Hi,
I have set-up following set-up.
The sip clients is connected to the asterisk and will also be registrar in the
asterisk.
The asterisk is register like a client to our sip server with same user names that the
clients have.
When I tried to call on of the sip-clients, the asterisk answer the
I'm doing a tutorial at Astricon and the plan is to use my laptop as a
demo server. Today it failed to boot and after a bit of sleuthing it
turns out the fan is sticking from time to time on bootup. Apparently
there is a sensor and if no spin, no go.
Moving everything right now would be
I've spent the afternoon recording all the files for the English speaking
VM etc. I've parked the file here http://www.g7ltt.com/VoIP/vmukmale.tgz
I did it with Audacity at 44.1KHz x 16bit and thenused sox to raise the
levels to -3db and then again to down sample them into 8KHz GSM files. The
few
[EMAIL PROTECTED] wrote:
I have a DSP based system that is working on a four port FXS system
using a 200MHz arm processor.
Well.. since we are talking about this topic I owe you guys notes of my
experience
with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch
etc.).
We
On Sun, Sep 19, 2004 at 02:53:52AM -0500, Brian Capouch wrote:
Moving everything right now would be draconian, yet I can't take a
chance on a no-boot while at the show. Does anyone know how bad a job
it is to dig into one of these units and clean up/oil/whatever the fan?
Not off hand but
Hi,
I'm currently thinking of putting more functionalities to Linux server box.
Major is Asterisk, but would also like to add video surveillance, home
automation and limited (for only domestic up to 4 users) web, file and
mailserver apps.
I know there are problems running Asterisk with other
I'm currently thinking of putting more functionalities to Linux server box.
Major is Asterisk, but would also like to add video surveillance, home
automation and limited (for only domestic up to 4 users) web, file and
mailserver apps.
I know there are problems running Asterisk with other
Hi All,
I hope I'm posting this to the appriopriate list, and that cross posting
to two lists is OK. (If not, I'm sure I'll hear about it quickly :))
I'm running Asterisk on my (new) VIA EPIA-V motherboard.
This seems to be the ideal platform for a home version of asterisk - its
small, quiet, low
Graham Turner [EMAIL PROTECTED] wrote:
dear all, i am looking to enable CALLERID on an Asterisk system
comprising a X101P FXO interface connecting to BT PSTN in the uk
seems this is supported by the interface but there seems to be varying
information on how to enable it in zapata.conf
1.
Message: 12
Date: Fri, 17 Sep 2004 20:35:32 -0400
From: Rollo Tomnasi [EMAIL PROTECTED]
Subject: [Asterisk-Users] Zaptel compile error - unresolved symbols
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII
Hello - any help is greatly
Erm, didn't think of that. Stupidly I deleted the individual wav files.
Not a problem though as I have the 3 master files that I recorded them all
into. I'll just have to slice it up again. That'll be a few days as I've
got family arriving today.
Mark
Linus Surguy said:
I've spent the
On Fri, Sep 17, 2004 at 11:09:49AM -0500, Paul Traue, Jr. wrote:
I've just installed asterisk as a new phone system for our office but am
having difficulty with the queues. Specifically I need a way to
redirect our sales queue to voicemail when no one is logged in to the
queue. I see I
HI,
I have the latest RC2 of Asterisk on a RH 9 non-modified-load box. I have
an Avaya IP phone that uses h323, so I am trying to compile h323 into
Asterisk. Now, I downloaded pwlib and openh323 tar files and I have
compiled this according to the instructions:
pwlib:
./configure
make opt
Why is wsusb loading? The X101P uses the wcfxo module.
Lyle
- Original Message -
From: Andy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 7:29 AM
Subject: [Asterisk-Users] X100p on VIA EPIA-V problems
Hi All,
I hope I'm posting this to
Kevin, thanks for post reply .
i have installed asterisk / zaptel from cvs distribution as of 17/09/04 so
i assume this does it
have configured zapata.conf as per instruction but i would have expected to
have seen the callerid on the asterisk console as it receives the call but
then may be not
I paused myself when I saw this.
The generic /etc/init.d/zaptel (that you get if you do make config)
tries to load
wct4xxp, wct1xxp, wcfxo, wcfxs, and wcusb
Paring down the list to just wcfxo generates exactly the same problems.
Cheers,
Andy.
Message: 12
Date: Sun, 19 Sep 2004
Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99
retail. They have a version with a router for $89.99. We picked the
non-router version up and it seems to be a rebadged Sipura SPA-2000. The
box has a Vonage service package inside as well, but it does work with other
Hello, I am hitting a brick wall with ASTCC.
