On Thu, 2003-12-18 at 04:03, Brian West wrote:
Stop using beta firmware... I honestly think that GrandStream needs to
either fix the phones or stop making them.. THEY SUCKS! I think I would
rather eat glass than work with a grandstream phone.
bkw
Brian, GS has people that works very hard
Brian West wrote:
Accually CDR will not be generated if the target is an appliction.
exten = 1234,1,AGI,outbound.agi|19
then ref the exten not the appliction it will generate a cdr record.
http://bugs.digium.com/bug_view_page.php?bug_id=240
Comment by John Todd in bugs added to please
Hi folks,
Does anybody have any idea what this is;
WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect process 256 frames
WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect process 256 frames
WARNING[1232119104]: File dsp.c, Line 1198
When setting
include = daytime|9:00-21:00|mo-fri|*|*
How does this determine what is different between 9 AM and 9 PM
And after hours ???
I want different hours on Saturday and Sunday
And a different welcome message after hours
Any help appreciated
Regards Mick
Hello,everyone,
I encoutered some difficult with IAX when I run
the asterisk.
internet -- asterisk + NAT -- DIAX
my * box and NAT are at the same linux box which connecting to the internet
using ADSL. The box has two network cards and two IP address,such as
public
Hi,
- Original Message -
From: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX quesitons please.
Hello,everyone,
I encoutered some difficult with IAX when I run the asterisk.
internet -- asterisk + NAT -- DIAX
my * box and NAT are at the same linux box which connecting to
Question2:
If I dial the IAX2 user registed to my * inside my NAT,it will
success,but
if I dial other IAX2 user registed to my * in the internet (not inside
my NAT),I alway get the result:
== Everyone is busy at this time
Take care that there is an issue with DIAX and IAX2... after some
But the issue is not: 'how does the alternative feature work', the issue
is 'why is the original feature absent'. I haven't heard anyone giving
any reason whatsoever why * does not allow a user to retrieve an on-hold
call with old-fashioned flashing (or pressing #). I think that is what
the
Mike M. Tkachuk wrote:
Hello,
I'm using satellite link (1024/256) Eutelsat.
With Gnugk and Asterisk. The average roundtrip
to my Gateway (DualTalk) is about 650 ms.
I think that's fine for non business telephony,
just for calling to friends.
Hi, thanks for that.
Could you give me a
Hi!
I have no problems setting up trunk groups in general, but is there a way to
set up a trunk group for outbound calls that includes channels on multiple
servers? I might have missed something somewhere, but I couldn't find any
reading about this topic. Thanks!
What exactly are you
Hi,
I have not incoming phone number to test, but I think I can call
you. If I have termination to your country I'll call you (please give
me your stationary phone, not mobile).
--
Best regards,
Mikemailto:[EMAIL PROTECTED]
Hi!
How can I make * ring one phone then if no answer
Go to a different extension ??
Read the handbook draft which is to be found on www.asterisk.org.
Or read the Wiki and search for the description of the application DIAL.
*sigh*
Cheers, Philipp
matt wrote:
The problem is that I don't want to call an extension, I want to call
the number that was specified in the the connection i.e.
[EMAIL PROTECTED] number will be different every time it
is called so I don't want to have to put in an exten for every phone
number in the city I'm trying
I had a working configuration whereby an incoming call on an ISDN line
would be sent out on the second ISDN line, but since I updated to the
latest version of Asterisk I get this error message:
WARNING[311315]: File res_parking.c, Line 226 (ast_bridge_call): Bridge
failed on channels
Hi!
I need to come up with a solution that the user can place the caller on
hold, the caller here MOH and the user hang the receiver up. Just as if
they hit, the hold button on the phone. This can be done, using ADSI if
need be.
What you are trying to do doesn't seem to make much sense.
Hi!
But the issue is not: 'how does the alternative feature work', the issue
is 'why is the original feature absent'. I haven't heard anyone giving
any reason whatsoever why * does not allow a user to retrieve an on-hold
call with old-fashioned flashing (or pressing #). I think that is
Matteo Brancaleoni [EMAIL PROTECTED] said:
because is different new. Has new powerful features, and old
functions has been abandoned for new ones.
