Can someone post an example of how you read in a channel variable from
asterisk through PHP. I tried the ones voip-info.org but none of them
seem to work, or at least I am not doing something write, but I have no
problem setting variables and other functions, just reading variables
into my
I have tried both ways (with PHPAGI and without), and neither works I
went back to a real simple test, and that doesn't even work.
Here is the CLI:
- Executing AGI(SIP/9897943713-9e04, VoiceMail.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VoiceMail.php
-- AGI
Hey guys, thanks for the suggestions, I finally figured it out.
I need to run the script using the CGI version of php or
#!/usr/bin/php-cgi -q...not really sure why, but it all started working,
AGI classes and all.
Strange, I run it with standard PHP
#!/usr/bin/php -q
Well, if it works, then
Which files must I copy?then..I'll use a ssh scritp for this, I want only
know which files I must copy...
the MySQL files are usually in /var/lib/mysql. The databse you want
to copy is asterisk
hth
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I am looking to make a linux application that will use a SIP or IAX
clinet to connect to my Asterisk server and make calls.
I would like it to be written in C, but beggers can't be choosers. Any
information that would help me with my development would be appreciated.
If you know of a project
I'm getting nowhere with this. Is it even possible to set a variable to the
result of a function call in AGI???
snip
SET VARIABLE DIALPATH ${DUNDILOOKUP(2944093|180net)}
in both cases, DIALPATH is set to a literal
${DUNDILOOKUP2944093|180net}
What am I doing wrong here?
You are telling
[incoming]
; incoming calls from the FXO port are directed to this context from
zapata.conf
exten = s,1,Answer()
exten = s,2,Dial(SIP/polycom)
Try this
exten = s,1,Dial(SIP/polycom,20)
exten = s,2,Hangup()
I think this way, * won't answer the line until your SIP phone
answers. If you
Hmn. Very nice! It works!
On the matter of timing --
Asterisk appears to wait two full PSTN rings before it dials the SIP
extension. Is there any way we can tighten up this interval? Is that
done in the Zap configuration? The driver? The dialplan?
Asterisk is waiting for the CallerID, which is
Should I read the phones from a DB and then write the answer to the same DB?
Does asterisk provide macros,scripts or methods to
open/query/read/write to a db for such tasks?
AGI is your answer.
I would write that in PHP because I'm used to it and it as all the DB
access you will need.
hth
Looks like no body know this
I think your problem is right there in zapata.conf :
callprogress = yes
set it to no. Asterisk think that the other end as not answered yet.
If you stay on the line (ignoring the ringing) you will be
disconnected after you dial timeout parameter.
hth
On 5/30/06, Erick Perez [EMAIL PROTECTED] wrote:
I have the following extension to dial outside via SIP
it's like this:
phoneasterisk-internet-SIP providerUSA
exten = _91NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN},55,o)
exten =
Can you, or anyone else comment on the speakerphone ability of the GVX-3000
? We run the GXP-2000's and for the most part are happy with them, but for
upper management we're looking at phones with better speakerphone. These
would be ideal if the speakerphone isn't as terrible as the GXP-2000.
How do you enable call waiting on sip phones? Ive looked and googled and
can only find call waiting pstn phones butnot for sip. Is their a way of
setting this up within the dailplan?
Which SIP phone ?
If you're using [EMAIL PROTECTED], you have to dial *70
hth
How is it implemented within the dialplan and can call waiting be
implemented for softphones? Is their a way to do this. In my sip.conf file
for one of my configured softphones ive used the limit-call parameter to
limit calls to one across this channel or softphone. Would this invalidate
I gathered that but it has its uses. Could you then give us soem tips on
how to get this working. Call forwarding is a done deal but i cant seem to
find any info on call waiting anywhere? Help needed. Customer fustrated.
Are you using [EMAIL PROTECTED] ?
If not, are you using AMP (now
3.Is there any other way to complete asterisk configuration from database?
