/modules/
Restart your Asterisk and try it. If you want to call a NAT'd Asterisk
box, my Free World Dialup number is 18924. Currently online.
--
Leif Madsen [EMAIL PROTECTED]
http://www.hacklocalhost.com
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Sokol has written, as well
as all the work Jared is doing on getting the backend stuff setup
(DocBook, CVS, etc.. etc..).
Basically, I started the initiative, but Jared is doing all the work
right now! :)
--
+--+
|Leif Madsen - http
it.
Which is why I am learning to code, but I'm still a ways behind from
being able to turn Asterisk into that kind of thing.
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com
if you dial into voicemailmain,
and then hitting 0 to get to some extra options (so you can record your
busy and away messages, etc...)
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif
?
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
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.02 CDN.
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX| 1-700-363-0761 |
|iptel| 8972
, Linux
and Windows.
I will give it a shot. It's been a little bit since using it.
Thanks,
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms
remember mentioning I was totally opposed to
CVS) or whatever the case may be.
Sometimes I think you are a very bitter man, just for the sake of being
bitter :)
--
+--+
|Leif Madsen - http://www.hacklocalhost.com
.
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX| 1700-363-0761 |
|iptel| 8972-1969sipph| 1-747-386-1618
Laboratory Document Control System, but I don't
have $999 for a single user license :) (or any money at all actually)
Thanks,
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost
...
Leif Madsen.
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,
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX| 1700-363-0761 |
|iptel| 8972-1969
.
Thanks,
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX| 1700-363-0761 |
|iptel| 8972-1969
string to
lookup in our customer database, pulling up the callers record on the
agents screen.
I don't have this problem, but for archival purposes, would you mind
posting how you resolved your problem with examples?
Thanks,
--
+--+
|Leif Madsen - http
!
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX| 1700-363-0761 |
|iptel| 8972-1969sipph| 1-747-386-1618
upgrade.
I don't use the cvs command directly at all anymore unless the first
time I'm downloading from cvs on that box.
HTH
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com
.
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX| 1700-363-0761 |
|iptel| 8972-1969sipph| 1-747-386
?
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX| 1700-363-0761 |
|iptel| 8972-1969sipph| 1-747-386-1618
Steven Critchfield wrote:
On Mon, 2003-10-06 at 23:59, Leif Madsen wrote:
Hey all,
I am in the middle of creating a new user wizard which will generate all
the .conf's the new Asterisk user will require to get themselves up and
running in Asterisk without having to touch a single configuration
I've missed, feel free to chime in!
Thanks,
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX
Olle E. Johansson wrote:
Leif Madsen wrote:
think of things I've missed, feel free to chime in!
Leif, why don't you put the script up on the wiki so that we all can
edit it and add on line with versioning?
It's an interesting idea.. but I'm not sure if a wiki is the best place
for code
+ Logging verbosity
+ Manager
--
+--+
|Leif Madsen - http://www.hacklocalhost.com|
+--+
|@| leif at hacklocalhost dot com |
| SMS| sms at hacklocalhost dot com |
| FWD| 18924 IAX
the ability to
transfer calls)..
I can make a call using iLBC but the sound that I hear is just a lot of
pop's and crunches..
Hey... they can't give you EVERYTHING. There has to be some reason to
buy the Pro version :)
Thanks,
Leif Madsen.
FWD: 18924
IAX: 1700-363-0761
Thanks,
Leif Madsen.
FWD: 18924
IAX: 1700-363-0761
___
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.
Thanks in advance,
+---+
|Leif Madsen - http://www.hacklocalhost.com |
+---+
|@| leif at hacklocalhost dot com |
| FWD| 18924 |
| IAX| 1700-363-0761
Leif Madsen wrote:
Hi All,
I have just compiled the newest version of mpg123 on a RedHat 9.0 system
(mpg321 has not been installed) and I am using the newest CVS version of
asterisk. Whenever I place any mp3 files in the
/var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery
indication?
