Christopher Chan wrote:
C F wrote:
I think yes, why you disagree?
Has Microsoft actually ever come with such useful features?
It would be great to demonstrate the complete instability/insecurity of
Windows based servers by have it shut down automatically in front the
boss with a recorded
On 19 Mar 2007, at 20:51, Scott Plante wrote:
Raw Hangup
The code says:
/* A call arrived for a nonexistent destination.
Unless it's an inval
frame, reply with an inval */
If you can, produce a packet dump of a failing call with ethereal - or
with iax2
On 20 Mar 2007, at 03:14, Leif Neland wrote:
Steve Totaro wrote:
Stephen Bosch wrote:
Olivier wrote:
I'm really after 1U-2U silent servers as I've got the feeling most
of them are too noisy for offices and most of our clients don't
have server rooms.
Try this:
dear all,
we need help for integration asterisk (sip) with mera
we have configure for sip.conf and extentions.conf
sip.conf
[mvts]
context=mvts
type=friend
host=10.10.0.2
dtmf=rfc2833
in extentions.conf
[mvts]
;
; mvts
exten = _01162.,1,SetCallerID(mvts)
exten = _01162.,2,SetCIDName(to mvts)
Hi,
Let's say you have an Asterisk server running.
Which parameters would you check to improve service continuity ?
I was thinking of :
- telco lines status (make sure every is up)
- registered hardphones
- config files backup (compare live and saved configuration files, if files
differ,
and this is the /var/spool/hylafax/log/c1: http://pastebin.ca/403282
cat /var/spool/hylafax/log/c3 :: http://pastebin.ca/403291
thank you
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that's work know i don't know where is the problem but i folowed the
links bellow:
http://www.guardiani.us/index.php/TrixBox_IAXModem_HylaFax#Satisfy_HylaFax_Deps
http://www.ecualug.org/?q=2007/02/27/comos/como_recibir_fax_en_asterisk
http://threebit.net/mail-archive/asterisk-users/msg04703.html
Hi All
I'm tracing a very strange problem which I could reproduce with different
versions up to 1.2.5 (sorry, didn't update to a newer one yet).
Scenario 1: Problem does not occure.
=
Sip Phone registered directly to the Asterisk.
exten = i,1,Zapateller()
exten =
Kevin P. Fleming ha scritto:
OK, then you'll need to get a verbose/debug console trace, and
preferably a packet capture of the IAX2 traffic on 'Server', and post a
bug on bugs.digium.com with those files attached.
___
While setting up the servers to
From: Olivier [EMAIL PROTECTED]
Date: Tue, 20 Mar 2007 09:49:34 +0100
Hi,
Let's say you have an Asterisk server running.
Which parameters would you check to improve service continuity ?
I was thinking of :
- telco lines status (make sure every is up)
- registered hardphones
If you use VoIP,
exten = i,1,Zapateller()
Same happens if I use PlayTones(info) instead of ZapaTeller().
Same happens if I use Progress() before ZapaTeller or Playtones.
Mit freundlichen Grüssen
Benoit Panizzon
--
I m p r o W a r e A G-System Services
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of younss azzayani
Sent: Tuesday, March 20, 2007 3:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Faxs any help :)
and this
I have a PRI switch type national
Asterisk 1.2.16
Zaptel 1.2.15
If I call an invalid number I get
* PROGRESS with cause code 28 received
Asterisk continues to attempt to connect the call until the timeout is
reached and I hear ringing.
I want to capture the progress code,
Hi, i try install FreePbx by tuturial in
http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443
but i have this error when i try install freepbx:
#pear install db
No releases available for package pear.php.net/db
Cannot initialize 'db'
Hi guys,
We are experiencing a problem with a T1 PRI connection. After trying a number
of variations in the configuration files, the behavior is always the same: no B
channels come up and the D channel doesn't appear to be working well. We can
see there are ATT Maintenance messages being
On 3/20/07, Olivier [EMAIL PROTECTED] wrote:
Let's say you have an Asterisk server running.
Which parameters would you check to improve service continuity ?
The tools I tend to use are vmstat, iftop (all VoIP, all the time),
show registry and df.