Have a look at this : http://www.voip-info.org/wiki-Asterisk+RealTime
hth
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I'd like to keep some of analog lines to support modem, fax and some older
stuff. What's the best choice? A channel bank or a TDM2400P card?
Can I use a TDM2400P board together with the actual TE410P?
As far as I know, Digium doesn't support FAX through TDM2400P, even
less a modem call.
I had to
Nuthin beats an Atlas:
http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8
A.pdf
Telephony Swiss army knife. You can make it do anything. Be prepared to crap
your pants when you see the price, though.
At that price, I'll keep my dedicated analog line.
but thanks for
I downloaded recordPad and recorded a wav file and tried playback on
asterisk got the same error as before -- WARNING [1225991360]
Format.wav.c:132 check_header:unexpected header size 18--
when I recorded in gsm format on my laptop asterisk did playback well
I used sox to resample the recorded
I used it on a 4 digit extension
Can we see the relevant part of the dialplan ?
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I can't believe i didn't see that!
i spent ages staring at those damn logs...
And I spent ages scrolling to the bottom of that email.
Please trim your reply to contain only the relevant parts.
This email alone is longer than the full thread
Thanks
It passes Configuring VLAN.
It gets its own and gateway IP addresses from DHCP server but remains stuck
in Configuring IP...
No FTP server is defined.
I believe Network Configuration menu is locked because Configuring IP
process in going on.
If my memory serves me right from last time I did
I have the following configured for my queue. However, it seems that
because I have 'memberAgents' setup people still join the queue even
when no one is logged in!How can I have agents assigned like this,
yet still not allow joining the queue if they are not logged in?
I think you have to
I have been having a strange issue with my Asterisk 1.2.1 server. I have a
TDM400 for the three POTS lines I have and I can receive calls without any
problems. But sometimes (not everytime, but 70%) when I dial out of those
lines it drops a number and of course gives me the telco error
Antonio,it changes slot of tdm04b and restarts the server.
Since he said in the email that the machine only had 1 PCI slot, I
don't think he can do that
Check in your BIOS, some let you assign a specific IRQ to the PCI slot
hth
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I was wondering if anyone has experience with intercepting an incoming
call. For example, suppose I'm away and see an after hours incoming
call on ZAP going to vmail. I'd like to answer that call. I've seen a
couple of ways to do this, but I'd like to know what anyone out there
is using
Anyone have anything else to add? Thanks
Well, if you can't have your hardphone auto-answer, my IAX softphone
can auto-answer incoming calls, as CallerID display and now support
receiving URL.
So you could send the agent an URL which would display the information
from the caller in a browser as
snip
echocancel=yes
echocancelwhenbridged=yes
You probably want that off, since when you bridge 2 ZAP channels you
should not have any echo.
echotraining=yes
snip
But, when I run asterisk and ask it about the zap channels, I get that echo
cancellation is OFF.
zap show channel 2Echo
Ok I can get this to work now the next problem is since the agent stays
off-hook when a call is presented to them there is no indication of what
call this is. Being an inbound call center we have 100's of clients. 1,000's
of toll frees and DNIS. We use the Asterisk callerID function to assign a
At that cost of downtime I would grab a module just to be sure and stick it
in, no need to actually configure that zapata channel but I would think the
card needs some module.
And what about an X100P clone, would that give an accurate timing source ?
you can try the following:
exten = s,1,Gotoif($[${CALLERIDNUM} = 1130851536 ]?10)
exten = s,2,Goto(from-pstn,s,1)
exten = s,10,disa(no-password,from-internal)
This would do the same without an IF
exten = s/1130851536,1,disa(no-password,from-internal)
hth
Why I did to mine is modify all the internal Vertical Service
Activation Codes to be **x instead of *x. There is probably a
better way, but this worked for me.
We tried that, but users report they are still having the same problem
(the site is located in a different country so I can't
In the PAP2's setup there are all of these Vertical Service Activation
Codes that start with star and Outbound Call Codec Selection Codes,
also the setup menu is accessed by pressing star four times, could they
be intefering with dialing numbers that start with a star? And is there
any way to
* Does anybody know of a softphone that works with Asterisk's
SendURL command? Cross-platform would be nice, open source ideal.