If someone can help me get a working Calling Card
system up and running and give me some explainations on how they did it I will
trade them a Grandstream 486 or a BT102 phone in white, your choice (brand
new)
I want someone who has done it
?
- Original Message -
From: Carlos Arnt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 5:14 AM
Subject: [Asterisk-Users] Dial 0 to outbound
Hi Folks.
I see that can put 0 to call out using a x101p (zaptel) or even a pstn
service.
Thats great, but when
I wouldn't trust it to do any real detection. I use the press # mod in
6 sec mod to be able to fwd to other phone #s without risking hitting
the answering machine or wrong person. I don't believe there is any
real way to detect what you are after as far as if the call is picked
up. You would get
If your phone is on a Zap, MGCP, or SCCP interface then look at the
ignorepat option in extensions.conf. If your phone is SIP or H323
then this is handled by the phone. Most SIP and H323 phones do not
allow you to continue dialtone after dialing a digit.
On Sun, 2004-09-19 at 11:28, Steve
ignorepat = 9 ; Continue dialtone after dialing 9
if i am reading your question correctly
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 12:28 PM
Subject: Re:
I just posted this app, which I've been testing for a while and should be
ready for inclusion in CVS.
http://bugs.digium.com/bug_view_page.php?bug_id=0002467
This app was put together so as to be able to deal with answering machines
when making outbound calls. The idea is that you probably
OL Hi all,
OL I am trying to install asterisk on my system, the compiplation and
OL installation process all seem to work fine (make ; make install ; make
OL samples).
OL But astersik fails to start. Is the sample configs not supposed to
OL work out of the box?
OL Even more confusing, it seems to
Matt Riddell:
This is really unlikely but is it possible he has an internal
firewall or something and the Asterisk box is in the DMZ?
Sadly not. That's the only thing I could reasonably expect to be causing the
problems, but they're on the same LAN in the same network and there's no
firewall
I'm running Asterisk on my (new) VIA EPIA-V motherboard.
This seems to be the ideal platform for a home version of asterisk - its
small, quiet, low power, and should have plenty of computing horsepower
if only it would work!
I'm running Redhat 9, Kernel 2.4.20-8, and Asterisk from
Does anyone know where to disable rtc support on redhat 9.0 using make
menuconfig?
I thought I disabled it but still got the following error when trying to
make zaprtc:
zaprtc.c:109: storage size of `rtc_irq_timer' isn't known
zaprtc.c:719: storage size of `rtc_fops' isn't known
zaprtc.c:107:
On 19/09/2004 16:12 Senad Jordanovic said the following:
Well.. since we are talking about this topic I owe you guys notes of my
experience
with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch
etc.).
i'd be eagerly awaiting these results. i've tested a 16MB image of
Please explain how you got the PAP2 to work with another
carrier? I spent over an hour on the phone with 3 levels of
Linksys support staff and 2 levels of Vonage staff telling me
that the PAP2 CAN NOT be used on any other service except
vonage because they burn the vonage information
I had 2 senior level management people at linksys corp confirm that this
would not be possible until December. They both told me that they are
currently in development of a 'non-locked' version but that it would not be
in stores until December.
Did you find these PAP2-NA at Fry's as well? Online
It was my understanding that you don't 'disable' rtc, but recompile it as a
kernel module.
Again, just my understanding as I can't try it until monday.
Matthew
- Original Message -
From: Chad Brown [EMAIL PROTECTED]
To: Michael Bielicki [EMAIL PROTECTED]; Asterisk Users Mailing List -
Matthew Boehm wrote:
I had 2 senior level management people at linksys corp confirm that this
would not be possible until December. They both told me that they are
currently in development of a 'non-locked' version but that it would not be
in stores until December.
Did you find these PAP2-NA at
Any help would be appreciated as I am a novice trying to work around a
difficult situation.
This is what the zaprtc helpfile says:
zaprtc, getting zaptel timing out of your realtime clock
Make sure that you _dont_ have rtc support
On Sun, 2004-09-19 at 11:58 -0700, Chad Brown wrote:
Any help would be appreciated as I am a novice trying to work around a
difficult situation.
This is what the zaprtc helpfile says:
zaprtc, getting zaptel timing out of your realtime clock
I had 2 senior level management people at linksys corp
confirm that this would not be possible until December. They
both told me that they are currently in development of a
'non-locked' version but that it would not be in stores until
December.