-HJC
thank you :)
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perhaps you should try
pear install DB
However note that this mailing list has nothing to do with pear.
Hi, i try install FreePbx by tuturial in
http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443
but i have this error when i try
Christopher, welcom to Vista, it's now possible.
On 3/20/07, Christopher Chan [EMAIL PROTECTED] wrote:
C F wrote:
I think yes, why you disagree?
Has Microsoft actually ever come with such useful features?
It would be great to demonstrate the complete instability/insecurity of
Windows based
younss azzayani wrote:
and this is the /var/spool/hylafax/log/c1:
http://pastebin.ca/403282
cat /var/spool/hylafax/log/c3 ::
http://pastebin.ca/403291
What does zttest say? If it's below 99.98% then hardware configuration
is where the problem is.
Lee.
Quoting C F [EMAIL PROTECTED]:
Awesome, the first PABX virus is just around the corner now that M$ has some
bait for it to infect.
In a world without borders we don't need windows or gates.
Christopher, welcom to Vista, it's now possible.
On 3/20/07, Christopher Chan [EMAIL PROTECTED]
d-channel is in midle
bchan=1-15,17-31
dchan=16
loadzone = it
defaultzone = it
Kanelbullar wrote:
Hi guys,
We are experiencing a problem with a T1 PRI connection. After trying a
number of variations in the configuration files, the behavior is
always the same: no B channels come up and
Simone Cittadini wrote:
I post once again here, sorry for the verbosity, if then in your opinion
there's still something wrong with * internals and not with my
understanding of the configs I'll open the bug.
I would encourage you to open the bug anyway; I am currently at a trade
show in
Thanks for your answer, Bruno. However, the configuration you provided is for
an E1 connection and we are using a T1, having channel 23 as D channel.
Bruno De Luca [EMAIL PROTECTED] escreveu: d-channel is in midle
bchan=1-15,17-31
dchan=16
loadzone = it
defaultzone = it
Kanelbullar wrote:
Hi!
Could you please tell me why have you chosen the CentOS instead of any other
Linux distribution?
--
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de shadowym
Enviada em: sábado, 3 de março de 2007 14:17
Para:
hello friends,
I want to install Asterisk on a Debian machine.
I need to download the sources or just with apt-get install is enought???
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Our setup is:
9.6k Modem -analog- Mitel SX-200 -(pri)- Asterisk -(pri) - Telco
The modem works fine with the Mitel directly connected to the Telco, but
once we add Asterisk in between connections start failing.
I suspect the issue is caused by the echo canceller, since I believe the
issue
We are connecting the GXP2000 to a Cisco POE switch 3650, it the
default mode at 15.4 for each phone the phones power up but we can only
have 24 on a 48 port switch, when we adjust the setting to 7000 (which
is what we calculate the phone to use) they don't power up ... we
tried all the power
The Asterisk and Zaptel development teams have released Zaptel version
1.2.16.
In addition to minor bug fixes, this release fixes a build-time problem
on systems where the default language is not English, and also corrects
a regression in the driver for the Digium dual- and quad-span cards with
Kanelbullar wrote:
Hi guys,
We are experiencing a problem with a T1 PRI connection. After trying a
number of variations in the configuration files, the behavior is
always the same: no B channels come up and the D channel doesn't
appear to be working well. We can see there are ATT
On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
hello friends,
I want to install Asterisk on a Debian machine.
I need to download the sources or just with apt-get install is enought???
Depends on the version you want to install. You can install with apt-get
install asterisk, of
On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
I want to install Asterisk on a Debian machine.
I need to download the sources or just with apt-get install is enought???
It depends on what version do you want to use. In sarge is only the
version 1.0.7. In etch is 1.2.13, but the 1.2
Josu Lazkano Lete wrote:
I need to download the sources or just with apt-get install is
enought???
apt-get is the easiest way, but won't give you the latest release.
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Hello, everyone.
I get
Best: 99.975586 -- Worst: 99.975586 -- Average: 99.975586
with zttest
Where do I have to start looking for hardware errors?
Thanks in advance
younss azzayani wrote:
and this is the /var/spool/hylafax/log/c1:
http://pastebin.ca/403282
cat
You could download the source from asterisk.org and follow the install
instructions. You could also use SVN to download the source. Also, there
are a few binary package links found at
http://www.voip-info.org/wiki/index.php?page=Asterisk+Download.