I'm currently working on an updated version of my MediaX phone and it
supports receiving URL. It works only on windows and is not open
source. But if you want to
Is this a problem? What is dnd anyway?
Not a problem, probably dialparties.agi checking if this extension as
DND enabled.
DND stand for Do Not Disturb
hth
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So, how do you know which conf files one can hand edit versus those that
might be overwritten?
You may only change the *_custom.conf files. :)
And the *_additional.conf files are the ones overwritten by the config
in the DB. So you can edit the other ones.
hth
You could, but it'll get overwritten by any FreePBX upgrades. The *.conf
and *_additional.conf files are controlled by FreePBX and can be
overwritten.
I thought I should clarify this statement: I meant that FreePBX could
overwrite both the *.conf and the *_additional.conf files. You are
How can I install a softphone on my USB flash drive like Xlite and have
it ready to go when I plug it in at any Windows XP computer?
(Same for a Linux softphone, both on one USB flash drive).
I believe Dan's softphone is suitable for this. See
http://www.laser.com/dante/diax/diax.html
There are 2 issues here.
1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to
transport audio for SIP (and other protocols). This means that ANY
jitter on the SIP Phone - Asterisk link will cause audio problems.
2) Asterisk times it's outgoing audio based on the incoming
The latency is very high, in that, it picks up after 8 rings. I don't know
what I can tune to reduce to 2 or 3 rings. If it's of any help , I am
posting a section of the log :
Do you get CallerID on that line ?
If, in zapata.conf, you have it set to get the CallerID
(usecallerid=yes) and the
is it possible to make an outgoing call transferable for the dialing phones
like the 't' or T option on the Dial-Command does this for incoming calls?
This is exactly what the option T does.
't' -- allow the called user transfer the calling user by hitting #.
'T' -- allow the calling
I am trying to integrate Asterisk with traditional phone central, this issue
is sometimes tough. After some testing and measuring I think what is
bothering my Asterisk; I need to dial a number digit after digit and not the
whole string, so for example:
1, 2, 3, 4, 5, 6
and not:
123456
How can I
[from-pstn]
include = from-pstn-custom ; create this context in extensions_custom.conf
to include customizations
include = ext-did
;exten = fax,1,Goto(ext-fax,in_fax,1)
exten = _.,1,Wait(1)
exten = _.,2,Goto(from-pstn,s,1)
Here is what is happening :
Your ZAP channels are in the context
On a related issue, at locations where we have 3 or 4 phone lines connected
to asterisk and they are all in use and someone dials 911 we want it to
disconnect one of the active calls so the 911 call can be made. Does
anyone know how to do this? Would I need to use a device like the above or
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
;exten = fax,1,Goto(ext-fax,in_fax,1)
exten = _.,1,Wait(1)
exten = _.,2,Goto(from-pstn,100,1)
Try somethin like
[from-pstn]
include = from-pstn-custom ; create
I actually tried that before but it didnt seem to work. I tried once again
and still nothing rings, whether I set the destination to a single
extension, or a ring group. But the suggestion from another user below did
work, but wont go to voicemail yet when its not answered.
[from-pstn]
My queues.conf looks like this:
[sales]
musiconhold = default
announce = queue-sales
strategy = ringall
wrapuptime=15
timeout = 30
maxlen = 0
announce-frequency = 90
announce-holdtime = yes
monitor-format = wav
monitor-join = yes
leavewhenempty = yes
joinempty = no
member =
What I would like is:
If someone calls extn 281 on my Alcatel PBX it is routed through to Extn 233
on my * thruogh FXO port/module 4
If someone calls extn 282 on my Alcatel PBX it is routed through to Extn 234
on my * thruogh FXO Port/module 3
I have SIP extn 233 set up.
