Sat, 2009-12-19 at 08:26 -0500, cov...@ccs.covici.com wrote:
> I have a strange suggestion -- have one extension answer the call and
> dial the extension you want -- then it should ring before dialing the
> second one.
Actually, that is pretty close to what I do on a *1.6 box and it works.
Her
mode=auto
host=
context=inbound
username=
secret=
allow=all
insecure=very
nat=yes
Context in extensions.conf:
[inbound]
exten => 8772709688,1,Goto(cci,s,1)
The context [cci] is shown above.
I appreciate the help, as I am confused!
--
Bob Smither, PhD Circuit Concepts,
http://downloads.asterisk.org/pub/telephony/asterisk/
>
> And any plan for Skype for Asterisk?
>
> Ira
http://www.digium.com/en/products/software/skypeforasterisk.php
(but it is not free)
--
Bob Smither
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On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote:
> Try putting the wait before the Answer.
>
> ...
> exten => s,n,Wait(10)
> exten => s,n,Answer
> ...
Thanks Steve. I tried that:
> On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither wrote:
> > Dear All,
&g
ny useful information - for example the Ringing app is
shown to run, but the caller does not hear it.
Any suggestions?
Many thanks!
--
Bob Smither
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On Mon, 2009-12-14 at 11:49 -0600, Bob Smither wrote:
> This has to be easy, but I have spent a fair amount of time looking for
> a solution to no avail. I am trying to get multiple phones to ring when
> a call comes into an Asterisk box from a particular phone number. What
> hap
. Although phone bt112 is called, it does
not ring. The phone bt116 is called and it does ring.
I have set the call-limit parameter to 10 within the [general] context
in sip.conf.
What am I missing?
Many
> Or charge for full access! Leave a few teasers, and charge some amount to
> see them all. I would pay - even close to attendance price... could only
> help you get past break even ;)
I agree, I would be quite willing to pay for full access to all the videos from
the Confere
The file pbx/pbx_lua.c needed to be changed slightly so as to find lua
on my Fedora9 system
[r...@hoho4 asterisk-1.6.1.5]# diff -u pbx/pbx_lua.c.orig pbx/pbx_lua.c
--- pbx/pbx_lua.c.orig 2009-08-29 14:39:46.0 -0500
+++ pbx/pbx_lua.c 2009-08-29 14:40:20.0 -0500
@@ -42,9 +42,9
What are the HA options for Switchvox systems?
Is it possible to set up redundant systems with DRBD?
I know on the digium website they talk about "Optional cold spare
failover" What does this mean? Is this an active spare ready for some
sort of automated failover?
Thanks for you
On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote:
> I am required to do some thing like Dail in modem .
> User will have to call a modem just like we do in dail up connection
> now we need to handle that request and retrieve some parameters
> from that send a HTTp request to a web serve
On Thu, 2009-03-26 at 07:19 -0700, Vieri wrote:
> Maybe I could simply do something like:
> asterisk -rx "show channels" | grep -c -i zap
> to get the number of zap/dahdi channels in use.
I was actually using a command similar to that up until a few months
ago.
/usr/sbin/asterisk -rx 'show cha
and see if the fix is really a fix or if they are going to
fail again. I dealt with Linda at Aastra:
Linda Berendt
Needless to say, we're not buying Aastra phones anymore. We're pretty
happy with Polycom phones right now.
Bob
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'm not sure what you should do with the
first line in that case.
Bob
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555; secret
LINE2 =
LINE2_PROXY = 1
LINE2_CALLID = NOC Tech
LINE2_AUTH= ; secret
Bob
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h
As soon as I trimmed the config file down to just the necessary
components, the phone started to work!
Bob
On Mon, 2009-02-23 at 21:07 -0500, M Hulber wrote:
> I have a new Polycom Spectralink 8002 and am having trouble with the
> configuration or the unit but I can't see what&#x
I found that after moving to Asterisk 1.6 and the latest SVN of
ASterisk-GUI, the link changed from:
http://localhost:8088/asterisk/static/config/cfgbasic.html
to:
http://localhost:8088/static/config/cfgbasic.html
I don't know if you'll find the same or not...
Bob
On Mon, 2009-02-2
familiar to anyone? I am open to the possibility of
using swift -o to generate to a WAV file, then using that file with
read(), but I would like to avoid the delays and additional complexity
associated with that technique, if possible.
Thanks!
Bob Hartwig
___
On Fri, 2009-02-13 at 15:24 -0500, Miguel Martinez wrote:
> I am new to the Asterisk world, but have decided to use the business
> edition, but am looking for a cost effective gui interface to manage
> the software.
Does the Asterisk-GUI work with Asterisk Business Edition?
___
this link:
http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm
States the following:
"Generic PBXs will not do for our broadcast application – they just
don’t have the features necessary. For example, while lines may
certainly be shared to multiple phones, there is no way to switch g
phones? Some older specs stated
they would be appropriate for businesses with 2-50 users, while the
current spec on the Digium site states they are appropriate for 2-20
users.