Bobby Crawford
_
From: [EMAIL
On 3/20/07, Mark Farver [EMAIL PROTECTED] wrote:
I suspect the issue is caused by the echo canceller, since I believe the
issue appear about the time we turned echo cancellation on (for the IAX
users). We don't need echo cancellation for PRI to PRI calls. I've
looked around, but I am finding
Hello,
What numbers do I dial to make an analog phone attached to an TDM400P
ring via the asterisk demo after installation and starting asterisks?
Thanks,
Ed
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This problem / messages has not gone away ...
No one's got any ideas or explanation about what the card is trying to tell
me?
--
Chris
Chris Earle [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
bristuff-0.2.0-RC8s
two isdn lines plugged into first two ports
and like I said,
Hi Josu,
I've done it both ways, and they both generally work equally well (so long
as the package maintainers are doing a decent job). As Victor mentioned
though, the version you wish to install plays a factor in this. I found the
Asterisk build in the repos to be a bit out dated.
Also, it's
zfter running zttool i got:
--- Results after 303 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.994914
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dima wrote:
I get
Best: 99.975586 -- Worst: 99.975586 -- Average: 99.975586
with zttest
Where do I have to start looking for hardware errors?
I would start with IRQ sharing. Make sure that your Zap hardware isn't
sharing an IRQ. Secondly, you want to have it with a fairly high
priority,
younss azzayani wrote:
zfter running zttool i got:
--- Results after 303 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.994914
Okay. Then at this point you need to put record in your iaxmodem
config file, restart iaxmodem, attempt the call again, and afterwards
send me
This can be done like this:
;user extensions
exten= 1,1,Dial(SIP/U1,,Tt)
exten= 2,1,Dial(SIP/U2,,Tt)
exten= 3,1,Dial(SIP/U2,,Tt)
;secretary extensions
exten= 4,1,Dial(SIP/Secretary,Tt)
the Tt option in dialplan lets the secretary to transfer the user
;conference extensions
exten=
I have a client application looking for an Asterisk based solution.
Client wants to deliver pre-recorded messages for a variety of clients.
Wondering if anyone is offering an middleware for Asterisk for
management of outbound messaging?
Email me.
Thanks
Cory Andrews
If anyone is at VON I would be curious to know what sort of hardware they
are using.
Quanta Computers-booth 1445
Dlink, Uniden apparently decided not to attend according to the exhibitors
list
From the back of the boxes it looks like ITX but maybe not ix86 CPU.
Nothing to indicate how they
Question:
After about having the server running for about an hour, our callers
occationally hear a high pitched beep that lasts the entire call. In
some cases, the noise doesn't start until a minute or 2 into the call,
while others last the entire call. In some of the more serious cases,
calls
span=1,0,0,esf,b8zs,crc4
This needs to be span=1,1,0,esf,b8zs
I'm not sure if the crc4 is necessary.
Doug
I concur with Doug. I have two PRI's in one system. My zaptel.conf
looks like this:
span=1,1,0,esf,b8zs # PRI line - LD Qwest (interstate)
bchan=1-23
dchan=24
Hi All -
I've been trying to compile Zaptel w/ HPEC, but I've been
unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I
believe I've got all the requisite files, and they're in the right
locations in the zaptel tree. When I compile, I get the following
warning from make:
Warning:
And I just saw that Zaptel 1.2.16 is out. I'll give that a try...
On 3/20/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi All -
I've been trying to compile Zaptel w/ HPEC, but I've been
unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I
believe I've got all the requisite files,
Indeed, the IRQ priority of my card is low. Sorry for being lame, but
does that have to be set in BIOS? Are there any best practices for
choosing an IRQ or I can set it to any free number?
I would start with IRQ sharing. Make sure that your Zap hardware isn't
sharing an IRQ. Secondly, you
And I just saw that Zaptel 1.2.16 is out. I'll give that a try...