I have SIP extn
I was able to configure (Incoming Calls) through AMP to make asterisk
answer my line after 3 rings and forward it to an extension. However, I was
unable to disable that feature?
In AMP, go on Maintenance-Config Edit
In you zapata-auto.conf (assuming you used genzaptelconf), change the
we would like to build IM-Voice community for our students around Asterisk,
Jingle, Jabber.
Can we already test those features ? Anyone already running such setup? Any
more info ?
Have you looked at Wildfire ? http://www.jivesoftware.org/wildfire/
There is an Asterisk-plugin that update
When dialing an outbound number, sometimes all the digits are not dialed
properly on the outside line. In the dial plan I added a SayDigits to the
outbound rule and it properly reads back the entire number that was entered
on the phone before dialing.
Is asterisk dialing too quickly, is
I am having issues with a TDM2400P. It appears when the ZAP channel dials
out, it randomly chops the first digit off of the number. I have tried
relaxdtmf=yes, turning up and down the txgain, turned off and on the echo
cancellation, generated new zaptel (with updated spinlock.h)...
I am at
So what context should I put a in?
If your voicemail context is default (that is, is what context the
mailbox is), you have to put the a exten in the default context.
hope that clear things up
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We are faced with a problem concerning queues.
When we have several calls in different queues, is there some sort of
way to open a channel between a (sip-)phone and a SPECIFIC call in a
queue using the Asterisk manager api?
We would like to do this even when we are not a member of that
Can Comedian Mail handle more than just an away and busy message? I've got
a client that would like even more of them.
I can write an app to replace messages externally, but I was wondering of
comedian could handle it internally.
As far as I know, no.
But, what I did for a customer
2)Read the book Asterisk, The future of telephony.
You can buy it or download it for free. I dont have
the link to it but if some one else does please post
it.
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
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how can I send text directly to a specific device, something like:
exten = 103,1,SendTextToDev(SIP/7, hello) ??
I don't think you can send to a particular device, but you can send it
to the device calling if it support it.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SendText
Im developing an ip telephony project and i need some help in order to
choose the better PCI card, the options at the moment are digium, sangoma
and voicetronix, the strongest ones are digium and sangoma but i dont know
how justify the election
If you want people to answer, maybe you should be
When a friend calls, I would like for him to enter a 4 digit password
in order to access to a sub-menu, if no password is entered, then the
welcome msg is said ...
Any hints on how to do that ??
In your incoming-rtc context, define an extension (let's say 1234)
exten =
Anyone knows if the SC430, based on the Intel E7230 chipset, is
compatible with the Digium cards? I've tried the compatibility page on
digium's website. It seems like they've pulled the old compatibility
list, now the links on the page only point back to the product pages.
Over here, Dell is
After updating your sip.conf and extensions.conf, did you reload
asterisk? Asterisk caches the config files and does not re-read them unless
you issue a sip reload, extensions reload or an all-in-one restart when
convenient at the CLI.
Actually, the all-in-one is done with only reload, no
When going option 5 you can dial some extensions such as 2802, it goes to the
extension (all extens start with 28 on the
system). However, just dialing something random like 2929 sends the caller
to option 2 of the main menu or 1010 sends
the caller to menu option 1 from the main menu.
I have a situation where I have 8 lines from the phone company in a hunt
group coming in to my asterisk box. These are the same lines I'm using
for outgoing calls ( named g0 ).
snìp
Is this possible? If it isn't, I plan to reverse the order in which the
lines are connected to my * box,
a message and it notifies the on call techs. My question is regarding
externnotify for the voice mail.conf. If I enabled that and set up a
call file, will it do it for every voice mail box I have on the system?
Is there a way I can limit it to just the one voice mail box on the
system? If
which doesn't work. So, what exten regex can I use that would catch anything
dialled, or how can I stop Asterisk from executing the AGI script a second
time when I use _.?
I think you can just add an extension h in that context, something like
exten = h,1,Hangup
hth
A client call.
A user answer.
