cdr_addon_mysql doesn't compile at all, no matter what the OS. I
contacted the authors a while back and they said they would get into
contact with Mark or something... Who knows what happened, but as far
as I know it's still broken.
I tried to compile addons on an x86_64 Opteron under FC3.
Scott Nelson wrote:
I am an Asterisk newby, and I cannot seem to get Caller ID information
from our T1 line. When calls appear at the phones, they say the call
came from asterisk and unknown number.
I know how Caller ID information is passed on an analog phone line
(between the rings) but
I'm staring at an RFP--this company wants to replace a 2000 position PBX
(at eight locations) with a new system. Their mindset is Nortel/Avaya
because they talk about 28-button digital sets. The do specify a few IP
phones for just one location, so they are aware of VoIP.
I'm going to bid on
You could also get an old, cheap computer off eBay put it between the
switch(es) and the dsl modem, install linux and then use it to do your
QoS prioritization. Not very elegant or professional looking, but it
would work if you don't care about such niceties.
You can buy 400 series servers from
Steve Underwood wrote:
Hi,
People often send me audio logs from spandsp's soft-fax machine, where
they have problems with corruption in the middle of a page for most or
all of their faxes. Their problems are usually due to frame slips.
However, recently I have received audio logs from two
Steven Critchfield wrote:
On Thu, 2005-03-31 at 00:32 -0500, Tim Bass wrote:
Hello All,
This is my first post. Sorry to post under such sad circumstances. Here is
the situation:
We installed a TE410P (today) in a SuperMicro 1U server today (Motherboard
X5DE8-GG), which was running great until
snacktime wrote:
We have 10 incoming POTS lines to our offices, and a nortel norstar
pbx. I've been looking at replacing it with * at some point in the
future, and the point that looks most cost effective is when we move
to PRI.
Problem is, I'm not really sure how to go about getting a good deal,
I have a SNOM 220 with a 20-button sidecar.
The configuration for the five lines (buttons) on the main phone is
straigth forward: display name, account, password, registrar.
I would like to get each of the sidecar buttons to register with
Asterisk in Line mode so that I can have incoming calls
then
please chime in.
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
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. Steve Underwood says 3.6.1 has problems.
Check Steve's FAQ at opencall.org for additional gotchas.
Mike
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
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the only thing available.
At that time there was _zero_ Linux representation in the computer
stores. If it weren't for Linus and RedHat, I'd be a VB programmer
right now. There is a certain amount of loyalty, you know...
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
enjoying the New Year.
Cheers,
Mike
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
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reports (with a
TDM22B installed):
VCORE 1.676V
DDR Vtt 1.344
+3.3V 3.28V
+5V 4.945
+12V12.544
5VSB4.945
The other card is also a TDM22B, and he DOA card is a TDM40B.
I've rotated all cards throught my test system with varying degrees of
flakines.
Cheers,
--
Michael Welter
protocol only? I've seen the
SMS functions in Asterisk--how are these intended to be used? Is it
Europe/GSM only?
Cheers
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
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to be present.
Am I using the correct syntax?
Thanks,
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Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
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: 2936506117 2936506698 ERR: 0
MIS: 0
http://www.microsoft.com/whdc/system/sysperf/apic.mspx
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
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, and the address book
information is then used to compose the cover sheet.
Cheers,
--
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Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
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Altus Snyman wrote:
How do I fax a .tiff file with asterisk?
Use Steve Underwood's spandsp library and TxFax function in Asterisk.
See http://opencall.org
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Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
it yourself. If you show a little effort on your part then
maybe, just maybe, someone on this list might help you.
Mike
--
Michael Welter
Introspect Telephony Corp.
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. Then use cdrecord to write
each file to a CD-ROM. You can then install FC3 as normal.
Mike
--
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Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
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Asterisk
device and take those five lines (KERNEL=...) and stick
them in the file /etc/udev/rules.d/50-udev.rules and then reboot.
Mike
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
initiative (and a little less sarcasm) then
the members of this list would be more inclined to help you.
Mike
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
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.
Mike
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Andy Burns wrote:
Michael Welter wrote:
In the zaptel directory, find the file README.udev. Find the #
Section for zaptel device and take those five lines (KERNEL=...) and
stick them in the file /etc/udev/rules.d/50-udev.rules and then reboot.