And no... the problem still exists with zaptel 1.2.16
On 3/20/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi All -
I've been trying to compile Zaptel w/ HPEC, but I've been
unsuccessful. The system is CentOS 4.4, zaptel
That is going to depend upon your motherboard hardware, most likely.
Understand that in most cases the NIC and hard drives are usually going
to be the most demanding interrupt resource competitors to the Zap
hardware, and that the Zap hardware cannot often operate properly (i.e.
for fax)
And I just saw that Zaptel 1.2.16 is out. I'll give that a try...
And no... the problem still exists with zaptel 1.2.16
Problem solved. The warning meant nothing, compile was fine. For
some reason the old non-HPEC zaptel kernel module wouldn't unload.
Now, why that was, I don't know.
I've downloaded:
asterisk-1.4.1
zaptel-1.4.0
I've compiled and installed zaptel. When I go to install asterisk I do:
./configure
make menuselect
I then take a look under the codec selection menu and I see that
codec_zap can not be compiled.
Hi Rob -
After about having the server running for about an hour, our callers
occationally hear a high pitched beep that lasts the entire call. In
some cases, the noise doesn't start until a minute or 2 into the call,
while others last the entire call. In some of the more serious cases,
calls
On 3/16/07, Trevor Peirce [EMAIL PROTECTED] wrote:
Mark Quitoriano wrote:
Hi i have an asterisk pbx with E1 port connected to another PBX. Im
trying to send the DNID/DNIS to the PBX here's my dialplan
exten = 888111,1,Dial(ZAP/g2)
exten = 888111,n,Hangup()
The PBX just get the
The decision to use CentOS was(is) simple for me. That is the standard OS
chosen by Asterisk and FreePBX developers more or less. At least it was in
the early days. From there, the majority of people using Asterisk/FreePBX
have chosen CentOS.
So in a nutshell, it is the most used, most tested,
Cory Andrews wrote:
I have a client application looking for an Asterisk based solution.
Client wants to deliver pre-recorded messages for a variety of clients.
Wondering if anyone is offering an middleware for Asterisk for
management of outbound messaging?
Someone can correct me if I'm
I ended up using text2wave to create a wav file and then added it to the
prompt list and that worked.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
William Moore wrote:
The name of the option is fairly clear. If you want echo cancellation
even when the call is bridged directly from card to card, set the
option to 'yes'. Otherwise, set the option to 'no'. This option
should be 'no' in the majority of cases.
Thanks, that's what I
These folks have 6-8 T's worth of outbound they do on a daily basis, I
need an interface that would allow them to stick a comma delimited file
or file(s) in every day via FTP, the file would contain call #'s, and
some additional variables, and then the Asterisk box would schedule the
calls. It
thanks, and sorry because the mailing list nothing to do with pear.
but it'sa for install freePBX.
pear install DB not works. more any sugestion?
2007/3/20, dima [EMAIL PROTECTED]:
perhaps you should try
pear install DB
However note that this mailing list has nothing to do with pear.
Hi, i
It sounds like what you want is called a predictive dialer? There are
several listed on the voip-info wiki.
On 3/20/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Cory Andrews wrote:
I have a client application looking for an Asterisk based solution.
Client wants to deliver pre-recorded messages
Hi Dima,
You're better off following the Ubuntu guide written by the FreePBX
developers: http://aussievoip.com.au/wiki/freePBX-Ubuntu
Alex
On 3/20/07, dima [EMAIL PROTECTED] wrote:
perhaps you should try
pear install DB
However note that this mailing list has nothing to do with pear.
Hi,
This is a PRI 24 channel line. We have backup pots lines, but they
aren't in use. The problem we were having was happening on only a single
channel or 2.