The application I'm thinking of would be VoIP only with a g.711 SIP
trunk and g.711 phones
Asterisk system
register to a local phone service using MGCP.
Thanks for your help.
Bob
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ith Polycom Phones?
Bob
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et me know. If there is a technological fix,
perhaps these threads will die down.
Bob G
On Sat, 2008-12-06 at 14:47 +1300, Duncan Turnbull wrote:
> I like the discussion, I doubt it will end.
>
> I prefer top posting because I reply to all my customers that way, my
> mail client isn
On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote:
> It's conceivable, but how would I verify this and how would I change
> it if that was the problem?
There's a few things you can do here.
1) Check the sip.conf on both sides to see what is defined there for the
trunk. Look for some dis
line are going across the VPN
as uLaw while the calls from the sip phones are using a compressed
codec?
Bob
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On Wed, 2008-10-22 at 12:11 -0500, Bob Pierce wrote:
> On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote:
> > Does anyone know what the significance is of the b1 being sent here?
> >
> > Or, is there a way to make Asterisk not send the b1 character as a
> > test?
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote:
> Does anyone know what the significance is of the b1 being sent here?
>
> Or, is there a way to make Asterisk not send the b1 character as a
> test?
As an update to this, I noticed the following lines in libpri.h near
line 236:
t the significance is of the b1 being sent here?
Or, is there a way to make Asterisk not send the b1 character as a test?
I've pasted a portion of the PRI debug trace below.
Thanks for your help.
Bob
-- Making new call for cr 32985
> Protocol Discriminator: Q.931 (8) len=55
>
ls. If I'm not
mistaken, that can mess up fax machine communications.
Bob
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asterisk-us
y this in a few weeks when I start rebuilding the
permanent system that will replace our current temporary system.
That should give us the opportunity to test it on the bench instead of
playing around with the production box.
I'll probably be back
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote:
> > Are there any plans to back port this feature into upcoming 1.4
> > releases?
> >
>
> No, new features are added only in trunk, and released in next major
> release (1.6).
So what would be involved in back porting this feature for our sy
On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote:
> I'd say - go for backport instead. shared_lastcall is commited in
> http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820&r2=86985
> and it seems that there are no bugfixes for it since. So, backporting
> should be fairly simpl
On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote:
> > I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately,
> the
> > shared_lastcall option is only in versions 1.6.0 and up.
> >
>
> Does anybody have a workaround for this in 1.4?
Or maybe
On Mon, 2008-08-25 at 17:46 -0500, Mark Michelson wrote:
> I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, the
> shared_lastcall option is only in versions 1.6.0 and up.
>
Does anybody have a workaround for this in 1.4?
___
-- Bandw
h better now, and the AutoFill parallel
call distribution is really nice.
Bob
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asterisk-users
oom on
box 1, but I can't think of how to do that right now.
Bob
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asterisk-user
On Sun, 2008-07-13 at 22:12 -0400, Noah Miller wrote:
> Hi Bob -
> Are you using any Xorcom hardware? If not, you can avoid this issue
> by disabling the appropriate items when you run "make menuselect"
> before compiling Zaptel.
Thanks! That allowed me to compile.
Re
ere
make[4]: *** [/projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o]
Error 1
...
I am not sure how to correct this.
Any suggestions?
Thanks!
--
Bob Smither <[EMAIL PROTECTED]>
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http://67.169.112.100/openmeetings/ Its OSS, runs on Linux and is not
buggy
- Original Message -
From: "Sanjoy Rath"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Asterisk video alternatives
Date: Mon, 9 Jun 2008 04:41:06 +
Hel
I just got off IM with the owner of Vmukti.Hardik said the Yate woint be
ready untillate August its Astrisk only for nowHe is also planning on
moving to Sliverlight so the servcie will work on all browsers in August.I
build a Free conferencing at 1ezphone.netIm doing any development on it
but its b
look at 1ezphone.net Its based off another OSS and runs on linux the user
Interface is flash like 1ezphone.com
- Original Message -
From: "Matias Surdi"
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk video alternatives
Date: Fri, 06 Jun 2008 10:57:37 +
very full featured
Date: Wed, 4 Jun 2008 20:59:28 -0500
On Wed, Jun 4, 2008 at 5:52 PM, Bob G wrote:
> None of them have features like hold, transfer, voice mail, dtmf,
conference
> as far as I know none of them has caller ID
>
> Only 1ezphone.com has all that and t
"
Subject: Re: [asterisk-users] Browser based VoIP client? None of them
are very full featured
Date: Wed, 4 Jun 2008 20:59:28 -0500
On Wed, Jun 4, 2008 at 5:52 PM, Bob G wrote:
> None of them have features like hold, transfer, voice mail, dtmf,
conference
> as far as I know
IP G450/S8700 system.)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob G
Sent: Wednesday, June 04, 2008 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Avaya IP Phones with *
Yes we do everyday here at Go
sorry its http://1ezphone.com/download
- Original Message -
From: "Bob G"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Browser based VoIP client? -
http://1ezphone.com/downloads
Date: Wed, 4 Jun 2008 17:46
None of them have features like hold, transfer, voice mail, dtmf,
conferenceas far as I know none of them has caller ID Only 1ezphone.com
has all that and the buttons are programmable for CRM features.