Another user, a manager, for instance. Dial a code:
For instance:
exten = 1010,1,() #Start to listen the call placed in the channel 1
exten = 1011,1,() #Start to listen the call placed in the channel 2
And so on...
What you are looking for is
What's the best way to increment a numeric variable in the dial plan?
I tried this...
exten = s,1,Set(mainLoop=${MATH(${mainLoop}+1)})
exten = s,1,Set(mainLoop=$[${mainLoop} + 1])
hth
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[internal]
exten = 5148346,1,Dial(Zap/g1/514836)
Anybody out there have any ideas on why all of the digits aren't being sent
out?
Shouldn't this be like this ?
exten = 5148346,1,Dial(Zap/g1/5148346)
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When agent tries to transfer a phone call (*2 - att transfer) he hangs up.
Why? When a phone call isn't from queue then att transfer works fine.
In features conf I have *1 for recording, *2 for att transfer and #1 for
blind. In queue blind transfer works. For disconnect I have #0.
I guess
I currently have a GotoIf statement that goes to a special extension
priority if the CID match with one of the numbers in my list of CIDs. The
way I've done it now is by multiple OR operators. There must be a better
way. Anyone got some suggestions?
This is basicly what I want. If CID
Oh. So how can I do this?
If I write something in PHP, how do I make it output to an Asterisk
variabel? I need to set a variable in asterisk to TRUE or FALSE based on the
result of the PHP-script.
You can use PHPAGI, this will make your life easier
http://phpagi.sourceforge.net/
hth
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get the system prompts to then speak
the selected langauge. For example, a caller has selected Spanish and so
is routed to the Spanish part of the IVR. At some point
On 1/27/06, roswel ajf [EMAIL PROTECTED] wrote:
we have got asterisk 1.0 (over 1 yrs old) version and very old zaptel
version. That code is working only with 8 or less ports (accumulative) on
digium fxs/fxo cards (2 cards with 4 ports each).
A lot of improvements/bug-fixes as gone in Asterisk
This may be obvious but I have not found the answer in the archives or web
searching. I am in the process of transitioning to Asterisk. While I have
two systems connected to the same PSTN line, I want to configure Asterisk to
not answer an incoming call. Is this a setting that you would
I can't find how to force an asterisk server to stay in the middle
between two asterisk clients, the iax2 reinvite pulls the call out of
the cdr, which is no good ...
The trick is to use some Dial options that forces * to stay in the
path, like t,T,h,H,w or W
See
Why not just put 'notransfer=yes' in the appropriate iax.conf user/peer entry?
Oups, answered too fast. That is what happens when I try to answer a
technical question before finishing my first coffee.
Thanks for the correction
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In my case, the phone number to forward is 3473774567, and the extension is
105, hence the syntax should be:
exten =
3473774567,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:105})
Let me explain you what this syntax is saying :
presuming this number is called from extension 7001
- Put in the DB, under
Wanted some advice for the docs that you'd recommend someone new to
Asterisk to read. I have a good knowledge of Unix and networking, so
that part shouldn't be a problem.
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
Welcome to *
Can you please explain?
Whats CM?
I think this is for (Cisco) Call Manager
Whats Astericks?
Maybe it's in the same part as Atérisk (only the french-speaking
will laugh this one)
;)
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I have been looking at the reference manual on asteriskguru.com. They
say it's a timeout but they don't indicate the units. Is it
milliseconds, microseconds or seconds?
Just for the lazy one, here is the link to the DIAL command
http://www.voip-info.org/wiki-Asterisk+cmd+dial
and
On 12/8/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I understand. But because the majority of calls are not to be
recorded, I don't have a need to keep Asterisk in the media path all
the time. That's why I'm wondering if you could dynamically keep it
in the media path or not.
Some options of
I'm trying to figure out how to setup live recording of a phone call.
I've read all the docs at the wiki, but can't seem to figure out how
to implement it.
I'm running asterisk 1.2
I have the Polycom IP500 SIP phones.