Rob
Noah Miller wrote:
Hi Rob -
After about having the server running for about an hour, our callers
occationally hear a high pitched
Cory Andrews wrote:
These folks have 6-8 T's worth of outbound they do on a daily basis, I
need an interface that would allow them to stick a comma delimited file
or file(s) in every day via FTP, the file would contain call #'s, and
some additional variables, and then the Asterisk box would
Hi
For starters, when you want to post a message to the list, don't just
reply to an existing message. Start a new message. If you look at the
archives of this list, you'll see your message as a reply to another
message. This is because when you reply, the miler preserves some
threading-related
Could you be having ECFO? See:
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022062.html
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022111.html
Rob Schall wrote:
This is a PRI 24 channel line. We have backup pots lines, but they
aren't in use. The problem we were
From: Rob Schall [EMAIL PROTECTED]
Date: Tue, 20 Mar 2007 16:00:01 -0500
Cory Andrews wrote:
These folks have 6-8 T's worth of outbound they do on a daily basis, I
need an interface that would allow them to stick a comma delimited file
or file(s) in every day via FTP, the file would contain
On Tue, Mar 20, 2007 at 12:07:30PM +, Carlos Jerónimo wrote:
Hi, i try install FreePbx by tuturial in
http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443
but i have this error when i try install freepbx:
#pear install db
mitcheloc wrote:
Is that FUD really necessary?
No. Everyone can see this will be a disaster without the FUD.
-Stephen-
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Looks like user interface is not a concern - if they are thinking of
FTP
text files. In this case, a simple script to kick off some call files
should suffice. Won't take a week. (Search for call file.) But
having to
deal with answering machines is always tricky for any automation.
Yuan
Sorry for my bad english,
I've developed a web interface for one of our customers, that allow them
to create lists of telephone numbers, (even from excel files), and then
a couple of scripts, one of them running on the background:
Script 1: Reads the file containing the telephone numbers and
I've never seen a PRI dchannel on a T1 on a timeslot other than the
24th. Are you sure that it's really on channel 23?
Matthew Fredrickson
On Mar 20, 2007, at 8:54 AM, Kanelbullar wrote:
Thanks for your answer, Bruno. However, the configuration you provided
is for an E1 connection and we
I got money back around 6 months ago . It was a via paypal claim and hey
didn't reply till paypal's deadline so i got $30 back .
On 17/03/07, Ira [EMAIL PROTECTED] wrote:
At 02:32 PM 3/16/2007, you wrote:
You were able to cancel service with Sellvoip? That's impressive, that
Actually, it's
I've never seen a PRI dchannel on a T1 on a timeslot other than the
24th. Are you sure that it's really on channel 23?
I think he meant channel 23 of channels 0~23, aka the 24th channel.
-MC
Matthew Fredrickson
On Mar 20, 2007, at 8:54 AM, Kanelbullar wrote:
Thanks for your answer,
Next generation bot nets -- forget about spam, telemarketing is the
next viral (literally) marketing concept!
On 3/20/07, Jon Pounder [EMAIL PROTECTED] wrote:
Quoting C F [EMAIL PROTECTED]:
Awesome, the first PABX virus is just around the corner now that M$ has some
bait for it to infect.
I've been running the 8/1/2004 Head release up until a little over a
week ago. I was forced to due to a card failure to upgrade to 1.2.16
without any advance preparation or testing (most of my connections
are via satellite to all corners of the globe with high latency).
Up until the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Are you using Answer() before VoiceMailMain()?
Stu
Timothy McKee wrote:
I've been running the 8/1/2004 Head release up until a little over a
week ago. I was forced to due to a card failure to upgrade to 1.2.16
without any advance preparation or
Hi to all,
I was looking in google and also in this mailing list, but I dont find the
solution to my problem, so I subscribe me to the list in order to post this
e-mail and find the solution.
This is the scenario:
GSM Phone - GSM Network TDM2406E --- ASterisk 1.4.0 (*)
VoIP
On 3/19/07, Brad Sumrall [EMAIL PROTECTED] wrote:
...music on hold...
Brad
Music on hold support is present, you can also add MP3 support with
asterisk-addons package, are you using MP3 without the correct format
installed? www.asterisk.org and download the add-ons package, read the
docs
I really have lost loads of faith for IAX. No authority found and
Rejected connect attempt messages for no apparent reason. Sometimes
computability issues between asterisk versions. Fax/T.38 support?? But
I have no complaints about when it actually does work.
Not that Asterisk has the best SIP
Jeremiah Millay wrote:
I'm a little confused. I'm running a sangoma a200 card in my server. I
thought I needed codec_zap. Do I need to wait until zaptel-1.4.1 gets
released to be able to compile this or do I just not need it all
together? Any insight would be appreciated.
You don't need
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