Yeah, so much is clear. However, because flash doesn't work at a certain
moment *and*, AFAIK, has no other functions at that time, I'm simply
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm having a problem with the following expression examples.
exten = s,1,NoOp($[$[${value} = 10] $[${value} 18]])
exten = s,1,GotoIf($[$[${value} = 10] $[${value} 18]]?3)
${value} is 13 in both examples above. First extension evaluates to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 18 December 2003 13:08, Tais M. Hansen wrote:
I'm having a problem with the following expression examples.
exten = s,1,NoOp($[$[${value} = 10] $[${value} 18]])
exten = s,1,GotoIf($[$[${value} = 10] $[${value} 18]]?3)
${value} is 13
Cees de Groot wrote:
Matteo Brancaleoni [EMAIL PROTECTED] said:
because is different new. Has new powerful features, and old
functions has been abandoned for new ones.
Yeah, so much is clear. However, because flash doesn't work at a certain
moment *and*, AFAIK, has no other functions
Michiel Betel [EMAIL PROTECTED] said:
I agree with Cees, however, not wanting to throw away the 3 way
conference feature, but giving the user a config choice might be best.
Therefore I'm now testing a patch which will allow/disallow the 3 way
conference. When disallowed it will fallback to
Ok...
Let me give a better example.
A caller calls in and a user picks up the phone. Then the user needs to
put the caller on hold so he can go check on something. He would like
to press the hold button on the phone and hang the receiver up. He can
do this, but the caller never hears MOH.
Asterisk Crash
I have an application that using the System() command. When ever I
invoke the command my asterisk crashes.
I have updated to the latest CVS and it crashes. Can someone offer some
suggestions on how to diagnose and correct this problem?
Thanks
Kevin
Extensions.conf
exten
Mike M. Tkachuk wrote:
Hi,
I have not incoming phone number to test, but I think I can call
you. If I have termination to your country I'll call you (please give
me your stationary phone, not mobile).
Ok, thanks for that.
(USA)1-212-400-7921
Ta
SJ
Hi!
A caller calls in and a user picks up the phone. Then the user needs
to put the caller on hold so he can go check on something. He would
like to press the hold button on the phone and hang the receiver up.
He can do this, but the caller never hears MOH. The user does what he
needs to
Hi All,
I was able to track down what I believe is a bug when using AGI
services. This bug may crash your system if your extensions.conf script
is intensive in using AGI services. Depending on your system's ulimit, *
keeps opening files until it reaches the system limit and then stops
responding.
Hi!
I'm somewhat unsynced with the features of todays CVS.
What's the status of the SQL-support for voicemail? Can you everything
(including messages) in the DB?
Do you still(?) have to recompile when you change the emailbody-variable
in voicemail.conf?
Is there anything else I should be
At 1:16 PM +0100 12/18/03, Tais M. Hansen wrote:
On Thursday 18 December 2003 13:08, Tais M. Hansen wrote:
I'm having a problem with the following expression examples.
exten = s,1,NoOp($[$[${value} = 10] $[${value} 18]])
exten = s,1,GotoIf($[$[${value} = 10] $[${value} 18]]?3)
${value} is
Hi,
I don't have a zaptel device for conferencing.
I read from the lists, that
ztdummy and zaprtc need to be installed to get conferencing.
I could able to compile successfully with ztdummy and when i receive the
call it says,
-- Goto (13732,s,1)
-- Executing
At 7:44 PM -0500 12/17/03, Sean Cheesman wrote:
Hello all,
I have no problems setting up trunk groups in general, but is there a way to
set up a trunk group for outbound calls that includes channels on multiple
servers? I might have missed something somewhere, but I couldn't find any
reading
On Wed, 10 Dec 2003, Cees de Groot wrote:
Chris Albertson [EMAIL PROTECTED] said:
My brother has the BEST solution for sales people. He makes
an appointment with them to come out and gives an address across the
street. It really wastes a real estate salesman or house painter's
time to
On Tue, 9 Dec 2003, John Breeden wrote:
Just started putting my first * together with a tdm400p and x100p.
Analog phones, xlite and diax I've got working.
Just got Grandstream budgetone-100.