Thanks for the reply, but as I mentioned those
devices? I don't recall
having any problems with SATA drives.
--
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Introspect Telephony Corp.
Denver, Colorado US
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http
Nils Segerdahl wrote:
On Fri, 7 Jan 2005, Ryan wrote:
I had the same problems using hfc cards with bristuff. (with patched
zaptel drivers).
Which zaptel patches did you use?
Thanks
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
me they had
a good TIFF library, and later found they also have one or more bad ones
too. :-)
Have you tried libtiff-3.7.1 yet?
Mike
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
you please explain what you did?
Thanks,
--
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Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
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? I
have a Canon fax that receives complete pages, but never the entire
transmission.
Thanks,
BTW, when the Canon disconnects, the Asterisk line remains off-hook.
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
You absolutely do need to worry about usb module.
http://www.microsoft.com/whdc/system/sysperf/apic.mspx
Warren Burstein wrote:
I'm not having any trouble with interrupts, but here's my
/proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the
SMP kernel (2.6.5-1.138). I don't
Warren Burstein wrote:
Michael Welter wrote that I should be worried about the usb module.
Would rmmod uhci_hcd be enough, or should I disable it in the BIOS
like Shoval said?
Also, after the rmmod, I still have the conflict with libata on 169
CPU0 CPU1
0:7311006
Matt Riddell wrote:
Steve Underwood wrote:
Matthew Boehm wrote:
I know myself, SS7 will be a make or break for our continued use of
Asterisk.
Our make/break is FoIP support. If Asterisk had some form of T.38 for
reliable fax transmission..or even just T38 pass-thru..
One down, one to go.
Michael Loftis wrote:
--On Monday, January 17, 2005 22:20 -0800 [EMAIL PROTECTED] wrote:
...
The basic arrangement would be:
Telco - T1 - Asterisk - T1 - Channel Bank - POTS - regular
phone
Have you thought about how the asterisk system will send CDR to the
property management system? How the
I've been having trouble getting into voicemail from a 7940 phone.
Turns-out that the phone was intermittently sending two of every DTMF
digit. I replaced the phone and everything is fine.
Could this be dirty contacts? Has anyone else seen this? Is there a
way to repair the phone?
Thanks
From an AGI script I'm setting two channel variables and then calling
Voicemail on several mailboxes. I know the channel variables are set
correctly because I can GET VARIABLE from within the script.
In voicemail.conf (in mailsubject and emailbody) I reference the channel
variables, but the
Michael Welter wrote:
From an AGI script I'm setting two channel variables and then calling
Voicemail on several mailboxes. I know the channel variables are set
correctly because I can GET VARIABLE from within the script.
In voicemail.conf (in mailsubject and emailbody) I reference
Carlos Chavez wrote:
I want to buy a new server to run Asterisk and after looking at prices
for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed
rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64?
Yes, Zaptel and Asterisk compile and run
Carlos Chavez wrote:
On Mon, 24 Jan 2005 15:06:27 -0500, Martin Roy wrote
I'm using a server with dual AMD opteron processors with a TDM04B
without any problem. The server is running Fedora Core 3 AMD 64bits.
Hope this answer your question...
Yes, thank you very much. I had no doubt that
Since we're chatting about tftp servers...
Let's say I have a new customer with Cisco 79xx phones, and he desires
to SIP register on my Asterisk system. I would have to provide the
SIPmac.cnf and SIPDefault.cnf files on my tftp server for his phones.
These files would be world readable, which
Howard Lowndes wrote:
On Thu, 2005-01-27 at 03:34, Michael Welter wrote:
Since we're chatting about tftp servers...
Let's say I have a new customer with Cisco 79xx phones, and he desires
to SIP register on my Asterisk system. I would have to provide the
SIPmac.cnf and SIPDefault.cnf files on my
Matthew Crocker wrote:
Yes, that's why I'm posting to this list. I certainly don't want to
have to track my customer's IP addresses. Is there a better way?
Have them go into manual mode and enter your SIP proxy information
directly into the phone, or, have them run their own tftp server and
If a board design is that sensitive to touching, it
certainly implies a design problem.
Twenty-plus years of doing electronic repair/diagnostic work says
that is no where near normal. Very very very sensitive to the injection
of electrical noise.
