Gondar Monn wrote:
That's what happens when illegal aliens, er, Undocumented Americans, do
all your contracting work.
Could it be that all fires that ever happened in the US were caused by
those guys ? ..
I guess 1+1=5 then .
No, but being Houston, they must have far more
I wanted to setup Oreka to monitor calls on a trixbox box I have
setup. Oreka doesn't seem to be catching all of the calls
though I have port mirroring setup on the port that trixbox is
connected to, mirrored to the port Oreka is connected to.
I have read
Hi,
I've read that the new dahdi channel supporting B410P doesn't support PtmP
when in NT mode.
Does it support PtmP when in TE mode ?
Cheers
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To
It doesn't matter where the program resides as long as it can interact with
your asterisk (FTP, HTTP, etc). You just have to have the right accesses
and security.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, December 08, 2008
hi
where i should load the module for the trasncoder wctc4XX (lspci shows
TC400P)
thanks
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
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Modprobe wctc400p will load the module. You will then need to (re)start
asteriskl.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: Tuesday, December 09, 2008 8:13 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] about trasncoders
Olivier wrote:
I've read that the new dahdi channel supporting B410P doesn't support
PtmP when in NT mode.
Does it support PtmP when in TE mode ?
Yes.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)
i know but i dont want to write modprobe every time i reboot the server...
there is a file but i cant remember the name...
2008/12/9 Danny Nicholas [EMAIL PROTECTED]
Modprobe wctc400p will load the module. You will then need to (re)start
asteriskl.
--
Usually aren't those loaded using zaptel. On my machines you edit the
/etc/sysconfig/zaptel file, and comment out the unused modules leaving only the
ones you need.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: Tuesday, December 09, 2008 10:31
To: Asterisk
This is correct. Actually you can leave them uncommented (I did :-) ) and
the O/S should only load the attached hardware. This depends on your
kernel however and best practice dictates that you only uncomment the
modules you originally modprobe'd for.
_
From: [EMAIL PROTECTED]
nop
is not there... is an asterisk file and is two files one for the module
and other one for the parameters.
2008/12/9 Darrin Henshaw [EMAIL PROTECTED]
Usually aren't those loaded using zaptel. On my machines you edit the
/etc/sysconfig/zaptel file, and comment out the unused modules
2008/12/8 Karl Fife [EMAIL PROTECTED]
Does anyone know of a small lightweight windows 'dialer' application I
can use to trigger a call (via call file or AMI) from any application? (The
call would be placed between the target number, and the preconfigured DN of
the hardphone at the user's
Have you looked at ADM http://adm.hamnett.org/ ?
I don't know how it works on Windows but on Linux you can highlight any
number on the screen then leftclick on dialer icon, middleclick into the
dialer to make a call.
regards,
Drew
Danny Nicholas wrote:
It doesn't matter where the program
Hi,
In voicemail.conf, you can read
; Look in /usr/share/zoneinfo/ for names of timezones.
; Look at the manual page for strftime for a quick tutorial on how the
; variable substitution is done on the values below.
Where can this manual page for strftime be found ?
man strftime and apt-cache
Hi,
In voicemail.conf:
; Supported values:
; 'filename'filename of a soundfile (single ticks around the filename
; required)
; ${VAR}variable substitution
; A or aDay of week (Saturday, Sunday, ...)
; B or b or h Month name (January, February, ...)
; d or e
2008/12/9 Kevin P. Fleming [EMAIL PROTECTED]
Olivier wrote:
I've read that the new dahdi channel supporting B410P doesn't support
PtmP when in NT mode.
Does it support PtmP when in TE mode ?
Yes.
Thanks for replying !
--
Kevin P. Fleming
Director of Software Technologies
Digium,
You should do a find /|grep zaptel and edit the file that isn't in the
/etc/asterisk directory.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: Tuesday, December 09, 2008 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
http://linux.die.net/man/3/strftime http://linux.die.net/man/3/strftime
has an explanation of this function.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Tuesday, December 09, 2008 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Tue, 9 Dec 2008, David fire wrote:
2008/12/8 Karl Fife [EMAIL PROTECTED]
Does anyone know of a small lightweight windows 'dialer' application I
can use to trigger a call (via call file or AMI) from any application? (The
call would be placed between the target number, and the
2008/12/9 Danny Nicholas [EMAIL PROTECTED]
You should do a find /|grep zaptel and edit the file that isn't in the
/etc/asterisk directory.
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *David fire
*Sent:* Tuesday, December 09, 2008
On Tue, Dec 09, 2008 at 03:02:56PM +0100, Olivier wrote:
Hi,
I've read that the new dahdi channel supporting B410P doesn't support PtmP
when in NT mode.