- Original Message -
From: "Tim Panton"
To: "Asterisk Users Mailing List - Non-Com
Yes we do everyday here at Google
- Original Message -
From: "Mark Best"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] Avaya IP Phones with *
Date: Wed, 4 Jun 2008 15:24:16 -0700
Does anyone have any experience getting Avaya phones wo
You can download a FREE browser softphone and or cliick to call that
supports UDP athttp://1ezphone.com/download It works well with Asterisk I
use it everyday
- Original Message -
From: "Hilary Miller"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [aste
you can download a FREE browser softphone and or clcik to call at
http://1ezphone.com/downloads Let me know if you have any porblems and I
can help you
- Original Message -
From: "Hilary Miller"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users]
I just finished an integartion with SalesForce.com's connector and it
worked pretty good.
- Original Message -
From: "Fernando Berretta"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] Asterisk - CRM Integration
Date: Tue, 29 Apr 2008 18:16
I use 1ezphone because its not activex and works all operating systems
and browser.Plus the codec is great and only uses 10k
- Original Message -
From: Steven
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Best Click-to-call client
Date: Thu, 24 Apr 2008 08:20:
might work nice here, or is there
some other way of tackling this problem that I may have overlooked?
Thank for your suggestions.
Bob
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Sorry
- Original Message -
From: "Tzafrir Cohen"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Drag and Drop transfer application
Date: Thu, 17 Apr 2008 00:00:57 +0300
On Wed, Apr 16, 2008 at 03:24:15P
Why the guy asked a question?
- Original Message -
From: "Lee Jenkins"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Drag and Drop transfer application
Date: Wed, 16 Apr 2008 16:21:54 -0400
Bob G wrote:
&g
Introducing Click-to-Call
Posted: 16 Apr 2008 9:55 AM PDT
The 1EZphone browser softphone has created so much buzz in the media that
a lot of individual users and companies who have a web-presence;
Websites, Online Advertising, Blogs, Customer support etc have asked for
a Click-to-Call service.
T
Rhino or audiocode PSTN gateway
- Original Message -
From: "mark morreny"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] PSTN to SIP
Date: Thu, 17 Apr 2008 01:25:49 +0800
Dear all, A quick question on deploying Asterisk over E1. I am
Introducing Click-to-Call
Posted: 16 Apr 2008 9:55 AM PDT
The 1EZphone browser softphone has created so much buzz in the media that
a lot of individual users and companies who have a web-presence;
Websites, Online Advertising, Blogs, Customer support etc have asked for
a Click-to-Call service.
T
need.
Have a great day.
Bob
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ge. Now I can be taking
off the messages & some one else can also be taking off the same
messages. We should not be able to do this!!"
Has anyone else seen this? Is there a way to setup the voice mail so
that each box can only be accessed by one pers
You can paste and copy nterface FastEthernet2/0/1 switchport access vlan
20 switchport mode access switchport voice vlan 120 srr-queue bandwidth
share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust
device cisco-phone mls qos trust cos auto qos voip cisco-phone
spanning-tree port
I would like to hire someone to automate my asterisk for hosted PBX
service for fetures like user signup, adding money and call bridging
Please contact me offline at [EMAIL PROTECTED]
- Original Message -
From: "Philipp Kempgen"
To: "Asterisk Users"
Subject: Re: [asterisk-users] Ru
Yes, try http://1ezphone.com its a browser softphone.
- Original Message -
From: Zoa
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] SIP Softphones and Citrix ?
Date: Fri, 01 Feb 2008 23:09:56 +0200
I'm working for zoiper.com and i
gt;http://lists.digium.com/mailman/listinfo/asterisk-users
>
> The Sangoma A500BRX is a 2 to 6 bri pci-x interface,
> although I've never tested it
> (A500BRECX comes with hw echo cancellation).
>
We've been using the Sangoma A104DE in production for almost a yea
On Wed, 2007-12-26 at 17:47 +0200, Tzafrir Cohen wrote:
> On Wed, Dec 26, 2007 at 08:38:31AM -0600, Bob Smither wrote:
> > On Wed, 2007-12-26 at 07:27 +0200, Tzafrir Cohen wrote:
> >
> > > Just to stress the point: you mention that you don't see the card on
> >
On Wed, 2007-12-26 at 10:22 -0500, dave cantera wrote:
> bob,
>
> look on p20 of 'the book' edition 2, or p16 edition 1
> this explains the 3.3v vs 5.0v issue with motherboard slots
>
> http://www.oreilly.com/catalog/9780596510480/
> daveC
The slots are 5V PCI o
On Wed, 2007-12-26 at 15:46 +1000, Mattt wrote:
> Sounds like a PCI bus version issue ;-)
Thanks Matt, but could you elaborate? The card supposedly supports both
5 and 3.3 volt PCI slots, and the slots on the older system that works
with the card appear the same as on the board giving me grief.
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