+ Mailbox number(s) to be associated with this channel
+ Context
Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel?
+
--
+--+
|Leif Madsen - http://www.hacklocalhost.com
you update from CVS, make sure you add
asterisk-addons to the end
ie. cvs checkout asterisk libpri zaptel asterisk-addons
Thanks,
Leif Madsen.
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with Martin and Mark
* Ability to work in Huntsville, AL
* Experience with Asterisk a plus
* Enthusiasm in Asterisk, Linux, and OSS in general a plus
* Telephony experience a plus
Too bad I live in Ontario, Canada.
Leif Madsen.
___
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available way
to auto restart asterisk ??
Please advise if you could.
I've never used it, but this sounds a lot like what the safe_asterisk
script does...
/usr/sbin/safe_asterisk
/usr/src/asterisk/safe_asterisk
HTH,
Leif Madsen.
FWD: 18924
IAX: 1700-363-0761
Brian West wrote:
http://www.loligo.com/asterisk/current/
Hey, didn't think of that. Good idea. Will go and check it out.
Thanks,
Leif Madsen.
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a select * from asteriskcdrdb.cdr; to see if
anything has been written, and sure enough, just Empty.
Any and all help greatly appreciated!
Thanks,
Leif Madsen.
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Tilghman Lesher wrote:
Try this command (from your bash shell):
grep cdr_mysql /var/log/asterisk/messages
messages does not exist... ?
Thanks,
Leif Madsen.
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believe)
;cdr_mysql.conf
[global]
hostname=localhost
dbname=asteriskcdrdb
password=password
user=asteriskuser
;port=3306
;sock=/tmp/mysql.sock
Thanks Tilghman!
Leif Madsen.
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information such as
passwords that you would not want available to anybody who could read
the logfile.
OK, I will take a look at that. Now that I have everything working
though, I can figure out exactly what I want to log, and tone it down
quite a bit.
Thanks again!
Leif Madsen
stages, it might be
nice to have documentation thought about right from the beginning (as
opposed to an after thought)
Just my 0.02 cents CDN.
Leif Madsen.
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Leif Madsen wrote:
OK.. lets just simplify this a bit.
remote --TDM400P-- * --IAX-- * --SIP-- remote
I want to make a call from the very left asterisk box, through the right
asterisk box to the remote end. So I have a phone plugged into the left
* box, which is connected via an IAX
which is also the NAT / FW box
recieves the connection from the SIP remote end, then forwards all the
traffic over IAX to the NAT'd * box. I just tested it, and it works
fine! Once I get some more complex extensions.conf files setup, I will
post them.
Thanks,
Leif Madsen.
BTW: As for just
= madsen:[EMAIL PROTECTED]
[madsen]
type=friend
username=madsen
secret=password
auth=md5
context=intern
host=dynamic
In zapata.conf, the context=intern
*** GW Box ***
extensions.conf
---
[globals]
FWDUSERID=18924
FWDUSERNAME=Leif Madsen
CALLFILENAME=foo
FOO=foo
[intern]
include
Leif Madsen wrote:
Hi all,
Here is a graphical diagram of what I am trying to do:
SIP --- GW/NAT/* --IAX-- * --TDM400P-- Analog Phone
So I have incoming SIP calls go to the * on the GW, which I then want to
forward over IAX to the second * box behind the NAT GW. If I was to
place a call
the left *
box, through the right * box to the remote SIP connection, I'd be very
grateful! I also promise to document anything and post it to the list
for archival purposes, and on a website for an online reference.
Thanks in advance,
Leif Madsen
, and looks very interesting and
useful. Is anyone actually using this to do what John is looking to do?
Thanks,
Leif Madsen.
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the CPU go to 100% just isn't very practical since I would like
to use it for more than just *.
If I can think if anything else, I will reply to this post.
Thanks in advance for any help or direction,
Leif Madsen.
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the
development suite in RH. That should pretty much do it...