The budgetone registers with * just fine. * accepts the dtmf and dials the
number. The remote
- Original Message -
From: Cees de Groot [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 7:58 AM
Subject: [Asterisk-Users] Re: transfer with threeway calling
Michiel Betel [EMAIL PROTECTED] said:
I agree with Cees, however, not wanting to throw away the 3 way
On Thu, 2003-12-18 at 02:26, [EMAIL PROTECTED] wrote:
When setting
include = daytime|9:00-21:00|mo-fri|*|*
How does this determine what is different between 9 AM and 9 PM
And after hours ???
I want different hours on Saturday and Sunday
And a different welcome message after hours
- Original Message -
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 7:08 AM
Subject: [Asterisk-Users] Expressions
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm having a problem with the following expression examples.
- Original Message -
From: Kannaiyan Natesan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 9:52 AM
Subject: [Asterisk-Users] Zaprtc compile error - virtual device for
conferencing
Hi,
I don't have a zaptel device for conferencing.
I read from the
At 10:46 AM -0800 12/17/03, Paul Mahler wrote:
While this thread is already in the archives, I'll throw my opinion on the
table, too.
The latency is about .25 seconds to and .25 seconds from the satellite.
There is additional propagation delay in the system. Also, TCP/IP relies on
propagation
On Thu, 18 Dec 2003 11:48:59 -0300
Paulo Mannheimer [EMAIL PROTECTED] wrote:
Hi All,
I was able to track down what I believe is a bug when using AGI
services. This bug may crash your system if your extensions.conf script
is intensive in using AGI services. Depending on your system's ulimit,
We have contracted with Eschelon to provide voice and data over a T1
link. The plan is to terminate this link at a T100P card in the * system.
The vendor has said that they will provide the D channel contiguous to
the voice channels (voice on channels 1-8 and D channel on 9). The
data
I have then downloaded newer version of the Asterisk about 10 days ago,
but
then my Asterisk would start to crash on me, the module would just stop
running by itself and I had to restart the Asterisk. Sometimes, it would
just stop running, but 75% of the time, I would see this error message:
- Original Message -
From: Greg Boehnlein [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 10:01 AM
Subject: Re: [Asterisk-Users] Re: Telemarketer Torture
On Wed, 10 Dec 2003, Cees de Groot wrote:
Chris Albertson [EMAIL PROTECTED] said:
My brother has the
Thanks for the reply but it could not solve my problem.
Did you modprobe ztdummy?
modprobe ztdummy
modprobe: Can't open dependencies file /lib/modules/2.4.20-6um/modules.dep
(No such file or directory)
Can you please guide me what should I do for this?
It should return nothing(successfully).
On Thu, 18 Dec 2003 14:10:08 +0200, Anton Yurchenko [EMAIL PROTECTED] wrote:
Hello,
I asked on the asterisk mailing list about dlink DG-104SH, some people
wrote that they had DG-104S working, so I kicked that 104SH , and got an
104S. And now I`m having trouble configuring it( I`m kinda new to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 18 December 2003 16:12, Andrew Thompson wrote:
Your example sort of confuses me. How do you know what NoOp returns?
It was two examples. I use NoOp() everytime I'm in doubt about the contents of
a specific variable or expression.
Even
Please forgive me if the answer is obvious, but my new Asterisk server
gives back a forbidden message when I try to call my UK office. It
should go out simply via X100p and PTSN. Here's the relevant lines from
extensions.conf.
[outbound-analog-int'l]
; allowed to call interntional long distance
I want to join two calls invoked from asterisk,
Here is my 1.call in /var/spool/asterisk/outgoing,
Channel: IAX2/[EMAIL PROTECTED]/847512,20,tr
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: 13732
Extension: s
Priority: 1
it successfully rings at extension 847512 and I could answer the call.
I use a Linksys BEFSR81 which ia an 8 port model with QoS. I paid about
$90 USD. I had to buy a QoS router when I first installed a Vonage line
about a year ago. Without it using FTP to d/l loarge files would simply
kill my calling.