Did you find these PAP2-NA at Fry's as
I have had no success with a stock redhat install.
The removal of rtc went fine, but I cannot compile the zaprtc.
I posted the compile output a while back, also having NO rtc in my kernel
and being an smp system, is this a problem ?
-Original Message-
From: [EMAIL PROTECTED]
Does anyone have one of these models? Can they confirm that it works with
any other SIP server? How is the PAP2-NA configured? Web? Phone?
The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2.
Thanks,
Matthew
- Original Message -
From: Marty Mastera [EMAIL
Matthew Boehm wrote:
I had 2 senior level management people at linksys corp confirm that this
would not be possible until December. They both told me that they are
currently in development of a 'non-locked' version but that it would not be
in stores until December.
Those kind of people only know
Hi,
http://www.costcentral.com/searchresult.php?keyword=PAP2searchin=1
Mfg Part # Stock Price
-- -- --
PAP2 No $49.86
PAP2-NA Yes $49.76
Best regards,
Miroslavmailto:[EMAIL PROTECTED]
Sunday,
Does anyone in New Zealand have any ATA devices in
stock I.e. Sipura SPA-2000?
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Jeremy McNamara wrote:
Matthew Boehm wrote:
I had 2 senior level management people at linksys corp confirm that this
would not be possible until December. They both told me that they are
currently in development of a 'non-locked' version but that it would
not be
in stores until December.
Those
Chad Brown wrote:
Does anyone know where to disable rtc support on redhat 9.0 using make
menuconfig?
I thought I disabled it but still got the following error when trying to
make zaprtc:
zaprtc.c:109: storage size of `rtc_irq_timer' isn't known
zaprtc.c:719: storage size of `rtc_fops' isn't known
On Saturday 18 September 2004 01:09, Matt Hohman wrote:
Google? Hey thanks for the info I haven't seen that before. Wonders of
modern technology. It's nice to use the list as a round table and get
some insight.
While I tend to agree with you the AB1's been discussed to death. Google
really
Date: Sun, 19 Sep 2004 12:59:52 -0600
From: Rich Adamson [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: [Asterisk-Dev] X100p on VIA EPIA-V
problems
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
I'm running Asterisk on my (new) VIA
Hi guys
I've got a need to do some call queueing, with the slightly unusual caveat
that the destination for the calls is not a phone or group of phones
connected to my local asterisk box, but an external PSTN number.
Can I setup a queue in asterisk and make the queue member an external
address
Kristian,
I have 2 X100P cards but neither work on my Compaq DL360 G2. The system
will not even boot! Take a look at my initial post and let me know if
you have any other advice. Regardless, thanks for your post!
-
I need
What you want to do is connect the remote phone number to an internal
extension.
You can do this in a couple of ways, using the Manager interface and the
Connect command. Alternatively, you can create a call file in
Asterisk's call spool (usually /var/spool/asterisk or whatever) which
has the
I'm running Asterisk on my (new) VIA EPIA-V motherboard.
This seems to be the ideal platform for a home version of asterisk - its
small, quiet, low power, and should have plenty of computing horsepower
if only it would work!
I'm running Redhat 9, Kernel 2.4.20-8, and Asterisk
Marconi Rivello [EMAIL PROTECTED] wrote:
Hi, I have a curiosity: how much does a regular PBX system cost? I'm
curious if using IP telephony in a building is cheaper than a regular
PBX, because of the high cost of the IP phones.
Take a look at
On Sun, 19 Sep 2004 07:29:16 -0500, Andy [EMAIL PROTECTED] wrote:
Everything works perfectly, except for the following problem:
Sporadically -- about once in 6 hrs, Asterisk reports a Red Alarm from
the X100p. Thereafter, the X100p no longer works -- no outgoing calls
can be placed; no
Hi everyone, Im a newbie to Asterisk. Will Asterisk
run on RH9, easily or does it have to run on FreeBSD? Will the drivers for
the Digium cards work on RH9?
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[EMAIL PROTECTED]
Hi
I am getting more hands on about Asterisk issues but I got a question to ask.
What is the common factor, to put all configurations bind to MySQL or
have them as they are originally on text configuration files.
