Thank you Thank you Thank you! I changed it to: exten =>
s/555333,1,Gosub(subBusy,s,1()) and it now works like a charm. Really
appreciate the help!
El sáb, 20 nov 2021 a las 10:55, escribió:
> On 11/20/2021 11:51 AM, Richard Reina wrote:
> > Since Macro is deprecated
Since Macro is deprecated I am trying to eliminate it from my diaplan. I
believe I have successfully done so in the example below.
; dial an internal extension
exten => 101,1 Macro(ext,100,Dahdi/15)
TO:
exten => 101,1,Dial(Dahdi/15,30)
So far it seems to work. However I also in my dialplan
I am getting zero interrupts for a new Digium TE134 Card on a new brand new
Dell T40 server with the latest BIOS. Is there something that I am missing
or is the card not compatible with Dell servers?
(cat /proc/interrupts ; sleep 1 ; cat /proc/interrupts) | grep -i wcte13xp0
16: 0
Enviado desde mi iPad
El abr 16, 2014, a las 6:45 p.m., Sean Darcy seandar...@gmail.com escribió:
On 04/15/2014 06:52 PM, Kai-Uwe Jensen wrote:
Oops, had it wrong. Here's how it works for me:
[callcentric-template](!)
type=friend
context=from-callcentric
fromdomain=callcentric.com
Hi Doug,
Thanks for the reply. Unfortunately I can't get my telco do do anything
because I can't provide proof that is a problem with their lines.
2011/10/22 Doug Lytle supp...@drdos.info
Richard Reina wrote:
I have a server that is hooked to a channel bank (Adit 600). It has eight
lines
I have a server that is hooked to a channel bank (Adit 600). It has eight
lines coming from a T1 through this adit's FXO card. This particular *
server is used primarily as a PBX and is not connected to the internet and
has worked fine since 2006. However recently occasional calls are dropped
I just checked my Master.csv file and found that no logs had been written to this file since 7:30 yesterday morning. The system is heavily used and nothing is being recorded. Has anyone ever seen this before? Does the Master.csv fill up? It's current size is 8591056 but there are no other
Nevermind this post. The machines time was incorectly updated. The Master.csv is fine and performing dutifuly.Richard Reina [EMAIL PROTECTED] wrote: I just checked my Master.csv file and found that no logs had been written to this file since 7:30 yesterday morning. The system is heavily used
: Did you check your mpg123 version ?, asteriskneeds a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject
I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf including installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now. However, no sound is heard and I get this message from the CLI
version ?, asterisk needs a specific version in order to work... - Original Message -From:Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02AM Subject: [Asterisk-Users] Music On Holdnot working with new 1.2.7.1 install I
Dear Asterisk Nation,I am attempting to write a perl AGI script that will give the caller status of a P.O. When I run the script directly (by hand) it executes. I know this because it leaves data in TEST.txt. However, when I try to execute it via extensions.conf the CLI says it executes
ecifically for speaking amounts of money). *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Richard Reina *Sent:* Friday, 24 February 2006 9:34 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-
I am trying to give users the option to query our accts. payable database by supplying their PO number. I able to write queries via perl-DBI-mysql but have no idea how to get * to do it from the IVR. Is this possible? Can anyone point me in the right direction for help or examples?Thanks,
I have been looking for analog phones for my * system that work with our plantronic amplifiers and headsets. The problem I am having with the Aastra phones that I have purchased (PT-390, 9116, 9120, 8009 ), is that they don't seem to stay hung up unless you physically hang up the handset
Thank you very much. Your response was very helpful.
--- Ken Godee [EMAIL PROTECTED] wrote:
Does anyone know where I can buy a 50ft crossover
cable to connect my
digium card -- I believe it's a T100P -- to my
Adit 600. The one I have
now works fine but I need a longer one.
Why not search the wiki first?
Actually, I did search the wiki first and was not able
to get the wires to line up inside the RJ-45
connector. So I am looking to buy. Also, if you read
at the bottom one person said that the pinout there
did not work for him.
Does anyone know where I can buy a 50ft crossover cable to connect my digium card -- I believe it's a T100P -- to my Adit 600. The one I have now works fine but I need a longer one.Thanks,Richard
Yahoo! FareChase - Search multiple travel sites in one click.
Does anyone know how to put an Aastra PT 390 in
headset mode, so it will only give a dial tone when
you are ready ? Right now I can't figure how to keep
it hung up? If I hit googbye it merely flashes (give
me a dial tone again).
Any help would be greatly appreciated?
After buying some additional lines from my telco, I
recently had my phone vendor wire the additional lines
from my phone box into an amphenol connector that's
plugged into my channel bank (Adit 600).
However, although I make the following changes in my
zaptel.conf and my zapata.conf files
Anyway, you should have this as your
first line in the
script.
#!/usr/bin/perl
___
I had #!/usr/bin/perl5 -w
I changed it to #!/usr/bin/perl and now it works.
Thanks for the help
__
Do you
I installed the AGI perl library then put the
following script in a file called
/var/lib/asterisk/agi-bin/send_clid.agi,
updated my [incoming] context with exten =
s,1,AGI(send_clid.agi) and did a restart now.
use Asterisk::AGI;
my $agi = Asterisk::AGI-new();
my %input = $agi-ReadParse();
my
Thanks very much for the suggestions. I've
implemented them, but the main problem seems to be
that the program send_clid.agi is not executing
despite the cli saying that it is. If you have other
ideas let me know.