Leif Madsen.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (Cygwin)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQE/b8pz6gq3eQ0gpNURAo8RAKC38nnhfyWzbs5ZN7O7ih4D7SENPgCfQI+f
rETTnoVp//V9mqGoENUkXxM=
=ABpe
-END
explanation how it usually works, and if
someone was implementing an Asterisk system, how they would go about
providing a call rollover (single advertised phone number, but allow
multiple incoming calls)
Thanks in advance,
Leif Madsen.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (Cygwin
, as
that is where I monitor for vulnerabilities in my boxes.
- --
Leif Madsen.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (Cygwin)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQE/aGm16gq3eQ0gpNURAohaAKCg9RL93co6fAfoxJA0fgrSsor0hgCdE1y1
C5sAMippFb6fK7q0xiik6O4
phpconfig if I remember correctly. I would probably start
with that and help expand upon it.
Currently, other than that, there is no sort of GUI interface for
Asterisk. If you were to write one, I'm sure it would be greatly
appreciated :)
Thanks,
Leif Madsen.
-BEGIN PGP SIGNATURE-
Version: GnuPG
list all the equipment, or if
you sell this type of hardware, please email me. I have been interested
in the CarrierAccess Adit 600 and perhaps the Mediatrix hardware. If
someone has a suggested hardware configuration for what I am doing, it
would be appreciated!!!
Thanks in advance,
Leif Madsen
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Leif Madsen wrote:
| I think I'm going to tie the channel bank into the Asterisk box using a
| pair of T1's using a pair of digium T400P's.
Oops.. I meant pair of T100P's :)
Thanks,
Leif Madsen.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2
I'm curious if anyone has used a radio for MOH? If so, how did you set
it up?
I have a client who is interested in using a radio for the music on
hold, since that is what they did with their old phone system.
Thanks,
Leif Madsen.
___
Asterisk-Users
150, So and i want
to take it?
| so i dial some number and i take it... How to do this?
Call parking?
- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
[EMAIL PROTECTED]
FWD: 18924 IAX: 1-700-674-5480
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (Cygwin
Hi all,
I have access to QTelNet products apparently, and was wondering if
anyone had used them before? Any good? Bad?
http://www.qtelnet.com/index.htm
Thanks!
Leif Madsen.
FWD: 18924
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http
Scott Dudley wrote:
I'd like to play with Asterisk and learn more. I have an FXO card but
need a station card. Can you recommend a cheap and readily available
FXS card that supports 1-4 stations?
Also, a peer of mine asked if any work has been done with cell phone
integration such that the
be accomplished without enabling IAX?
I can imagine this has been asked on the list.. But here are some
links to sample configuration files.
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extension
s.conf
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf
- --
Leif
version 4.2 :)
- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
@: [EMAIL PROTECTED]
ICQ: 3445119FWD: 18924
IAX: 1-700-674-5480
-BEGIN PGP SIGNATURE-
Version: PGPfreeware 6.5.8 for non-commercial use http://www.pgp.com
iQA
: Friday, September 12, 2003 3:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call Parking
To followup this is what debug shows:
Does anyone have any idea what the problem is?
- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
@: [EMAIL PROTECTED
informative.
- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
@: [EMAIL PROTECTED]
ICQ: 3445119FWD: 18924
IAX: 1-700-674-5480
-BEGIN PGP SIGNATURE-
Version: PGPfreeware 6.5.8 for non-commercial use http://www.pgp.com
iQA
files.
- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
@: [EMAIL PROTECTED]
ICQ: 3445119FWD: 18924
IAX: 1-700-674-5480
-BEGIN PGP SIGNATURE-
Version: PGPfreeware 6.5.8 for non-commercial use http://www.pgp.com
iQA/AwUBP18//+oKt3kNIKTVEQKWmQCdE
provided.
- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
@: [EMAIL PROTECTED]
ICQ: 3445119FWD: 18924
IAX: 1-700-674-5480
-BEGIN PGP SIGNATURE-
Version: PGPfreeware 6.5.8 for non-commercial use http://www.pgp.com
iQA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alvaro Ivan Parres Peredo
Sent: Wednesday, September 10, 2003 4:15 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] NO TONE ON ZAPATA FXS CHANNEL
Hi
I've problem, i cant get tone on a FXS
, but I haven't found anything yet.
Has anyone had this problem and resolved it? I am calling from
analog device to analog device, using call parking.
Suggestions? Using the newest CVS also.
- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
@: [EMAIL PROTECTED]
ICQ
in sip.conf?
What is the general consciences for the allow=all
statement? Should it be used, should it be specific towards
those codecs supported, or removed?
And... is this syntax allowed:
disallow=all
allow=gsm
allow=ulaw
allow=alaw
- --
Leif Madsen - Telecommunications Technology
Sheridan
= fwd-out
include = local
include = iaxtel-out
[local]
exten = 6500,1,Wait,2
exten = 6500,2,VoicemailMain2
exten = 6500,3,DISA,no-password|intern
- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
@: [EMAIL PROTECTED]
ICQ: 3445119FWD: 18924
-BEGIN
I'll just add an asterisk
specific section and create little documents with that... although
that's not what I'm really after.
Leif Madsen.
FWD# 18924
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for. I'll let you know after I
do some testing with it.
Leif Madsen.
FWD #18924
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would be nice, as well as maybe a history
timeline or something like that?
Just some thoughts. If you know of something like this, please let me
know. In the meantime, I'll be googling some more.
Thanks,
--
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
ICQ: 3445119
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Saturday, September 06, 2003 10:42 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] OT: Creating documentation using a web
interface
Hello all,
I would like
).
Getting closer though. .. back to the google :)
Jac Kersing mentioned to me to try twiki. Has anyone else used this
with success to create documentation?
Installing now as a test platform.
Thanks,
Leif Madsen.
___
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[EMAIL
have no given enough
information somewhere, feel free to ask and I will explain anything that
I can.
Thanks in advance,
Leif Madsen.
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: [Asterisk-Users] I can't get a two way conversation going?
This is going to sound like a stupid question but I am going to ask it
anyways. I have been a victim of my own stupidity on this one before. Ok
here goes... is the microphone muted?
-Original Message-
From: Leif Madsen [mailto
should be looking for?
Thanks,
BTW: Neither are behind a NAT. Both computers have external IP
addresses. No firewalls enabled.
--
Leif Madsen - Telecommunications Technology
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4233 (handle_response): Failed
to authenticate on REGISTER to
'sip:[EMAIL PROTECTED];tag=as4b14216d'
Any idea's what this could be? Is this an asterisk setup problem, or
could it be FWD? Sorry to ask on this list, but I wasn't sure which one
to ask on.
Thanks,
--
Leif Madsen - Telecommunications
I've also been having problems with X-Lite build 1016, even then I seem
to be able to get MSN Messenger working. I haven't had the time to
debug what is going on though.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stuart Hirst
I think this is the same problem I was having yesterday. I still have
to go back through my config files and find out exactly what I did, but
I should be doing that sometime today. I'll let you know when I find
out.
Leif.
-Original Message-
From: [EMAIL PROTECTED]
context=ext
allow=all
register = 33***:[EMAIL PROTECTED]/6000
[fwd.pulver.com]
type=friend
secret=
username=33***
host=fwd.pulver.com
canreinvite=no
Thanks,
--
Leif Madsen - Telecommunications Technology
Pager (416) ***- (don't use anymore, SMS me)
Cell (519) ***-
SMS [EMAIL
I have actually fixed the problem now. I will reply when I know how I
fixed it :)
Leif.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Thursday, July 10, 2003 11:18 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk
a dial tone or get asterisk to call the
other phone. It doesn't seem to detect the ring tones when I dial
either.
If you could CC: any replies to [EMAIL PROTECTED] it would be
appreciated. I do watch the mailing list, but sometimes I miss stuff.
Thanks in advance,
--
Leif Madsen
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