I'm going to do a bunch more testing this
Not to mention the CPU spikes every n seconds. Rich, while you're
testing, would you keep an eye on 'vmstst 1' and the 'system' (not user)
CPU utilization?
That cpu spiking is another issue separate from the stability issue
(I think). Not sure where the discussion of the spiking ended up a
Ty Carter wrote:
Just as an FYI I have a configuration suppestion to pass along...
If you have:
Fedora Core 3
Asterisk 1.0.x
Digium Wildcard T100P PCI Card
In order to get the card to work you MUST edit
/etc/udev/rules.d/50-udev.rules
Goto the end of the file
Eric Bishop wrote:
Can any give me or point me to a short and simple explanation of what HDLC is?
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Brian M. Arlinghaus wrote:
Martin,
In the 7960s (my firmware is P0S3-07-3-00), you can press settings and
then #9 to unlock the phone. From the settings menu, #3 will take you
into Network Configuration. Near the bottom (#25), you must disable
DHCP and Save. Then, from within the Network
Michael Graves wrote:
OK, I asked this about a week back and met with no repsonse at all. But
perhaps its worth trying again.
Does anyone on-list have * running BRI to their local telco? I'm
considering this as an alternative to my TDM400p card.
I had an HFC card and a BRI circuit from Qwest,
Adrian Chapman wrote:
Changing the order of things in extensions.conf around a smidge got it
all working nicely :-
[inbound-from-pstn]
include = default
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,Playback(thank-you-for-calling-please-wait-a-moment)
exten = fax,1,Macro(faxreceive)
exten =
[EMAIL PROTECTED] wrote:
I think you might be missing the point here. SER is a raw SIP processor.
So for a second throw everything you know about Asterisk + SIP out the
window and go back to vanilla SIP. Getting used to a B2BUA in the call
path kinda beats some of the raw power of SIP up. Think
David J Carter wrote:
How do you want Switch to appear to Asterisk.
1. As an extension. Then use an FXS connection to a CO line input.
The extension interface at the PBX will be supplying battery and dial
tone. Therefore, you would want to use the FXO (red) daughter board on
your TDM400P card.
Steve Underwood wrote:
This seems to be a problem with the current wctdm driver. It seems to be
broken for audio going out. I used to be able to send faxes reliably
using spandsp and a TDM40P card, but I no longer can. I haven't had time
to look in detail at what is wrong.
And I think the CPU
John Middleton wrote:
Has anyone any experience of the above.
Key feature for me is tracking incoming and outgoing emails and
linking them to the contact record.
Thanks, sorry for the OT ;-)
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Remco Barende wrote:
Hi list!
I have some sip phones and Sipura ATA 2000's. However after dialling a
number I need to dial a # to control a device.
When I dial # Asterisk kicks in and puts the call on hold. How can I
change this?
Do you have the T in your Dial statment? Remove the T and try it.
FYI, I didn't read your message. With hundreds of messages/day, I use
the subject line to decide whether or not to read. Whenever I get a
message with (no subject) it is an instant delete.
Also, for those of you who think you're still on a 300baud modem and
have to conserve every keystroke,
Rich Adamson wrote:
BTW, spandsp-pre9 is the most current.
Rich, spandsp-0.0.2pre10 has been on Steve Underwood's site for some
time. There are also test app_rxfax and app_txfax modules available.
I have two Canon fax machines that consistently fail on the third page.
From the traces, I get a
zimdog wrote:
I am wondering what has been done as far as running multiple instances
of *. Like if you were providing a virtual PBX for small companies. I
imagine it can be done in one of the following ways and am looking to
see how other users are doing it. I have searched the wiki and have
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Hello Keith,
My name is Michael Welter, and I have been installing Asterisk systems
for two years. You may call me on 303-718-2804.
Mike
Chris Blake wrote:
Greetings *`s,
I have a Digium TDM01B card which I want to connect to a standard phone
socket on the wall, for the purposes of testing [EMAIL PROTECTED]
On the 4 pin connector going to the wall socket, I have the wires from a
CAT5 cable inserted as follows :
Brown/White -
Peter Svensson wrote:
On Tue, 22 Feb 2005, Daniel Nyström wrote:
A little off-topic maybe, but it's still for the Adit used with Asterisk. ;)
I wonder where I can buy 50 pin Amphenol cables, with connector on one side,
and open cables on the other for mounting in our own patch panels.