Michael
On Wed, 17 Dec 2003 17:01:52 +0100, Thilo Salmon wrote:
It doesn't matter for the zaptel (since you can set dchan=any_channel) but
in chan_zap.c in asterisk dchannel for t1 cards is hardcoded to by on 24th
channel. You can change that though.
regards
Martin
On Thu, 18 Dec 2003, Michael Welter wrote:
We have contracted with Eschelon to provide
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Hash: SHA1
On Thursday 18 December 2003 15:43, John Todd wrote:
Answer() apparently changes channels and thus clears variables set prior
to the Answer() call. :(
If Answer() clears variables, then this is a bug, where bug is
defined as behavior that occurs
Hello,
We have updated the Wiki page for Polycom phones:
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones
We posted several configuration specs as well as a link to an admin guide
for the phone.
We also posted a link on there to two firmware versions for download.
The official
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 11:06 AM
Subject: [Asterisk-Users] International calling forbidden?
Please forgive me if the answer is obvious, but my new Asterisk server
gives back a forbidden message
Duh,
I finally go it! Missing } afterter PSTNOUTBOUND.
Michael
On Thu, 18 Dec 2003 10:06:23 -0600, Michael Graves wrote:
Please forgive me if the answer is obvious, but my new Asterisk server
gives back a forbidden message when I try to call my UK office. It
should go out simply via X100p and
I used root user but I do not understand what you mean running it by
prompt or screen.
These are the packages I installed. Do you know if some are missing?
David
[EMAIL PROTECTED] root]# rpm -q kernel-source readline readline-devel
openssl openssl-devel bison cvs gcc newt-devel ncurses-devel
Hello,
On Thu, 2003-12-18 at 13:06, Michael Graves wrote:
[outbound-analog-int'l]
; allowed to call interntional long distance numbers via PSTN
; dial 8 to signify overseas calling
exten = _8011,1,Dial(${PSTNOUTBOUND/${EXTEN},70)
exten = _8011,2,Macro(fastbusy)
The
SW wrote:
Hi folks,
Does anybody have any idea what this is;
WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect process 256 frames
Do not try to do inband DTMF on G.729
Jeremy McNamara
___
Asterisk-Users mailing list
On Thu, 2003-12-18 at 07:25, Kevin wrote:
Asterisk Crash
I have an application that using the System() command. When ever I
invoke the command my asterisk crashes.
I have updated to the latest CVS and it crashes. Can someone offer some
suggestions on how to diagnose and correct this
Great ;-)
Can someone else confirm this doesn't have any side effects besides
solving the problem?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angel
Carpintero
Sent: quinta-feira, 18 de dezembro de 2003 12:24
To: [EMAIL PROTECTED]
Subject: Re:
I am thinking about using the G729 codecs on my endpoint devices and
purchasing some G729 licenses for Asterisk but I have several questions:
1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I?
2. If I have G729A on one end and G729B on the other, are they compatible?
I have
On systems even key systems it is customary to have an internal
dial tone.
Since Asterisk simply ignores the 9 and keeps the tone going
it is hard to tell for some new users if they can make a call.
My first idea was to change the generated dial tone via
source. Then if the user
It doesn't matter ... A B are compatible
On Thu, 18 Dec 2003, Clif Jones wrote:
I am thinking about using the G729 codecs on my endpoint devices and
purchasing some G729 licenses for Asterisk but I have several questions:
1. Which G729 codec is sold by Digium for Asterisk, G729, G729A,
As far as I understand it, daytime is a context?
so you just use like
[daytime]
s,1,blahblah etc
[weekend]
s,1,blahblahweekend etc
[EMAIL PROTECTED] wrote:
Matt
I understand that bit but
How do I express the sound file for after that time period ??
Here is what I need to do
include =
On Thursday 18 December 2003 13:31, Alex Lopez wrote:
On systems even key systems it is customary to have an 'internal'
dial tone.
Since Asterisk simply ignores the 9 and keeps the tone going it is
hard to tell for some 'new users' if they can make a call.
My first idea was to change the
ztdummy isn't compiled by default.. you have to take the # from infront of
it in the Makefile. But then again it only works with usb-uhci and not
usb-ohci. Buy an x100p and call it a day.
bkw
On Thu, 18 Dec 2003, Kannaiyan Natesan wrote:
Thanks for the reply but it could not solve my
I have three cisco 7910 phones connected to * through skinny protocol. When
one of the phones is called, and the phone is ringing, you can hear what's
going on in the room even though the caller hasn't answered. It's crazy and
very hard to ignore when someone is calling :) God forbid you should
On Thu, 2003-12-18 at 13:30, mattf wrote:
Hello,
We have updated the Wiki page for Polycom phones:
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones
We posted several configuration specs as well as a link to an admin guide
for the phone.
