Have you considered putting an advertisement in the newspaper?
PaulH
On 05/11/09 06:43, Carlos Cuervo wrote:
Hello,
I've been tasked to look for ways to resell to others the service that
one of a trunk provides.. In other words, i want to configure my
current Asterisk (Ver. 1.4.26.1) with
On 29/10/09 22:40, Matt Riddell wrote:
:D
I should hope not!!
If everyone was as smart as me, how would I take over the world?
With violence, just like everyone else!
PaulH
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I have used both misdn and dahdi_bri over the last year, and would happy
take dahdi if for no other reason that it's much easier to install.
A patch is available to allow dahdi_bri to work with Asterisk 1.4, and I
have used that successfully.
PaulH
On 25/10/09 03:26, Olivier wrote:
On 24/10/09 00:59, Lyle Giese wrote:
PATRICK KANGETHE wrote:
I want to interface asterisk with a legacy pbx that has around 23
extensions through my 8 fxs card, how do i work around this?
Hint: I have already terminated 8 extensions from the legacy PBX, i
was thinking whether i can peer the
Jeff LaCoursiere wrote:
Steve Edwards asterisk@sedwards.com wrote:
Since I'm an old-school C programmer, I use emacs as my editor. I fire
up gdb (the GNU C (amongst other languages) debugger) in a window, give it
a command like b main; r dummy-input-for-block-ani and I can step
The list will need to see your dialplan or a CLI dump to help you with this.
PaulH
B.Masoud @ SH wrote:
I am not sure why I am getting this message,
I have an outbound route that goes to asterisk gateway1 then asterisk
gateway2
When all lines on asterisk gateway1 are full, I get the
I have used the group function to limit the calls entering a queue for a
similar reason to yourself.
PaulH
Niccolò Belli wrote:
Hi,
I explain what I want to do..
All the operators share their phones. The number of the operator isn't
constant, so it's possible that two operators share all
Is inbound working?
Can you see action on the CLI when you send a call to the lines attached
to the card?
PaulH
B.Masoud @ SH wrote:
Hi
I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls
to that trunk, I am getting all circuits are busy now, do I have to do
Death to all sip users!
Paulh
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Built for firefox.
PaulH
Stefan Schmidt wrote:
Hello,
iam searching for an Firefox plugin which can make an sip Invite and
Redirect after 200 OK, so i dont have to use a softphone, just to
initialise a call by clicking on a number
i've found
but still dont found proper solution
in,fact i need dialing on IP base in which dialing by using IP address
will send call to remote machine or same machine
regards
Dhaval
On Fri, Sep 18, 2009 at 5:59 PM, Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au wrote:
I have
I have used the SIPPEER function to find if a phone is local and
available before.
PaulH
Asterisk User wrote:
Hi,
I have a generalized syntax for dial application in my dialplan where
I send calls to particular server.
Here is my dial sysntax...
exten =
It's easier to work with the closed hours then - use a goto just for
Sunday/Monday
PaulH
James Hankins wrote:
Greetings folks, new to this, trying to get the syntax correct for a
day of week routing.
exten = 345,1,Answer()
exten = 345,n,GotoIfTime(10:00-17:00|tuethusat|*|*?open,345,1)
Has anyone seen this one before?
full:[Sep 14 11:49:01] DEBUG[15771] chan_sip.c: SIP attended transfer:
Error: No target channel
It coincided with a failed attended transfer...
Ideas?
PaulH
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I would strongly suggest you browse:
http://www.asteriskdocs.org/
Kind regards,
PaulH
Songtao Yu wrote:
Hi All,
I am new to Asterisk. Now I got one question on the identifier
parameter of the Dial() command. I saw as below:
exten = 20,1,Dia(Zap/3/5551234).
Would you please let me know
I have also seen:
PSTN asterisk legacy
Which also gives you a migration path
PaulH
Research wrote:
Hello team;
While am aware and active user of astersk monitor function for
recording, i would like to know if i can use asterisk as a pure
recording server(like nice or
to be queued, you create a queue with only one member, and
have agents log on and log off as necessary; if you don't want callers
to be queued, likely I would not use a queue but woul dial the agent
straight.
l.