In Europe,
VoIP Services wrote:
Some country codes are three digits long. Some are two.
e.g. UK 44 , Bermuda 441
And some country codes are one digit, like 1 for US
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I see a lot of chatter in the archives about intercom and paging, but
has anyone addressed zone paging? Each of the 50 telephones in a large
clinic would be members of one or more paging zones. Someone could then
page Dr. X in zone #1. Would this be possible with analog phones? SIP?
My client would like to add new functionality, such as call queuing, to
their PBX. They presently have a Telrad PBX; it has a T1/PRI circuit
from the CLEC and a few POTS lines. They use DID for some incoming
calls and an attendant for calls on the main number.
The client's commitment to
Does asterisk/libpri recognize the NI2 protocol for calledid name?
Thanks,
Mike
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I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to
* are perfect. However, when calling from * to Diax, I get the following:
channel.c:1097 ast_read: Dropping incompatible voice frame on
IAX2[mike]/3 of format GSM since our native format has changed to ULAW
In iax.conf
Why doesn't someone just ask Markster how many are on the list?
Inquiring minds want to know...
Chris Albertson wrote:
Maybe the words nine million was not ment to be taken
literally. What if he said about a gazillion Then we'd
all be arguing if gazillion == 1x10^14 or 1x10^16
Have you ever
My CLEC just called and asked if we will support the MI2 protocol on
our proposed T1 circuit. I think this is for CallerID name. Will the
T100P support this?
Thanks,
Mike
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Sorry, it's the NI2 protocol. A previous poster said the N2
protocol was supported by the T400P. Is NI2 the same as N2? Does
NI2 mean switchtype=national?
Thanks
Michael Welter wrote:
My CLEC just called and asked if we will support the MI2 protocol on
our proposed T1 circuit. I think
And check the firmware revision in your Adtran. I believe current is L36.
Alfred R. Nurnberger wrote:
The only thing I can think of in respect to analog DID lines is answer
supervision.
DID lines provide one way - outbound audio - before answer and cut through
bidirectional audio only after
My T1 PRI installs soon. It has eight voice and the rest data. Does
the PRI d-chan share one of the data channels or is it separate? Will I
get 16 data channels or 15?
Thanks,
Mike
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I have 1.0.4.45 (beta) on my tftp server. Try it at 66.250.23.58.
Cheers,
Michael Welter
Rick Smith wrote:
I'm getting 1.0.4.30 I think it is, in new phones, but all that's on the
website is 1.0.3.81
Where do you download newer versions ?
And, will anyone else's firmware work
Did you say you were using Adtran FXS cards?
Bisker, Scott (7805) wrote:
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 EM wink lines from telco. I have them all plugging into an Adtran 750
Shouldn't they be FXO cards for CO lines?
Bisker, Scott (7805) wrote:
Yes. Adtran FXS cards.
Did you say you were using Adtran FXS cards?
Bisker, Scott (7805) wrote:
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now
While the rest of you were chatting about the smallest * server, I was
sitting her staring at the telephone hanging on the wall.
It is a Western Electric set in a varnished pine box with an earpiece
you hold in your left hand and a mouthpiece attached to the box. You
crank the magneto with
I have a Gigabyte K7 motherboard with an Athlon 2400+ processor. Before
the T1 install I had two T100P cards, one for the channel bank and the
other unused. This ran perfect for a month.
Last week we installed a new integrated T1 into the unused T100P (to
replace POTS lines and DSL.) In
system Freeze, Nothing at all
workings, except the reset button.
You setup is vastly different from mine to.
Dual Pentium III SMP, X100P Dual TDM400P
What type and version of Linux?
Mine is RH9 2.4.20-8???
Would love to track this one down...
--- Michael Welter [EMAIL PROTECTED] wrote:
I
the new T1 circuit. Each T100P has a
separate IRQ.
Any feedback would be appreciated.
Mike
Michael Welter wrote:
I have a Gigabyte K7 motherboard with an Athlon 2400+ processor. Before
the T1 install I had two T100P cards, one for the channel bank and the
other unused. This ran perfect
have further information?
