H I guess from a troubleshooting standpoint to try and pinpoint the
problem what I would do is remove all cards from the system and then only
replace the cards that are absolutely necessary like your SCSI card and your
Video card and of course the T100P and then check /proc/interrupts to
On Tue, 31 Aug 2004, Benjamin Johnson wrote:
I found the same with lots of headsets and my 7940, but I've just
plugged the headset from my Norstar system into the *handset* port on my
and it works perfectly. It's not ideal but it'll do for now!
Ah, yeah, didn't think of that - works fine.
On
On Tue, 2004-08-31 at 15:58, Pliva, Josef wrote:
Unfortunately, I am seeing great many missed IRQs continually...if in fact
it is that which causes the loss of D-channel.
Then you need to find out why interrupts are being locked for long
enough to make the T100P miss interrupts. Common causes:
Kris Boutilier [EMAIL PROTECTED] wrote:
Inquiring (management) minds want to know. I'm assuming it's because 'it's
funny how simple it really is to write a really decent voicemail system'?
Perhaps it was written by someone with a red nose, oversized shoes and
a custard pie. I don't know
I've wondered that myself... obviously the writer has a sense of humor! :)
I like the sound of Digium Mail, it sounds cool...
- Original Message -
From: Kevin Walsh [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, August 31,
suppose I have agents waiting on a queue and I configure asterisk to dial
out and to forward the call to the first agent enqueued. Asterisk will do
it even if the answer to the call is busy.
Is it possible to configure asterisk to detect the busy signal and, in
that case, dial another
hmm Meridian Voice Mail == Comedian Voice Mail:)
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To UNSUBSCRIBE or update options visit:
On Tue, 31 Aug 2004 14:14:40 -0400, Jon Bebeau [EMAIL PROTECTED] wrote:
Hello all,
I'm working an a switchboard console for Asterisk and would like to investigate using
IAX Client library to Asterisk. I don't seem to be able to find the source. I'm
planning on a Win32 app. Guidance on where
Chris Shaw [EMAIL PROTECTED] lazily top-posted:
I've wondered that myself... obviously the writer has a sense of humor! :)
I like the sound of Digium Mail, it sounds cool...
I like the sound of, err, nothing.
Mine just prompts for Mailbox?
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/
On Tuesday 31 August 2004 17:36, Kevin Walsh wrote:
Spam-dialling should be made illegal. I, for one, wouldn't spend two
seconds adding features to support this sort of usage.
I can think of at least one legitimate use for this -- reverse spam dialling,
or at least real person detection. I
Lol reverse hold!
I can't see that working ever though, I tried it once and the agent at the
other end hung up on me... I had to wait another hour in the queue...
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 2:53 PM
I had always thought it was because an early clone of 'meridian
mail' was called 'chameleon mail' and 'comedian mail' is a really good
take off on 'chameleon mail'.
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On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
Paul,
What you can do is modify the source code for the voicemail application.
Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
'vm-password' to 'pls-enter-vm-password'.
Recompile and install.
Then in your
On Tuesday 31 August 2004 23:22, Umar Sear wrote:
On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
Paul,
What you can do is modify the source code for the voicemail application.
Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
'vm-password' to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Nobody knows about that strange behaviour of Zap channels or
at least if is that right?
Thanks in advance.
Rodrigo P. Telles wrote:
| Hi,
|
| I'm using a TDM400 with one FXS and one FXO module (developer kit) and
| I've been testing termination
Hi all,
I have next configuration:
SIP Provider---ADSL router---localnet
192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones
first localnet 192.168.20.0
second localnet 172.28.240.0
in second localnet we have
The vm-password file is used else where, such as queues. If you changed
it, then you would change for all the other applications.
I guess, since we alter the source code often.. it's not that big of a
deal. We just create our own patch files and if we update from cvs, we
patch against the new
I have Asterisk RC2 setup here with a Fritz ISDN card on Debian Woody.
I'm using chan_capi-0.3.5 and fcpci-suse8.2-03.11.02. Settings are
pretty much the default, except in order to get the fcpci module to
compile, I had to follow the instructions here:
hi;
anyone can recommend a good TTS for the dutch language compat in
linux?
--
Best regards,
Danny mailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures division - internet solutions
place princesse elisabeth 9/11 - 1030 Brussels - Belgium
Tel :
Does anybody knows if it's posible or if there is some develoment in
course to be able to use longer transmit packet sizes (as long as I know
this is fixed in 20ms now) with the compressed voip codecs in asterisk
(g729, g726, gsm, etc).
I need to use asterisk to connect remote sip clients with
The standard for loop start does not send answer supervision, so * and all
other telcom devices that do CDR records have to 'assume' that the call was
answered.
Lyle
- Original Message -
From: Rodrigo P. Telles [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Hi all,
Has anyone gotten custom ring tones to work using ALERT_INFO with the
Cisco 7940 SIP phone? I've read the wiki, but just can't get this to
work. I'm currently using the 7.2 SIP image.