Does it support PtmP when in TE mode ?
Rather: it is libpri that does not support that. The DAHDI driver
technically does not deal with
Here is another resource
http://us2.php.net/strftime
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On Tue, Dec 09, 2008 at 04:07:54PM +0100, Olivier wrote:
Hi,
In voicemail.conf, you can read
; Look in /usr/share/zoneinfo/ for names of timezones.
; Look at the manual page for strftime for a quick tutorial on how the
; variable substitution is done on the values below.
Where can this
On Tue, Dec 09, 2008 at 12:30:34PM -0200, David fire wrote:
i know but i dont want to write modprobe every time i reboot the server...
there is a file but i cant remember the name...
Why doesn't it load automatically (through hotplugging) at boot time?
Is it blacklisted?
If it is, I believe
David fire wrote:
First zaptel is dahdi now
second it isnot in /etc/dahdi or /etc/asterisk
i know because i remember one time a guy from digium toldme where to put
it
Hopefully you are now using DAHDI so I'll just comment about DAHDI loads the
drivers.
When you run make
Shaun Ruffell wrote:
Hopefully you are now using DAHDI so I'll just comment about DAHDI loads the
drivers.
Must be nice, every time I have tried to load DAHDI it kernel panics.
Don't try it if you have a card that uses tor2.
This is now the second time in what, less than a year, that
2008/12/9 Tzafrir Cohen [EMAIL PROTECTED]
On Tue, Dec 09, 2008 at 04:07:54PM +0100, Olivier wrote:
Hi,
In voicemail.conf, you can read
; Look in /usr/share/zoneinfo/ for names of timezones.
; Look at the manual page for strftime for a quick tutorial on how the
; variable substitution
RE Kushner List Account wrote:
Shaun Ruffell wrote:
Hopefully you are now using DAHDI so I'll just comment about DAHDI loads the
drivers.
Must be nice, every time I have tried to load DAHDI it kernel panics.
Don't try it if you have a card that uses tor2.
This is now the second
2008/12/9 Shaun Ruffell [EMAIL PROTECTED]
David fire wrote:
First zaptel is dahdi now
second it isnot in /etc/dahdi or /etc/asterisk
i know because i remember one time a guy from digium toldme where to put
it
Hopefully you are now using DAHDI so I'll just comment about
Hi I have
On func_odbc
[EXEC]
readhandle=ressqlserver
writehandle=ressqlserver
readsql=${ARG1}
writesql=${ARG1}
I'm trying an update on dialplan:
exten= 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})})
On Cli:
WARNING[3579]: func_odbc.c:353
On Tuesday 09 December 2008 09:14:11 Olivier wrote:
Hi,
In voicemail.conf:
; Supported values:
; 'filename'filename of a soundfile (single ticks around the filename
; required)
; ${VAR}variable substitution
; A or aDay of week (Saturday, Sunday, ...)
; B
On Tuesday 09 December 2008 11:45:59 Sebastian wrote:
Hi I have
On func_odbc
[EXEC]
readhandle=ressqlserver
writehandle=ressqlserver
readsql=${ARG1}
writesql=${ARG1}
I'm trying an update on dialplan:
exten= 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})})
On Cli:
2008/12/9 Tilghman Lesher [EMAIL PROTECTED]
On Tuesday 09 December 2008 09:14:11 Olivier wrote:
Hi,
In voicemail.conf:
; Supported values:
; 'filename'filename of a soundfile (single ticks around the filename
; required)
; ${VAR}variable substitution
;
Hi,
Say I wanted to know what context a SIP registration is using to dial out in
my dialplan, what would I do?
For example, I have phones on a local-calls-only context (as defined in
sip.conf), others in unrestricted-calls. In my dialplan, I`d like to act
on that knowledge.
Mike
Core show channels will show you this. Here is an example
Channel Location State Application(Data)
Zap/1-1 [EMAIL PROTECTED]: Dialing AppDial((Outgoing Line))
SIP/104-085ff278 [EMAIL PROTECTED] RingDial(Zap/g1/ww2975000|60)
2 active channels
Mike wrote:
Hi,
Say I wanted to know what context a SIP registration is using to dial out in
my dialplan, what would I do?
For example, I have phones on a local-calls-only context (as defined in
sip.conf), others in unrestricted-calls. In my dialplan, I`d like to act
on that knowledge.