Maybe this questions can be out of focus, but it will clear up some
ideas in the
I am working on a proper MySQL iaxfriends now just getting ready to post on
bug site
Tested and works great
It loads all your iaxfriends into registry when you do a reload... So if you
add new users to MySQL you have to do a reload
-Original Message-
From: [EMAIL PROTECTED]
Depends. Do you have daily (sometimes hourly) configuration changes to your
*.conf files? I do. Therefor for me its better for the conf files to be
stored in database. I've even rewritten some of the * code to pull
information out dynamically instead of having to reload each time.
If most of your
On Sun, 19 Sep 2004 15:10:27 -0700 (PDT), Nick Bachmann
[EMAIL PROTECTED] wrote:
Take a look at
http://www.buyerzone.com/telecom_equipment/phone_systems/buyers_guide7.html.
As the article pointed out, TCO is important as well. Commercial PBXs
usually require technicians with special software
Maybe is not right to ask, but those changes are GPL to the community?
If yes, any README to integrate them, or detailed explanation
If not, no problem, is nice to know that those changes exist
Regards
John Fach
On Sun, 19 Sep 2004 18:06:56 -0500, Matthew Boehm [EMAIL PROTECTED] wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=0002469
New Patch for MySql
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Linux
Dominicana
Sent: Sunday, September 19, 2004 7:19 PM
To: Matthew Boehm
Cc: Asterisk Users Mailing List - Non-Commercial
Yes. Mine are GPL to the community. I have already posted 1 such patch to
the list. I will post my others once I believe they are stable.
Matthew
- Original Message -
From: Linux Dominicana [EMAIL PROTECTED]
To: Matthew Boehm [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List -
Have you also posted them at bugs.digium.com ?
Thanks
Duane Cox
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: Linux Dominicana [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 6:44 PM
Subject:
Rich Adamson wrote:
My understanding from the 'expert' is the PCI issues have something
to do with a poor PCI chip design on the motherboard. The folks that
are heavy into audio apps tend to swap out their VIA motherboards.
Guess that implies there aren't any workarounds.
From the Ardour (Linux
Hi,
I don't know quite how to ask this question, because my knowledge is so
limited at this time. I have an h323 phone that I am trying to use to do
VOIP to phones on the PSTN. I want to sign up for a service and not have it
go out my POTS line. I do have a Quicknet Line jack in my RH 9 box
On 19 Sep 2004 at 18:19, Nick Barnes wrote:
Thanks for your comments and suggestions though - keep them coming,
there's sure to be something I missed.
I think I may just recompile * and try it all from scratch again.
Yeah. I'd delete the source and regrab it from CVS. Don't worry
about
Hi,
Would anyone know of a way to set the time automatically on an ADSI
capable phone from *?
The phone in question is a Aastra 480e.
While I am at it, does anyone have any helpful docs on the ADSI script
programming? I have managed to do basic functions by modifying the
asterisk.adsi file
HI all,
I would like to build a 12 CO by 36 phone system with
Voicemail, I am trying to decide which machine would be a cost effective
solution. Would a Pentium 4 2.6 GHZ with 1G of Ram be suitable?
___
Asterisk-Users mailing list
some conferencing systems want you to hit octothorpe (aka pound, hash,
etc.). once connected, i would have expected * to be transparent to
all dtmf codes. it seems not to be. wiki has not been helpful, it
seems to have most references to do with octothorpe in dial plan.
so, what do i not
Rich Adamson wrote:
My understanding from the 'expert' is the PCI issues have something
to do with a poor PCI chip design on the motherboard. The folks that
are heavy into audio apps tend to swap out their VIA motherboards.
Guess that implies there aren't any workarounds.
From the
I will post my others once I believe they are stable.
Matthew
- Original Message -
From: Duane Cox [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 7:15 PM
Subject: Re: [Asterisk-Users] Configuration on
I have the following system setup:
ECS KT600-A motherboard
Athlon XP 2.6 FSB 333
2x512 MB 333 DDR memory
TDM400P (3 FXS, 1 FXO)
X101P (Encore MD 3200) modem
nVidia MX 4000
Works all right, I have been restarting asterisk about once a week, due to
other issues. Here's the lspci:
:00:00.0
easily
- Original Message -
From:
Henry Devito
To: [EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 6:30
PM
Subject: [Asterisk-Users] Asterisk and
Red Hat 9
Hi everyone, Im a newbie to
Asterisk. Will Asterisk run on RH9, easily or does it have
Chad Brown wrote:
Kristian,
I have 2 X100P cards but neither work on my Compaq DL360 G2. The system
will not even boot! Take a look at my initial post and let me know if
you have any other advice. Regardless, thanks for your post!