In a perfect world, I would dial something to start recording, and
then
That helps, but I'm still missing one piece.
I want to be able to press a button during the call to start and stop
recording.
I tried using:
exten = s,1,Dial(101,20,Ww)
But it doesn't seem to do anything.
On the console, put verbose to something like 50 (set verbose 50) and
you should see
Hi!
I have been using Asterisk-1.0.3 for quite some time now.My main aim
nowadays is to make iax-iax calls for which i am usin DIAX soft phone.The
problem is that sometimes the phone doesn`t register and at others it gets
out of the registration(after being registere for some time).Can anyone
Because the stock firmware does not support QoS. You need one that runs
linux and then load the hacked firmware by either sveasoft or I prefer
OpenWRT since you can run OpenVPN other packages.
Actually, QoS is in the standard frimware since somewhere around
version 3.something
Can anyone recommend a proper Softphone?
Have you tried mine : http://www.marccharbonneau.com/asterisk/mediaxphone.php
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I have seen some work on various IAX activex phones but not really sure if
they can do what I want or if there is something that can.
I just want a simple link on a webpage that dials an extension
automagically. I do not need a nice phone displayed, the ability to dial
any numbers, just
Hi there,
is there any free softphone that i can customize accoring to my needs ??
You could use IaxComm : http://iaxclient.sourceforge.net/iaxcomm/index.html
hth
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suggest me some resources where to start over
IAX library : http://iaxclient.sourceforge.net/
softphone with sourcecode that use this library :
http://iaxclient.sourceforge.net/iaxcomm/index.html
hth
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} 1) what ports to forward on the router to the Asterisk machine
}
} For SIP: 5060 1-2
} For IAX: 4569
} (This assumes that you haven't changed the defaults in sip.conf/
} iax.conf)
If I understand correctly, I only need IAX on the LAN since I am using a
SIP provider. Are those
ALSO, on an unrelated note, i want to pass back info back and forth between
my AGI script and Asterisk. I know you can pass info INTO AGI, but can you
pass the info back OUT of AGI into the Asterisk extensions.conf dialplan?
You can set variables from AGI and access those from the dialplan
Rise and shine you sweet wonderful geek
That prompt would be perfect for a wake-up call.
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can someone tell me about a good iax softphone ??
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php
works only on windows
for one that works on Windows and Linux :
http://iaxclient.sourceforge.net/iaxcomm/index.html
there is also DIAX :
My question is, will this support more than 1 simultaneous connection from
the same outside IP address, or will only one soft phone function?
or, put another way:
Can multiple soft phones (running on separate computers) be used
simultaneously from the same outside IP address?
I've
Is there a place where all the parameters are documented ?
In example (my example!) I would like to know the meaning of a lot of
parameter that can be used in sip.conf,
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
How did I found this ?
I saw an add in my latest Linux Journal advertising Sangoma's new AA
series of FXO/FXS analog cards with on-board echo cancellation, but I
can't find any information at all on them. Even the link given in the
advertisement is a dead end as far as I can tell. Anybody else
seen/heard anything
I have the same problem, after about a month the card doesn't report any
incoming calls anymore to the console. Don't know the rev of my card yet,
unloading asterisk and unloading the modules and then restarting
everything does seem to help though, no need to reboot.
There was a bug
I'm curious if someone could point out a dirty trick to get the voice to
play right, for internal and external callers, depending upon what number
they dialed into, or what organization the user belongs to, without massive
dialplan facelift.
You could probably just use the language variable,
Hi,
Is there somebody using an Access Bank I with Asterisk that could
share the secret ingredients needed to make it work ?
I've searched around and found some info, I tryed almost every
configuration possible but I can't seem to find the right combination.
If someone could provide me with the
The main regret I have about hardware RAID is that the card is sharing
an IRQ with one of the Digium cards. This whole IRQ thing is driving me
crazy ... I disabled everything I could in BIOS and that freed up some
IRQs, but there's no way to assign a particular IRQ to a particular
device. I
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