PS. this is quite an unusual requirement, what is it for?
2009/9/1 Paul Hales pdha
Matt Riddell wrote:
On 3/09/09 11:34 AM, Paul Hales wrote:
Hmmm.any idea how I can use hints to monitor their mobile phones?
Unless the call came in via Asterisk, you can't.
The calls will - so it should be able (at the very least with the
asterisk internal DB - which I
could then call Local/1...@gaents directkly or make it a member of
the queue (with known issues on some version of *) :-)
l.
2009/9/2 Paul Hales pdha...@optusnet.com.au
mailto:pdha...@optusnet.com.au
A situation where staff want a mobile and their SIP handset to
share an
extension
is set to BUSY, otherwise set to AVAIL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales
Sent: Wednesday, September 02, 2009 2:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I couldn't find any information on this brand of phone on the internet
at all.
PaulH
hadi motamedi wrote:
Sorry for lack of enough information . I mean my subscriber when goes
off hook he will see his own number displayed on his phone . I need to
disable this feature on my Asterisk .The
I have a _very_ specific situation where I need queues to work in a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant past (pre 1.0)...but I am
Miguel Molina wrote:
Paul Hales escribió:
I have a _very_ specific situation where I need queues to work in a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked
But they do taste similar.
PaulH
Darrick Hartman wrote:
Polycom sip.cfg is not the same as the Asterisk sip.conf file
hadi motamedi wrote:
Thank you for your reply . Please find attached my Asterisk sip.conf .
Can you please let me know what modifications are needed ?
Regards
Matt Riddell wrote:
What is a subs?
A submarine. I think.
PaulH
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Sticky Park sounds like somewhere you go late at night wearing a plastic
raincoat.
PaulH
Mat Murdock wrote:
My company for various reasons has asked that I come up with a way to
have previously parked calls be re-parked in the same parking slot. I
have looked at setting up asterisk so
Use a standard network cable - but you have to activate the 'terminate'
jumper on the NT end.
- Also, the new BRI stuff in dahdi is much easier to work with than misdn.
PaulH
voip crazy wrote:
Hello all,
I'm trying to conect two asterisk servers using two B410p Digium
cards. One card on
Can I assume that you meant to add that the person's phone would be used
to listen to the message?
PaulH
Olivier wrote:
Hi,
I've lastly read a Request For Quotation asking for a software option
I've never heard about before.
It's about a player plugin with which, when using an
In australia, I would usually suggest a mix of E1 and SIP for calls - it
doesn't cost any money to receive calls via E1, and redundancy is an
old, valuable friend of mine.
PaulH
Stephen Fierbaugh (PBT) wrote:
I am a Linux sysadmin who has been tasked with developing the phone
system for our
Can I assume that your project has stalled?
PaulH
logan wrote:
Thanks Paul. Your help is much appreciated here.
I don't really understand this question - Asterisk can make calls over
phone lines. And it does it well.
Surely, Asterisk does that well, but Asterisk needs to have
Some thoughts inline:
logan wrote:
Hi Paul,
Thanks a lot for the response.
I'm a novice so pardon me for the stupid questions. I thought that maybe the
PSTN lines don't allow more than 1 simultaneous calls on a line, but on GSM
it might be possible.
I basically want to know how
logan wrote:
Thanks Paul. Your help is much appreciated here.
No problem - been working on telephone systems for about 12 years now -
which doesn't even make me an old hand...
Surely, Asterisk does that well, but Asterisk needs to have multiple phone
lines for that. I thought that a
Sadly, at the end of the day the answers will probably be no, no, no and no.
PaulH
logan wrote:
Hi,
I'm an absolute newbie and wanted to know the following.
I want to have a setup where I have a PSTN line connected to my
Asterisk box and want to know if it is possible to make more than
logan wrote:
Hi Trevor,
Thanks for the response.
I was thinking if a GSM to VoIP gateway can do the job of multiple outgoing
calls. I could be wrong but seems like cellphones do allow you to make
multiple calls at a time (only one is active or one active conference). If
we assume that
Always a great readthanks.