Thanks,
Michael Welter
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Yes, I tried Wait(1) but still no joy.
Steven Critchfield wrote:
On Wed, 2004-02-11 at 12:13, Michael Welter wrote:
I have a new T1 PRI circuit from Eschelon. They're sending the caller
name in the facility record.
Is it possible for * to capture this information? I remember an old
post
Yes, it is in the CDR. I'll put PRI in debug and try to determine just
when the facility record arrives.
Thanks,
Mike
Steven Critchfield wrote:
On Wed, 2004-02-11 at 12:25, Michael Welter wrote:
Yes, I tried Wait(1) but still no joy.
Is it in your CDRs?
Steven Critchfield wrote:
On Wed
Side question: should all of us on RH9 do the LD_ASSUME_KERNEL=2.4.1 ?
TC wrote:
-do you use hyperthreading
-do you use the LD_ASSUME_KERNEL=2.4.1 b4 loading asterisk
-have you compiled zaptel with the SMP flag on
Can anybody site some real hardcore technical facts
about SMP hyperthreading
On my Eschelon T1, all I get are the last four digits.
Steven Critchfield wrote:
From your zapata.conf file below, I see you have configured for a PRI.
PRI by default is treated like a DID. You MUST define a extension entry
for every incoming call. If you had looked at the console error messages
I presently have a T1 with eight voice channels and four data channels.
Channels 1-4 are data, 16-23 are voice, and 24 is the dchan.
The vendor plugs the T1 into a Vina Integrator 300 which splits the
data out to a LAN jack. This device is only capable of a half duplex
LAN connection which
I bought mine off of eBay.
Each cabinet should contain a PSU board (power) and a BCU (control)
board. Then you have six slots for FXO/FXS cards. You'll also need the
power supply (which mounts on the side.) Some units have a cabinet
containing four 12V batteries.
There are also two slots
a relay clicking with each ring in the paging controller.
Does anyone have experience with configuring these devices for paging?
Thank you,
Michael Welter
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(the channel
number?), but then I am able to page. Does anyone know how to get rid
of the DTMF tones?
Thanks,
Michael Welter
Michael Welter wrote:
I've a Valcon V-2001A paging controller connected to an Adtran 750 FXS
port. The V-2001A looks like an FXS loop start extension.
When I call
/mailman/listinfo/asterisk-users
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+1 303 674 2575
[EMAIL PROTECTED]
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got busy signal. Anyone know what would
be the dialplan (extension.conf) to accept T1 line calls.
Thanks.
-Tri.
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Introspect Consulting, Inc.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
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The freeze-ups were due to a NetGear NIC card. Haven't had a freeze
since I removed that card.
Mike
Steve wrote:
On Monday 09 February 2004 11:45 am, Michael Welter wrote:
I have a Gigabyte K7 motherboard with an Athlon 2400+ processor.
Before the T1 install I had two T100P cards, one
Has anyone backended a Fujitsu 9600 with an asterisk system? Does
anyone know anything about Fujitsu's em link signaling interface (T1)?
Mike
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To
for guests?
Wow, what a market this could be!
Bill Michaelson wrote:
Anybody know how to implement a hotel wake-up call feature with *?
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Michael Welter
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Woops, should have said symbolic link :-)
cd /usr/src
ln -s linux-2.4.25 linux-2.4
Cheers
Michael Welter wrote:
I think the zaptel compile depends on a softlink 'linux-2.4' pointing to
'linux-2.4.25' (or whatever). Your compiles may be including .h files
from an older source tree.
Mike
A few days ago the 7960 phones were delivered. Today the received the
power adapters. However, we've seen nothing about the SIP licenses
(which were bundled into the price.)
Does anyone have a tftp site that I can use to download the firmware. I
would like to use this site until Lewan Assoc.
Does anyone know if version 3.1 is Call Manager or SIP? Thanks.
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address. At the login
screen type the password admin. You will then get the config page.
For the tftp server you can use mine. It has the .50 firmware, and the
IP is 66.250.23.58. This works for my GS.
Once you set the config parameters correctly the phone should be ok.
Cheers,
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Michael
. Is there
something else I should be doing?
Thanks,
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Introspect Consulting, Inc.
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+1 303 674 2575
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P.S. linux-2.4.25 on Debian
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Michael Welter
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