version of ALSA 1.3c fixed all my
problems. It now works wonderfully with gnophone even. Now I can use
it under Linux, not just Windows anymore, which makes me very happy.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version
I can't find this in the archives, pardon me if it has already been
hashed out. I have recently learned of Asterisk and are trying to get
my hands around the scope.
On our University campus we have all of our users in a LDAP directory.
It would be great if we could interface with this store
, since there's
currently no description of an IVR-only system. (I assume that below
the 60-channel mark you'd still call it successful.)
Thanks, Jason
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is 2.4G Xeon's, Fedora 1, and
TE410P's. E400P's and TE410P's were about the same in performance in
my load testing.
Out of curiosity, what was the least powerful system that yielded the
same maximum call rate?
Thanks,
Jason
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hank smith wrote:
hello I am just curious if there is any windows alternitives to Asterisk?
can I also use them with free world dialup?
thanks
hank
I've never used either of these and I'm certainly no authority on the
subject, but here are a couple Windows based alternatives I've come
across
. That way no matter what there IP is, asterisk will
see that they have *IP* registered in the SIP peers and dial that to that IP.
Or am I just doing something wrong in the configuration?
Thanks in advance,
Jason
winmail.dat
Hi all
Does anyone know where I can get hold of the German 1TR6 ISDN signalling
protocol specification.
Thanks
Jason
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Hey Steve
Apparently so :-(. It is used in our legacy PBX with which I would like to
connect my Asterisk box.
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Underwood
Sent: 03 March 2004 03:56 PM
To: [EMAIL PROTECTED
, but you may want to add it to the bug tracker.
Jason
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On Mon, 1 Mar 2004 10:02:50 -0500
Glenn Dalgliesh [EMAIL PROTECTED] wrote:
I seem to be having trouble with cvs login. anyone having similar
problems. It just hangs after entering the password
Happened to me last night. Worked after repeated attempts.
an ethernet port to hook up to the phone
Or am I just dreaming up a new product to market?
Jason
winmail.dat
compiles just fine on BSD, if you are using 4.x-RELEASE, and
not using chan_h323, chan_oss, zaptel libpri.
Correct. I'm using it with many SIP-based VoIP connections now, plus the
one IAX2 connection as mentioned above.
--
Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn
instructions on using CAPI or other ISDN based cards) with Asterisk.
Thanks alot,
Jason
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the first ring.
Does anyone have a config they could share with me on how to make this
setup work? This sounds like it should be fairly trivial, but I've
beaten my head against the wall on this for a few days. =)
Thanks alot,
Jason
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include the demo. In a production system, you
; probably don't want to have the demo there.
;
include = local
exten = 431,1,Dial,SIP/sipphone
Regovich, Timothy wrote:
Jason,
Include your sip and extensions files so people can take a look.
T
-Original Message-
From: [EMAIL PROTECTED
Have a look here.
http://www.ams.net/products/product_info.cfm?Product_ID=10891
JR
On Fri, Feb 20, 2004 at 06:44:45AM -0700, Chris Hirsch said:
Heison Chak wrote:
$8/yr from Cisco.
Seriously? Wow from Cisco thats actually reasonable! ehhe
Yeah, can I get a part number for that
I recently ordered a few phones from them for some testing I'm doing, but
the charges still haven't appeared on my credit card nor have I received
any confirmation email (not sure if I'll get one though). Does anyone know
if they're severely backlogged for orders?
--
Jason T. Nelson [EMAIL
something and dont know it or just uninformed on
what this takes? All the information I have read says all I need is a Digium T100P
card terminating my T1 and a ethernet card for my local network for my VOIP phones, is
this correct?
*Setup*
Redhat 9.0
Digium T100P
Ethernet Card
Thanks,
Jason
Hi Stan,
Thanks for the info, it helps me out loads.
JR
On Wed, Feb 11, 2004 at 04:34:46PM -, Jason Ross wrote:
This is slightly off topic so sorry for the intrusion.
I've got a couple of 7940 phones I'd like to put on Smartnet but I'm
looking for what I need to order, what it roughly
.
Thanks,
Jason
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the network device to behave correctly, we haven't had a
single problem.
--Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, February 09, 2004 10:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP phones
Hi Paul,
I believe it is...
and this in meetme.conf
[rooms]
conf = 18
conf = 18,1234 ; 1234 is the PIN.
HTH,
Jason
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Steve Kennedy wrote:
Probably a dumb question, but what's the best Linux variant to use to
build/run an Asterisk server.
Hardware is Compaq DL360 with a Widcard 410.
Debian/Fedora Core ?
Steve
As others have said: Whatever you're most comfortable with. Having said
that though, I'm partial
Hi Geert,
Jason Ross wrote:
G'Day,
I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm
having DMTF problems no matter what configuration I try. And as yet I
haven't downgraded it to see if an earlier release makes a difference
Just wondering if anyone can provide some
, for awhile. I'm not sure what
triggers this behvaior. Anyone else getting this behavior?
I wish the lists were searchable... :(
Thanks.
Cheers
Jason
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a:
restart gracefully
triggered the behavior.
Thanks for the quick reply.
Cheers
Jason
Mark Spencer wrote:
In the mean time try running asterisk with no console. This is bug #864.
Preliminary analysis shows that after a restart now, one of the
ioctl()'s performed by editline fails with -1
of code for this phone may be.
TIA,
Jason
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need to do.
TIA,
Jason
And the errors are...
bash-2.05b# patch -C openbsd.patch
Hmm... Looks like a unified diff to me...
The text leading up to this was:
--
|Index: Makefile
|===
|+++ Makefile 5 Feb 2004 13
Joel,
Compared to iaxtel or FWD, there is a significantly higher amount of
latency, but it is workable.
You will see and hear more latency because the service is based out of
Australia. If you run a ping to the host you'll see nice and long round
trip times.
HTH,
JR
Yes, Nortel Meridian's can get 5 9's easily. They are very expensive, but we have one running at a government site in Indiana that has been up for 15 years without interruption. When you upgrade the 1 control unit, the other 1 is servicing all the requests. There is a brief period of time when
Attached is a simple patch for the format_wav_gsm.c file. It should
fix the size tags in the gsm WAV headers, allowing picky software like
WMP to play them without complaining. Also uploaded to the bugtracker,
id 254.
format_wav_gsm.c.diff
Description: Binary data
else noticed this?
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alexandru Coseru
Sent: 20 January 2004 09:17 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SS7 over Asterisk ?
Now I understood that I can't use Asterisk
[Sorry if this gets posted twice -- I sent it with the wrong account and
it's stuck in moderator review...]
I downloaded the files from the bug tracker and had a look at them. The
original msg.WAV is slightly malformed: it's chunk tags are too big.
A lot of audio programs ignore this because
to listen to the line (as Steve suggested
in an earlier post) and will post my findings.
Good luck
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Reinhard Max
Sent: 14 January 2004 10:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
on other
platforms.
I have already started playing with trying to figure out why Asterisk runs
so badly under FreeBSD, such as eating 100% of the CPU without warning
plus the decidedly non-standard directory structure (as far as FreeBSD's
hier(7) cares about).
--
Jason T. Nelson [EMAIL PROTECTED
to Asterisk, just so I can spend my development time trying to make
FreeBSD work.
No, judging from the other messages I've read in this thread, it looks like
there are plenty of other people willing to make that effort. :)
--
Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx
have no idea where to start.
Would someone (Steve Underwood ;-) )mind at least putting me on the right
track so I can address this issue?
Thanks in advance Steve
Jason
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the archives
and perceived what appears to be a slightly hostile attitude towards those
who ask about Asterisk support of other free operating systems even without
using Digium hardware. Is this Linux-specific bias intentional or accidental?
--
Jason T. Nelson [EMAIL PROTECTED
software DSP
code and I therefore can't see a reason for the limitation to zaptel.
Thanks anyway
Jason
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: 13 January 2004 06:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Fax
On Tuesday 13 January 2004 03:42
of
systems. I wasn't aware that an idea could be patented or even an
implementation, but basically, from what I saw they patented a flowchart
(!!).
Basically, check into the relevant patents for this kind of system.
Just giving you a heads up.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc
I've been trying to get a console channel working without success.
The sound card, which is built into the motherboard, is a VIA
Technologies, Inc. VT82C686 AC97 Audio Controller.