PaulH
Alex Balashov wrote:
Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's
classic How to Ask Questions the Smart Way to the OpenSIPS-users
mailing list[1], I'm going to repost it here:
Yes, but not as your first Asterisk implementation.
PaulH
gergis.rasmy wrote:
i am asked to implement a call center of 50 seats for my company , and
i was wondering if Asterisk can fit this as a relaibale and low price
system
is it mature enough for this task??
best regards
Gers
'One touch park' was designed to work around this issue.
PaulH
Danny Nicholas wrote:
Hi gang,
When I try to park a call using blind-transfer (#1), the caller hears
the lot instead of the transferring party. Attended transfer and blind
transfer from the phone buttons (Polycom 501) work
The queue option
ringinuse = no
might be what you are looking for.
PaulH
Kev Szaszvari wrote:
Hi All
I am using asterisk 1.4.21.1
Im not sure if this is a issue but it has become one for me :)
When agents are logged in to a queue (AgentCallBackLogin) and they receive a
direct line
= default
strategy = leastrecent
timeout = 5
retry = 1
wrapuptime= 3
autofill = yes
autopause = no
maxlen = 0
joinempty = yes
leavewhenempty = no
ringinuse = no
- Original Message -
From: Paul Hales
[mailto:pdha...@optusnet.com.au]
To: Asterisk Users Mailing List -
Non
I can definitely recommend the 'sit down and play with it' website.
Worked for me.
PaulH
David @ULC wrote:
What the best website and book to start learning asterisk ?
___
From memory, it is doable but this is a feature that Polycom never quite
finished writing.
PaulH
On Fri, 2009-06-19 at 10:58 +0200, Louis-David Mitterrand wrote:
Hi,
Is there a way on Polycom phones to show an agent whether he is logged
in or not?
Farooq Hussain wrote:
Dear All,
Please help as I new to Asterisk. I want know something what is Trunk
what it will do. And I want to create a Dialplan like bellow:
1. It ask to dial a extension.
2. User will dial a extension.
3. User will be routed to that extension.
I also want
Alex Samad wrote:
Hi
I have an account with mynetphone (australia), which gives me two voip
(sip) accounts, which i used to have connected to a spa9000.
this is behind a firewall, so on the spa9000 I would listen on another
port apart from 5060. so on the firewall 5060 would go to voip1
Todd S wrote:
What's the bets way to verify T.38 is being used on both incoming an
outgoing transaction?
3 to 1 in favour of not working. ;)
PaulH
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Digium PSTN cards seem to work.
PaulH
Manoj Panicker - FOES wrote:
Hi
Which is the best interface card to connect* PSTN* line with
Asterisk. Can somebody please help. My intention is to route the
incoming PSTN calls to internal IP Phones through Asterisk and Vice
versa. The
Not true. I am always wrong.
(wait...is that a paradox?)
PaulH
ContactTel Business wrote:
Niecly said.. hoeever, these list are not for astrix users, butt for
bashing, didnot you realise this ?
It had where 4 years more , know that this is fluent in this site.
Translated as in :
I think you have your line types mixed up - FXS is for phones, FXO is
for lines.
An analogue passthorugh setup _is_ doable, just not overly recommended.
PaulH
Alex Samad wrote:
Hi
I am in the middle of move a small business over from legacy PABX + PSTN
lines to VOIP infrastructure.
I
Alex Samad wrote:
On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
I think you have your line types mixed up - FXS is for phones, FXO is
for lines.
sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
that a attached fxs presents internally as a fxo
I
Have you tried plugging analog phones into the FXS ports in the Asterisk
box?
That should let you know what the Asterisk is really doing with it's FXS
ports.
PaulH
Alex Samad wrote:
On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
I think you have your line types mixed up
Alex Samad wrote:
On Thu, May 14, 2009 at 03:31:18PM +1000, Paul Hales wrote:
Have you tried plugging analog phones into the FXS ports in the Asterisk
box?
good ideal, but trying to find an old style phone the site has a
commander PABX with digital handsets. I will see if I can
Alex Samad wrote:
On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote:
What happens if you make a call in from the old fax line and send that
over to the old PABX? Does that work OK?
not sure what you are asking here. I have checked an incoming call
through the FXO(PSTN
I got very excited when I read the title of this email - I was hoping
someone had learnt to speak g729.