2008/12/9 Olivier [EMAIL PROTECTED]
2008/12/9 Tilghman Lesher [EMAIL PROTECTED]
On Tuesday 09 December 2008 09:14:11 Olivier wrote:
Hi,
In voicemail.conf:
; Supported values:
; 'filename'filename of a soundfile (single ticks around the
filename
; required)
;
Hi,
I can't find a way to let users defined in a specific section to use a given
timezone.
Am missing something obvious ?
voicemail.conf:
[general]
tz=paris24
envelope=on
[zonemessages]
paris24=Europe/Paris|'vm-received' q 'digits/at' R
[mysection]
7530 = 1234,[EMAIL
On Dec 9, 2008, at 11:17 AM, Mike wrote:
Hi,
Say I wanted to know what context a SIP registration is using to
dial out in my dialplan, what would I do?
For example, I have phones on a local-calls-only context (as
defined in sip.conf), others in unrestricted-calls. In my
dialplan,
What parameter???
2008/1/17 gincantalupo [EMAIL PROTECTED]
Hi Olle,
that was a phone misconfigurationa parameter had a wrong value.
The message has disappeared and now the phone seems to work!
Thank you!
Giorgio
Johansson Olle E wrote:
10 jan 2008 kl. 16.48 skrev gincantalupo:
On Tuesday 09 December 2008 13:17:18 Mike wrote:
Say I wanted to know what context a SIP registration is using to dial out
in my dialplan, what would I do?
For example, I have phones on a local-calls-only context (as defined in
sip.conf), others in unrestricted-calls. In my dialplan, I`d
Having big problems and for months. Our service provider (via:talk)
says they are Asterisk friendly but they are not. Here are the
specifics (please read the bottom of the msg too)
System: Dell SM Business server 2GB RAM, Core II Processor (should be
plenty)
OS: open SUSE 11
Asterisk
The Asterisk development team is pleased to announce the release of versions
2.1.0 of both dahdi-linux and dahdi-tools. DAHDI now includes a native driver
for the B410P four port BRI module. The Digium Asterisk Hardware Device
Interface (DAHDI) is a collection of drivers and utilities for
Great, just what I needed. Thanks!
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: Tuesday, December 09, 2008 15:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk variable for
Hi John,
No you`re not over simplifying, that would be a great idea if I wasn't
building dynamically my sip registrations in realtime based on my own web
portal and was already finding the setvar column cluttered enough for other
values. That of course wasn't explained in my original question,
Dave,
Well that worked perfectly. Thank you.
A more advanced question then, for those who want to be challenged:
Brief: How can I find the values in the setvar field of a SIP peer, if I
don`t have access to ${var_a}.
Details:
When I place an outgoing call, I can easily access a variable that
Tilgham,
Sure, that works too, but I needed access to context taken from the sip
entry because I needed to goto(${that_context}), and that context varies
depending on the phone used.
This is when setting __TRANSFER_CONTEXT in my cookie cutter really
big/complexe/database driven extension. I
Mike wrote:
Sure, that works too, but I needed access to context taken from the sip
entry because I needed to goto(${that_context}), and that context varies
depending on the phone used.
This is when setting __TRANSFER_CONTEXT in my cookie cutter really
big/complexe/database driven
I could, but let's say phone B is limited to local calls, I wouldn`t want
the user to be able to transfer to non-local phone numbers.
Can you explain how your idea makes it simpler or better? I might be missing
the point.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi Gordon,
DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has the DIDs
that you require. And they can forward to IAX if that is preferable to you.
Regards,
Gideon
Date: Mon, 8 Dec 2008 17:27:44 +
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject:
2008/12/9 Olivier [EMAIL PROTECTED]
Hi,
I can't find a way to let users defined in a specific section to use a
given timezone.
Am missing something obvious ?
voicemail.conf:
[general]
tz=paris24
envelope=on
[zonemessages]
paris24=Europe/Paris|'vm-received' q 'digits/at' R
Hi,
When listening to the time and date a voicemail message was received, you
can hear french sentences like :
message reçu à vingt-et-un heure
(message received at twenty and one hour)
It should be (in whatever french flavour):
message reçu à vingt-et-une heure
In short, instead of:
--
This should be sufficient to get it to work from zoiper to zoiper.
http://asteriskguru.org/tutorials/zoiper2zoiperfaxt38.html
If you would still experience any issues, please send us a packet
capture + a description of the setup.
Cheers and good luck!
Zoa
Olivier wrote:
Hello,
2008/12/5
Brent Vrieze schrieb:
Here is what happens:
1. Asterisk verifies connection to the server and we get this. (CLI
output)
-- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net'
mapped to host galvatron.vtnoc.net, port 5060
-- ast_get_srv: SRV lookup for
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