Using the oss drivers (vi82cxxx_audio) in kernel 2.4.23 and chan_oss, I
just get beeps and screeches.
Using alsa
are feeding back into the
wrap-around mic? I have a USB Plantronics DSP 400 headset running under
ALSA sound system. Is it feeding through the plastic parts and entering
the microphone sitting out near my mouth?
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com
I would love to try it Thanks very
much
Jason
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Azher Amin
Sent: 02 December 2003 07:03 AM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
PREPAID APPLECATION
Hi Bart,
I would like to test
is in
the correct location and all dependencies appear to be satisfied. I've
also had a search in the archive and I can't find anything relating to
this problem so I was wondering if anyone had an idea as to what may be
causing this issue.
Thanks,
Jason
Brian West wrote:
Ya learn to search the archives. This has been covered MANY MANY times.
And I still haven't gotten the echo to go away completely. I usually
just end up making it worse. :)
bkw
On Sun, 23 Nov 2003, VoIP Fan wrote:
Hello:
I have installed *. I configured my SIP
just have a voicemail system, and not a very
useable one at that.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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set to 3.0. It appears that one
only needs to shutdown and restart asterisk. At least, I've heard
different effects when I do this without unloading and reloading the
wcfxo and zaptel modules.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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Jason A. Pattie wrote:
| I have this exact same problem as well. We have a scenario in which we
| are not using any analog extensions, just SIP and IAX software based
| phones (DIAX, X-Lite, gnophone, (trying to use) linphone, etc.) with a
| single
?
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[EMAIL PROTECTED]
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iD8DBQE/uqlYuYsUrHkpYtARAr0qAJsGa4gRcMu+RaoauDorqeDS0ZOsQQCbBWat
6YEb5X8Ovi+3SFYsb/jEUwE=
=owz8
Anton L. Kapela wrote:
Jeremy McNamara said:
Flebay it and use the proceeds to support Digium by purchasing Zaptel
hardware. http://www.digium.com/
I like the zaptel cards, I really do. The price per port is better
than other fxo/fxs's in almost all cases and the support I've received
. They sucked up an rRAM disk image off the
| net at boot time.
Was this by chance using LTSP? (www.ltsp.org)
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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Asterisk is an H.323 gateway - if you want it to be ;-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: 06 November 2003 07:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] H323 Gateway
Hi all,
Anyone know of a small H323
then configured asterisk to listen on the one IP and ser to
listen on the other. In fact I even have an H.323 gatekeeper running on
the same box as well and everything is looking good.
Hope that helps - give me a shout if you need any more help
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED
no
replies. The previous posts however did not contain too much information
about the sequence of events leading up to the error. Hopeefully this
post will genereate a solution!!!???
I wonder if someone knows what is going on here?
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto
period. Then there is a period of silence for less than 2 seconds, and
then the cycle is repeated.
| I don't have an answer for you but at least it may stop others falling
| into the same problem if somthing can be identified as the cause..
|
| later..
Thanks.
- --
Jason A. Pattie
[EMAIL PROTECTED
as well as GSM. They all do the same. ulaw seems to
work better.
I have exactly this same problem as well. It's even worse when running
X-Lite under Wine under Linux.
- --
Jason A. Pattie
[EMAIL PROTECTED]
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Asterisk doesnt answer the ringing phone, so according to some doc that I
read I should try and see if I get I/O on /dev/zap/1. I dont see any from
cat /dev/zap/1 while it is ringing or otherwise. Dmesg seems to have
a few odd things (out of space), not sure how relevant. Does anyone have
The gazel 128 PCI cards are great - never had any problems. They also
offer a USB one that is supported by ISDN4Linux. You can get these
products from www.bewan.com
HTH
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Marc V.
Liotier
for your sip proxies.
Good luck
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Albertson
Sent: 14 October 2003 07:21 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] */SER/FW
I think the way to run SER and * on the same box
up in /proc/zaptel).
Incoming calls are routed to the zap/x channel, but no ring.
I'm hoping I'm overlooking something stupid.
Thanks ahead of time...