Oh that was you? I read that post earlier but obviously didn't put
Henry Devito wrote:
Hi everyone, Im a newbie to Asterisk. Will Asterisk run on RH9,
easily or does it have to run on FreeBSD? Will the drivers for the
Digium cards work on RH9?
Last I heard RedHat was the development platform for *. It doesn't have
to run on FreeBSD, and in my experience
I believe the problem being refered to is described below, which affects
VIA chipsets made between 1997-2002. The VT8233 used in the Mini-ITX is
on the 'hit list'. I understand there was a patch to the linux kernel
which was supposed to have been merged into kernel 2.4.25 to address
this
Quoting Matthew Boehm [EMAIL PROTECTED]:
Does anyone have one of these models? Can they confirm that it works with
any other SIP server? How is the PAP2-NA configured? Web? Phone?
The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2.
The product manager for this devices
OK, I've posted the orignal WAV files in 44.1KHZ x 16bit mono format here
http://g7ltt.dyndns.org:8010/VoIP/vmukmale-wav.tgz (26MB!)
Mark
Mark Phillips said:
Erm, didn't think of that. Stupidly I deleted the individual wav files.
Not a problem though as I have the 3 master files that I
On 19 Sep 2004 at 17:56, Randy Bush wrote:
some conferencing systems want you to hit octothorpe (aka pound, hash,
etc.). once connected, i would have expected * to be transparent to
all dtmf codes. it seems not to be. wiki has not been helpful, it
seems to have most references to do with
Hi everyone, I'm a newbie to Asterisk. Will Asterisk run on RH9, easily
or
does it have to run on FreeBSD? Will the drivers for the Digium cards
work
on RH9?
I am running Asterisk on RH9. I did not encounter any problems with both
Asterisk and the drivers for the Digium card.
I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im
wondering if this ATA supports auto-provisioning.
Matthew Boehm wrote:
Does anyone have one of these models? Can they confirm that it works with
any other SIP server? How is the PAP2-NA configured? Web? Phone?
The pdf I
we run Asterisk on RedHat 9 with no problems. Works
great!
Paul
Paul
Mahler [EMAIL PROTECTED]
Signate, LLC665 Third
StreetSuite 100San Francisco,
CA94107-1901Asterisk Services and
Training
From: [EMAIL PROTECTED]
Yes, I tried adding an adaptec that I thought was uhci. This didn't
work. I didn't check the specs but figured that they would be uhci given
the history of ohci.
Regardless, I will confirm with adaptec.
Thanks,
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Antonio Rabena wrote:
I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im
wondering if this ATA supports auto-provisioning.
Can you confirm if under Advanced Settings there is a Provisioning
Tab, and under it there is a space for a Profile Rule?
--
Andres
Network Admin
Yes there is.. the default setting is /init.cfg. Not sure about these
parameters.. I cant find this provisioning setting on the user-guide.
maybe anyone can help?
Regards,
Antonio Rabena
Andres wrote:
Antonio Rabena wrote:
I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web.
I ran the LiveCD version of Asterisk on my hardware and it worked. I am
trying to run it natively on a 2.6 kernel (Gentoo distro), but it keeps
getting a seg fault using the sample configuration files. Does Asterisk
not work with the 2.6.8 kernel?
TIA
Chuck Wegrzyn
So here is the big question.. just how similar is the hardware in the
Linksys PAP2-NA to the Sipura 2000? It would be interesting to see if a
PAP2-NA will take the Sipura 2000 firmware That would be great,
since I'm guessing there will be quite a bit of lag time between the
Sipura
Would anyone know of a way to set the time automatically on an ADSI
capable phone from *?
just call the phone its part of the adv callerid fsk tones it will set it as
soon as ypou call the adsi phone
i have some 350's and 480's and the the time is set
The phone in question is a Aastra 480e.
From your experience, could you give us the merits and demerits of
these ATA devices as well as the IAXy.
They are essentially a Sipura SPA-2000. One of my customers uses the Sipura
exclusively for his customers and they work very well. Setup is easy, and
they support the CLASS type features
I am trying to obtain optimum gain settings for a bank of analog lines
connected to a channel bank. My telco has provided a 'Type 102' test line to
use for incoming level calibration. This is functionally equivalent to app
Milliwatt(), but provides tone from the CO inwards.
Question is, how
Hi there,
I am selling my 7960, 7905g and tdm400p with one fxs module.
E-mail me for more info. Just for info the phones both have 3 year
service contracts (smart-net) on them.
[EMAIL PROTECTED]
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