We also posted a link on there to two
Sorry I can't help you anymore than that...hopefully some guru can...I
checked all of your versions against mine and they're EXACTLY the same...
Was it a clean install? I.e. newish?
Hopefully someone else will continue the thread from here...
Sorry,
Matt
David Luyens wrote:
I used root
Stevie
If you do not have any thing intelligent to say
Why waste both your time and ours
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, 19 December 2003 1:40 AM
To: [EMAIL PROTECTED]
Subject: Re:
Thanks
Regards Mick
[weekend]
s,1,blahblahweekend etc
[EMAIL PROTECTED] wrote:
Matt
I understand that bit but
How do I express the sound file for after that time period ??
Here is what I need to do
include = daytime|9:00-21:00|mo-fri|*|*
include = weekend|10:00-19:00|sat-sun|*|*
I
Found it. Anyone interested can look in RFC3551 RTP Profile for Audio
and Video Conferences with Minimal Control.
You can piece together that G.729, G.729a G.729b will play together
and the other annexes will not due to
bandwidth differences.
Clif Jones wrote:
I am thinking about using the
Hi,
I'm trying to setup Moh default config.
When I dial the ext. I get this:
WARNING[1200884528]: File res_musiconhold.c, Line 303 (moh0_exec): Unable to start
music on hold
(class 'default') on channel SIP/user1-f2d3
What could it be?
Tx.,
___
Are you sweet with it now?
The other option is to go to the documentation link on digium's website
where there are demo
config files...that's probably the single thing that helped me the most...
Also the people on the irc group can be nice from time to
timealthough it helps to be demure
Howdy,
I recently saw something strange with a call between *'s over IAX2.
There are actually 3 *'s involved. The setup is like this:
SIP phone --(ulaw over LAN)-- *1 IAX2 (ulaw over
Internet) -*2(GSM over Internet)
---*3(ulaw over LAN)--
The name bru1voip kindof looks familarish to me ;) (hi btw), but may
just be concidental.
What are the specs on the box you are trying to compile it on? (disk/ram/etc)
If its a pitiful machine, compile it on another machine, and transfer it
over. Sometimes Makefiles have a PREFIX (or so) option,
reorganized
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 3:30 PM
Subject: RE: [Asterisk-Users] after hours
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent:
On Thursday 18 December 2003 14:04, Tilghman Lesher wrote:
In the zaptel driver, create a new tone in zonedata.c at the END of
a zone (so you don't throw off existing tone indexes). Then, in
asterisk, in the specific channel driver (e.g. chan_zap.c), locate
ast_ignore_pattern and change
Hi,
I discovered a problem in asterisk with the following scenerio:
1) I make an outbound call
2) Called person answers phone
3) I hit the flashhook to initiate a 3-way call
4) I hear dial tone and called person is on hold
5) I hang up my phone
6) called person hangs up their phone
7) my phone
Although it's hard to see the original proverb writer saying RTFM
:-)
Matt
Andrew Thompson wrote:
Give a man a fish and he eats for a day. Teach him to fish and he eats for
a lifetime.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Yeah
And give him a gun
And look at what happened at Port Arthur
Regards Mick
Although it's hard to see the original proverb writer saying RTFM
:-)
Matt
Andrew Thompson wrote:
Give a man a fish and he eats for a day. Teach him to fish and he eats
for a lifetime.
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a coat closet.
Sadly, there is not enough space (or ventilation) for the monitor and
keyboard. This will all change when we move to new quarters, but...
Does anyone have
You can see any thing
Sorry I could not resist
If you need to admin Linux without a monitor
Try webmin
Regards Mick
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a coat closet.
Sadly, there is not enough space
well i never had a asterisk server with a monitor or keyword all my
servers i do remote login with ssh its better more private.
Miguel
On Thu, 2003-12-18 at 22:02, Michael Welter wrote:
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on
You will probably find quite a few people do. I know I do. I have a monitor
hooked up to a keyboard switch that is attached to my Asterisk server but I
never click over to it. I use SecureCRT and monitor the console that way only.