--Jason
Here are some (possibly) relevant snippits from various places:
o T100 LED shows green...
o Not showing any errors in /var/log/asterisk
is originating from so if anyone has any
helpful insight or feedback they could provide on this matter it would be
greatly appreciated.
Some DMS100 working configs would be great too.
Jason Helmich
MIS, Blue Sky Communications
PGP Key ID: 0x4CF71E92
[EMAIL PROTECTED]
011.684.258.1077
too! (using Wine). There are a few gotchas that you have to
work around, but other than that, it works perfectly fine. See my
previous posts about this topic.
Also, I think that should be http://www.xten.com.
- --
Jason A. Pattie
[EMAIL PROTECTED]
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with the TDM400P card or if they are ADSI
compatible at all? I kind of doubt they will work if they are not
compatible, but I don't know what it would take to plug them directly
into a * box.
Thanks.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE
, and then it works perfectly.
Now I have a decent cross-platform SIP softphone client. :)
(now if only my laptop could record sound, I could really use it)
- --
Jason A. Pattie
[EMAIL PROTECTED]
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to * properly. Don't know what's going on there. linphone
does have the ability to inject DTMF digits and seems to work properly
from that respect.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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. But, why buy a
headset for the softphone? You can do echo cancellation, right?
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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Jason A. Pattie wrote:
| Mark Spencer wrote:
| | Remember the S100U has to generate high voltage (-48V when idle, plus
| | about 70v a/c on top of that when ringing), and it does it all with 5V.
| | It does get a little warm. That's normal.
|
| Wow
it's just a
kernel or driver issue.
Thanks.
NOTE: Dialing into the X101P on either system worked flawlessly. I
don't think there is anything wrong with the X101P.
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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Jason A. Pattie wrote:
| It's even worse on the usb-ohci machine. Most of the time, the S100U
| doesn't even give me dialtone.
Sorry, forgot to mention that the S100U device is detected and setup
correctly on this machine and no error messages
..
|
| My recommendation is to get a new one ad see if it solves your problems..
|
| Later..
|
|
|-BEGIN PGP SIGNED MESSAGE-
|Hash: SHA1
|
|Jason A. Pattie wrote:
|| It's even worse on the usb-ohci machine. Most of the time, the S100U
|| doesn't even give me dialtone.
|
|Sorry, forgot
it
running now for around 2 - 3 hours and it is still quite cool to the
touch. I will install the wcusb driver and see if the unit heats up.
Maybe all these issues that people have with these devices are heat related?
- --
Jason A. Pattie
[EMAIL PROTECTED]
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-BEGIN PGP SIGNED MESSAGE-
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Jason A. Pattie wrote:
| I have another observation for you. Do your S100U's get warm? I've
| left this one plugged into the USB port and the red LED is lit on the
| unit even though the wcusb driver is not loaded at this time. I noticed
:
|
| -BEGIN PGP SIGNED MESSAGE-
| Hash: SHA1
|
| Jason A. Pattie wrote:
| | I have another observation for you. Do your S100U's get warm? I've
| | left this one plugged into the USB port and the red LED is lit on the
| | unit even though the wcusb driver is not loaded at this time. I
| noticed
gotta put a drain on a USB
system. Imagine if you wanted to load an entire 127 extension phone
bank off one USB port (daisy chained USB hubs) ... Supposedly it's
theoretically possible to connect a maximum of 127 USB devices to a USB
port (or something like that).
- --
Jason A. Pattie
[EMAIL
that?
The SIP code is the same for the two phones, as far as I can see the
only difference is the 4 extra lines you can configure. I've got a
couple of 7940 phones and one SNOM 100, I think I'll be sticking with
the Cisco product as it seems to be more reliable and generally just
works better.
HTH,
Jason
learn how the site:lists.digium.com facility
works. If you search this archive you will find links to other useful
sources of information.
Just a general request to the maintainer of asterisk.org could you add
a links page to useful * sites? It may help reduce these types of
requests.
Jason
duplicate extensions between the two PBXs?
Eg: 555-x100 and 555-x100 on same * server
2. Is there a way to configure follow-me-calling to have an extension
attempt to ring a series of numbers, eventually dropping to
voicemail?
Regards,
Jason Smith | Tier-1 Co-location
1301 - 1377 of 1377 matches
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