Thanks,
Chris
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Hi all!
I downloaded right mpg123, chabged path to mpg123 binary in
app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But
MP3Player refuses to do properly:
-- Accepting AUTHENTICATED call from x.x.x.x, requested format =
1024, actual format = 1024
-- Executing Answer(IAX2/[EMAIL
Luis Vazquez [EMAIL PROTECTED] wrote:
Does anybody knows if it's posible or if there is some develoment in
course to be able to use longer transmit packet sizes (as long as I know
this is fixed in 20ms now) with the compressed voip codecs in asterisk
(g729, g726, gsm, etc). I need to use
On Tue, 31 Aug 2004 15:34:51 +0200, Axel Eble [EMAIL PROTECTED] wrote:
On Tue, 31 Aug 2004 15:22:26 +0200, Michael Labuschke
[EMAIL PROTECTED] wrote:
Pick up mobile phone.. enter sms .. send it to the * phone number.
Done
On the * side.. follow the sms howto (voip-info.org might have
Maxim-
This will not work through a FWD DID as you suggest. BT requires each
telephone number to be registered in order to receive SMS messages. You
need a either an analogue, BRI, or PRI line that terminates in your asterisk
box directly. The way a line gets registered is that you must
I set up my own STUN server and turned reinvite
off.
Lyle
- Original Message -
From:
[EMAIL PROTECTED]
To: '[EMAIL PROTECTED]'
Sent: Tuesday, August 31, 2004 8:53
AM
Subject: [Asterisk-Users] SIP
registration with public dynamic ip address
Hi, I'm trying
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Marlowe
Sent: Monday, August 30, 2004 12:55 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration
I just got a Polycom soundpoint and I set it up
try steven sokol's iaxphone and see if you have the same problems dialing
his box while taking * out of the equation. same problem=network, no
problem = *
http://www.sokol-associates.com/IaxPhoneDownload.htm
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users
Correct. TDM (time division multiplex) FXO is for analog ports coming from
the telco.
- Original Message -
From: Marcello Lupo [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 1:52 PM
Subject: [Asterisk-Users] Analog lines and TDM card
Hi,
sorry to bother you,
I need to know how to setup the data side of the T1 on my Linux Box. I
have found information about configuring a PRI and HDLC but nothing
about the Frame-Relay type setup for data.
The following is information from our T1 provider.
Network T1:
Framing = ESF
Line code = B8ZS
On Aug 31, 2004, at 8:42 AM, Steve Underwood wrote:
Chris Shaw wrote:
- Channel Support:
IAX2 in asterisk
IAX2 in libiax2
Other IP channels in asterisk (RTP-based ones, I guess are all that
is
left).
CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a
complete
solution...
Tobias Jönsson wrote:
Sorry, I did not know these american specialities. I just noticed in
Larry's PRI debug info that he received a STATUS message during the
waiting, so I thought that the waiting could lead to some kind of
timeout at the telco. In EuroISDN the callerid always come in first
Just out of curiosity,
What version of CVS and Polycom SIP software are you running happily?
Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0?
I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with
poor results. Transferring did not work as expected. Using the # key to
do
i'm new here and i need help on how where can i get
software version 4.0.x of the mediatrix and how can i
install it...
mediatrix unit im using has a software version of
2.4.9.57. i would like to use H.323 not SIP...
please need help asap!... hope to hear from anyone of
you soon..
thanks in
Chris Jensen wrote:
I am hooking up to a DMS500 (100250 together) and wanted to see if
anyone had any experience with this. We have the GR-303 span up, the
IDT is built.
I have not yet heard of anyone doing this, but would be _extremely_
interested in your experiences. Please keep in touch with
Look up the word persist in the XML config file...
- Brent
On Tue, 31 Aug 2004, Reid A. Forrest wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Marlowe
Sent: Monday, August 30, 2004 12:55 PM
To: [EMAIL PROTECTED]
On Tue, 31 Aug 2004 15:58:16 -0500, B. J. Bomar [EMAIL PROTECTED] wrote:
I use a Plantronics Supra H51 plugged straight into the headset port, and it
works great.
B. J.
Same here.
They're wonderful headsets.
-Shaun
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A customer of mine has 3 TDM400P cards in a box running asterisk. On
each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an outgoing call.
Channels 1-12 are in group 1.
If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when
he dials out, it
Hi,
I'm trying to dial in from one phone and give it access to another
line (ie incoming on zap/1 and outgoing on zap/2)... how can I transfer
the call from channel 1 and give it the dial tone on channel 2? I can
use dial but that takes a phone number, which I want the user to be able
to
On 1 Sep 2004 at 17:15, [EMAIL PROTECTED] wrote:
A customer of mine has 3 TDM400P cards in a box running asterisk. On
each card he has four FXO modules.
I have set up the dialplan to dial via group 1 for an outgoing call.
Channels 1-12 are in group 1.
If he plugs a telephone cable
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