Ah well.
PaulH
Adrian Marsh wrote:
Hi,
I’m having problems with an asterisk server that’s not offering Codecs
for ulaw and alaw as it should.
I’ve three servers in total: a1, a2 and
You don't need freepbx.
In all honestly, building a system from scratch isn't too bad if you
having some decent (or indecent?) linux karma.
If you don't, it's going to be a rather unfun time.
PaulH
John F. Ervin wrote:
So, people have recommended building a system from scratch, start with
George Kwabenah Appiah wrote:
Are there (??) instructions for people who are experienced at the
Trixbox level but wish to move on?
If you'd like to get a solid foundation on Asterisk and how the
various pieces fit together, I suggest you invest a couple hours and
go through the
I am still not sure what you are askingis it something to do with NTP?
PaulH
Daniel - Asterisk wrote:
I guess it was a problem with my connection, here the complete question..
Dear all,
I wanna know what can I do to get the PBX's clock from an external AMI
server, especially with
Steve Howes wrote:
On 28 Apr 2009, at 16:49, Daniel - Asterisk wrote:
Dear all,
I wanna know what can I do to get the PBX's clock from
You sir, are made of fail.
I had to admit, I laughed.
PaulH
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Is a drive image out of the question?
PaulH
David Klaverstyn wrote:
Hi All,
I’m in the process of writing an install script and I would like to
change some settings for the install process but I don’t want the user
to go into menuselect and make the changes manually.
Is there a way to
I would upgrade to the latest 1.4, if stable is what is needed.
PaulH
Gabriel - IP Guys wrote:
Dear All,
I have a asterisk setup that is currently running on version 1.4.15 –
I wish to upgrade or migrate this instance to the current asterisk
stable, 1.6.0.6. It is my intention to build a
If there is an asterisk users group in your area, visit and ask lots of
questions.
PaulH
Roland Roland wrote:
Hi all,
a few month ago I got the task of setting up asterisk for my company.
I had 94 employee to set this up for ...
I never heard of asterisk before to b honest, so after
Karl Fife wrote:
On the landing page of the Polycom web site there's a We're
listening nanosurvey, asking what is the one thing Polycom can do to
improve their products. The link points here:
http://polycom.zuberance.com/survey.htm
I wrote a sentence about tweaking the user interface on
The Asterisk console is pretty goodbut there was a text version of
one of the softphones once (sjphone, if I remember correctly)
PaulH
Hans Witvliet wrote:
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
Are both of your agents logged in?
What does the CLI show?
PaulH
edw...@web.de wrote:
Hi,
I have a problem with queue strategy. But only 1 of my agents ring,
when someone call.
my queue.conf:
[MyQueue]
strategy=ringall
member = Agent/201
member = Agent/202
announce-holdtime = yes
Joseph wrote:
I'm using Asterisk 1.4.22.1
When I'm on active call it happens many times the call gets interrupted by
music-on-hold without my pressing any button.
MOH just kicks in and int erupt the call and I have no way of getting the
call back.
Did anybody experienced anything like
Any idea what this means? And why they are different?
-
Extension Changed 22142[default] new state Idle for Notify User 31001
(queued)
Extension Changed 22142[default] new state Idle for Notify User 30060
-
I have googled and searched, and can't find anything on this subject.
Does anyone
SIP wrote:
I believe SNOM 300s do PoE (might have to check that, though) and are
around $100. We've little experience with them, but we use an office
full of Snom 320s, and we're nothing but pleased with them. Good
speaker, good handset, lots of excellent options. And reasonably priced.
I am probably missing something, being a newbie. I have a 4 port
fxs/fxo (2/2) card.
My land line is going to one of the FXO port and my home phone is connected
to one of the FXS port.
I want to be able to call my phone number from external phone (cell phone)
and have my home phone ring.
I can't force them to use star codes to set DND in astdb).