No need really for a monitor unless you want the graphical view
There are no issues. There is no reason to have a K/B or
monitor on the server. Just sucks up power and adds to
global warming. You may also want to pull any CDROM or
flopy drive from the box too for the same reason.
Seriusly, you should be using ssh from a remote machine to
access the
On Thu, Dec 18, 2003 at 03:02:42PM -0700, Michael Welter wrote:
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a coat closet.
Sadly, there is not enough space (or ventilation) for the monitor and
keyboard. This
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Welter
Sent: Thursday, December 18, 2003 5:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Headless Linux system for Asterisk
Because of space limitations and because of the
Michael Welter wrote:
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a coat closet.
Sadly, there is not enough space (or ventilation) for the monitor and
keyboard. This will all change when we move to new quarters,
Hi, the machine is a dual xeon, 1 G of memory and 160 G harddisk...
I am not such a linux guru and your suggestion kind sounds like chinees,
but thanks anyway.
I will try to do a new install and let you know...
David
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL
Thanks, Jeremy, that was indeed the problem.
Message: 2
Date: Thu, 18 Dec 2003 12:56:48 -0500
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unable to detect process 256 frames
Reply-To: [EMAIL PROTECTED]
SW wrote:
Hi folks,
Does anybody have any
Michael Welter wrote:
Does anyone have experience running Linux/Asterisk without a monitor?
What, if any, are the issues?
Most of my Linux boxes are sans-monitor. I highly recommend it. Just
be sure SSH works well and you haven't firewalled your admin workstation
from being able to
On Thu, 2003-12-18 at 17:02, Michael Welter wrote:
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a coat closet.
Sadly, there is not enough space (or ventilation) for the monitor and
keyboard. This will all change
I am currently using a copy of Asterisk checked out as the code of 10 days
ago from Asterisk and the:
sip show inuse
reports that I have 3 incoming connections to one of the Grandstream
phones, even though that isn't the case.
I believe I have tracked the problem down to the following error
Hi Clif,
My experience with G.729 and asterisk is not good.
My first registration was good, it worked. Then I bought more license and
tried to upgrade it, it blew everything off. Still waiting Digium support to
give me a helping hand.
If you use pass-through feature then I guess you are fine. I
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a
coat closet.
Sadly, there is not enough space (or ventilation) for the
monitor and
keyboard. This will all change when we move to new quarters, but...
Does
Hi. I just started trying to play with Sphinx. I followed their site as far as
running sphinx-server. It is listening on the default port. I copied sphinx2-simple
to another file and changed sphinx2-continuous to sphinx2-server.
So, I ran eagi-sphinx-test under asterisk. What exactly is it
Well I feel you are right there are a few people on this list
That could use a good kick.
Aren't there Andrew
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Thompson
Sent: Friday, 19 December 2003 7:41 AM
To: [EMAIL
Michael Welter wrote:
Does anyone have experience running Linux/Asterisk without a monitor?
What, if any, are the issues?
There is no fancy GUI, unlike that other semi-popular OS, so there is
absolutely no need for a monitor and keyboard on a Linux box.
Just make sure you make sure the BIOS
I'm also getting this issue, for some reason some calls yield mass VNAK's. I
also get iseq problems, but that might be my code - eg
DEBUG[98311]: File chan_iax2.c, Line 4368 (socket_read): Received iseqno 4
not within window 0-2
DEBUG[98311]: File chan_iax2.c, Line 4368 (socket_read): Received
On Thu, Dec 18, 2003 at 03:02:42PM -0700, Michael Welter wrote:
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a coat closet.
Sadly, there is not enough space (or ventilation) for the monitor and
keyboard. This
Reinstaling the OS is not going to do anything ut get you
right back to where yu are now.
Look at the last command the the Makefile tried to run.
Did it choke while running bison? If so run the bison
command by hand. What happens? You need to cd to
the directory where Makewas at when it
Run using a serial console
(http://www.tldp.org/HOWTO/Remote-Serial-Console-HOWTO/). No monitor,
VGA adapter, keyboard etc needed. Use SSH to log into the asterisk box
for any maintenance, etc. If the box gets hosed, connect the serial
port to a working PC and fire up minicom and your all
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