Once again, someone who underestimates the power of physical violence.
PaulH
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To
I wish it was available too - I have just had to back dahdi out of a
system and revert to misdn after a whole day of testing.
PaulH
Andrew Thomas wrote:
I have LibPri installed and working (.../wPRI).
So, if I understand Tzafrir correctly - DAHDI support for the B410P isn't
available in
Noojeeclick?
http://www.noojee.com.au/Page/NoojeeClick
ADM? (asterisk desktop manager?)
PaulH
Alan Lord (News) wrote:
Dean Collins wrote:
ADA Forums: http://forums.digium.com/index.php?c=8
ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe
ADA Administrators Guide:
Can I assume that you want this only for blind transfers?
I have done this previously, but I lost my copy of the work (and it was
a proof of concept only)
It involved the ${BLINDTRANSFER} variable, which catches the number that
made the blind transfer and making macro-stdexten (or your
Have a look at 'call files' on voip-info.org
Great fun, especially for load testing.
PaulH
Joseph L. Casale wrote:
Any way to initiate a call and execute a playback of an audio file from the
cli?
My only chance to debug or make changes is usually when no one's at the
office including
I tried to get some of that stuff going a while ago, and it just didn't
work - like the polycom user status stuff (at lunch) Asterisk sees the
bits of info but doesn't want to handle it.
PaulH
lord_f...@iinet.net.au wrote:
hi all,
i have 2 x-lite version 3.0 softphones configured on
Please read this book:
http://downloads.oreilly.com/books/9780596510480.pdf
PaulH
Chuck Coleman wrote:
Call from '6000' to extension 'xx' rejected because extension
not found.
Check features.conf - all the codes are set in that file.
PaulH
Catalin S. wrote:
Hello ppl,
I have a problem with my asterisk when i want to call some destination
through my peers and I must enter DTMF digits to select some
extension/conference number or password to access some
I should get all my non-human friends to call this number...or do you
want to call non-humans?
PaulH
Edwin Quijada wrote:
NVLineDetect , I dont find it in the web for asterisk 1.4
Anybody has a link that works?
*---*
*-Edwin Quijada
If you are being paid to work on an Asterisk system, you are in over your
head. You are defrauding your boss and most likely will give him and
everyone in the company a bad impression of Asterisk.
Continuing to answer your questions will only continue to enable you.
Please take a step
Ignacio wrote:
Hello,
I have some zaptel cards, and I would like to install them in some
user's computers. Is there any way to connect those cards with
asterisk server (which is in another computer)?
All manuals I have read explain how to connect asterisk and zaptel
cards in the same
Why do you need so many Asterisk installs?
With the ability of Asterisk to handle hundreds of lines/phones/etc, the
need for several Asterisk server is generally for very specific situations.
PaulH
Ignacio wrote:
Jeff I will take a more depth look at those linksys devices this
weekend but
The old classic is to say something like ' your callerid is blocked,
please get out your credit card'
PaulH
Alfred Monticello wrote:
I'm thinking of starting a partyline, where people call in and talk to
other people. For record keeping and billing purposes, I'd like to go
by the callers
I have had to install a TDM800 in a site, as the telco has held off
installing ISDN indefinitely..
It's all fine except for the fact that it takes ages to hang up the line
(6 or more rings), and sometimes doesn't even bother. This is only on
incoming calls - outgoing calls work perfectly.
Is
I don't think scary is a strong enough wordterrifying? horrifying?
abominable?
PaulH
Steve Totaro wrote:
Your carrier is running Trixbox? That is scary.
Anyways, they are obviously routing calls to the wrong machine. If
your side worked properly before and now does not, then they
This whole thread is getting stupid and I'd hope the people involved would
desist from this O/T drivel.
If you want a switch go to the shop, hand over some money and buy one... Like
every one else does and they're perfectly happy with their purchase.
The O.P. is not going to change the
My memory of a HP procurve (a 2626 PWR from memory) was that it was
quite noisy - have they changed?
PaulH
OCG Technical Support wrote:
Check out the HP ProCurve Switch 2610-24-PWR
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Quick solution that comes into mind:
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if (${DIALSTATUS}=BUSY) {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h = {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if (${call_to}!=) {
Yes, the more expensive ones do. The majority do not.
Linksys phones.
I have a Snom 320 and an Aastra 480i on my desk, and one of the reasons
I love them (especially the Aastra) is the BLF features.
Its not so much knowing if the user is busy or not, its the ability to
be
There are no good Mexican restaurants near my house.
PaulH
Jose P. Espinal wrote:
And your problem is... ?
Bayardo Sanchez wrote:
i have a problem need help
== Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on
'SIP/8022-b7225740'
-- Got SIP response 503
Brian J. Murrell wrote:
Slightly OT, but I'm wondering if anyone here has come across a soft
ATA. That is, software that will perform the functions of a basic POTS
line ATA on Linux with a zaptel driven card.
I have a Linux machine with a zaptel card in it and I want to have
another
Not sure if this is still valid - I used it on a project quite a while ago:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
PaulH
Ken D'Ambrosio wrote:
Hi, all. I want to execute a script, and return the value of said
(Python) script to the dialplan. I thought something like
Google T38 - it's a big subject.
PaulH
amir...@namche.com wrote:
Hi all,
Can we configure sip based outgoing fax on asterisk or we must need zap
channel attached with it?
Thanx in Advance.
Amir Shrestha
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I have used the xorcom usb units for fax a few times, and they work
pretty well.
PaulH
Gregory Malsack wrote:
Hello All,
I have a need to connect an analog device to an asterisk server. The
analog device has 4 analog lines going into it (it’s a fax solution).
The fax solution answers
This has worked fine for me (as far as I know). Is there some flaw I
am not seeing? I see a lot of small businesses that require a 9 to
dial out, even though they don't have very many extensions. Couldn't
they do what I did and not have to dial 9?
Many older systems _cannot_ process
Andrew Joakimsen wrote:
On Wed, Dec 31, 2008 at 22:09, Paul Hales pdha...@optusnet.com.au wrote:
Karl Fife wrote:
Allison Smith just created a hysterical parody music on hold Parody.
Whatever you were doing, stop, and dial this number to listen to it:
360-519-5689. 2 minutes.
I
Karl Fife wrote:
Allison Smith just created a hysterical parody music on hold Parody.
Whatever you were doing, stop, and dial this number to listen to it:
360-519-5689. 2 minutes.
I just gave her a few ideas, but she took it and ran with it--she
chose the audio and did the mix-down and
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
I am strongly against anyone posting anything with their bottom.
later,
PaulH
This may be an obvious thing, but you didn't mention checking whether
or not the card was still seated in the slot properly after the move.
I know from experience that when you move offices, even if you take
all the precautions possible, a card can get bumped just enough to
jostle the
Langdon Stevenson wrote:
Paul Hales wrote:
It looked like the card was still there - from memory the lspci command
said it was.
PaulH
That is correct, lspci shows the card is there. I have also tried
moving the card to a different slot to be sure.
Langdon
So - the current state
Langdon Stevenson wrote:
Yes, that is the current state of play and yes, it looks like I will
have to build from source.
I haven't done this before and am pretty busy at the moment, so it
will take me a while. I will post back when I have done so.
Thanks for the input (to all who have
Have you tried loading the zaptel driver for your card manually?
PaulH
Langdon Stevenson wrote:
Hi
I have a Dell PE2300 with a Digium TDM400P line card in it (with one
module to handle an inbound phone line). This is running on a Fedora 8
system with Asterisk 1.4.21.2-1.fc8
This
.
Is there anything else that I should be doing?
Regards,
Langdon
Paul Hales wrote:
Have you tried loading the zaptel driver for your card manually?
PaulH
Langdon Stevenson wrote:
Hi
I have a Dell PE2300 with a Digium TDM400P line card in it (with one
module to handle an inbound phone line
There are a few web-based ones - is that an option at all?
PaulH
Danny Nicholas wrote:
This sounds like a job for a VB.NET programmer. The program would run
like a DDE server and ftp a call file to your asterisk server on the
desired action.
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