Thanks again,
Richard
--- Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Richard
As soon I do a reload I see contant ringing like this
on the CLI:
-- Zap/14-1 is ringing
-- Zap/23-1 is ringing
-- Zap/22-1 is ringing
-- Zap/20-1 is ringing
-- Zap/19-1 is ringing
-- Zap/14-1 is ringing
-- Zap/23-1 is ringing
-- Zap/22-1 is ringing
-- Zap/20-1 is ringing
-- Zap/19-1 is ringing
Do you have the Adit600 configured correctly? It's
not stuck in a test mode
or anything?
I have no idea if it's configured correctly. We just
kind of hooked it up when the install was done a
couple months ago.
-A.
___
Asterisk-Users mailing
You can do this with an agi script.
We are doing this in our app too.
We have a webbased crm app and * looks up the number
there
and inserts a record into a table so our app can
read that.
When the call hangs up, the record is deleted from
the
database.
It's not really that hard to
Michiel,
Thanks very much for the resonse. I am confused
however by fopen(/var/log/asterisk/my_agi.log
my * system has not such log file only the Master.cvs
which only seems to log a call one its teminated?
--- Michiel van Baak [EMAIL PROTECTED] wrote:
On 04:07, Mon 28 Mar 05, Richard Reina
--- Michiel van Baak [EMAIL PROTECTED] wrote:
I created that file myself.
That way I can put debug information into that
logfile while
developing that agi script.
It's part of my skeleton agi script ;)
Please pardon my ignorance, but how did you get
asterisk to pass that into to your
I would like to have asterisk pass along the caller ID
phone number to a database server on a my local
network (the same network that the * server resides on
) so that our customer service app. can pull up
customer data automatially. Asterisk passes along
caller ID to the phones fine, can
Yes there is. Try:
Dial(Zap/G1/w${EXTEN})
The capital G makes * grab channels in the opposite
order as little g. Hope that helps.
Richard
--- Alejandro G [EMAIL PROTECTED] wrote:
Hi,
If I have a PRI with all channels grouped in
group=1, I understand when I
want to make an outgoing
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On February 12, 2005 07:31 pm, Richard Reina wrote:
On thing that is odd is that although the t1 cross
over cable is plugged in to both * and the Adit.
Both
t1 and t1 leds on the Adit are red. How can they
both
have the same status
For whatever it's worth, it was the crossover cable.
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On February 12, 2005 09:21 pm, David Coulson wrote:
If he gets a green light with a loopback plug
wired like that, his
controller is definatly screwed up :-)
1-4
2-5
That was how I
After months of setting up Asterisk. I completed the
final testing last night We would go live on Monday.
Or so I thought.
As I moved the Adit 600 back out of the way, sliding
it six inches. I noticed the major light was red as
were both T1 and T2 lights ( eventhough only one t1
port is being
a monitor to?
What alarms is * reporting?
* seems to behave as if nothings wrong. There are no
errors.
Lyle
- Original Message -
From: Richard Reina [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, February 12, 2005 11:05 AM
Subject: [Asterisk-Users
cable have some loose wiring?
There should be a 'craft' port to hook up a serial
port with a term program
and you can poke around and see what alarms it's
reporting.
What alarms is * reporting?
Lyle
- Original Message -
From: Richard Reina [EMAIL PROTECTED]
To: asterisk
Googling the archives there is some debate about what
are good analog phones to use with *. Aastra seems
popular, but they are somewhat pricey and the
proprietary seems like it can be a headache. Can
someone weigh in on what would be good analog phones
for a small office (8 lines and 20 phones)
Does anyone kmow what these errors mean or how they
can be fixed. I'm using asterisk on a Fedora Core 2
box with a TDM400P with 2 fxo and 2 fxs ports.
Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469
ss_thread: Channel Zap/4-1 in prering state, but I
have nothing to do. Terminating simple switch,
For threeway calling (analog phone) I just hit the
flash button get a dial tone, dial the number and hit
the flash key again.
--- PHP Mechanic [EMAIL PROTECTED] wrote:
Hi, I have a TDM411B and when I am using asterisk
I can't get hook/flash
or
hold to work when using asterisk, which
I am using version: CVS-v1-0-12/13/04-18:46:23 with a
TDM400p (2 fxo, 2 fxs ports) and I keep getting errors
along with phantom calls:
Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363
ss_thread: Got event 17 (Polarity Reversal)...
Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434
ss_thread: CallerID
I am hoping to replace my Nortel 8x24 switch with
Asterisk. Right now my cabling comes from my outside
phone box into my office and into a punchdown block
and leaves the punchdown block as an amphenol
connector which currently plugs into the Nortel
swicth. A second amphenol connector them plugs
--- Henry Devito [EMAIL PROTECTED] wrote:
I am hoping to replace my Nortel 8x24 switch with
Asterisk. Right now my cabling comes from my
outside
phone box into my office and into a punchdown
block
and leaves the punchdown block as an amphenol
connector which currently plugs into the
600 do I use for
hooking up my eight analog incoming phone lines?
Thanks again for your help.
If my questions are unclear (not suprising since I am
completely clueless) feel free to call me toll free at
888-448-7874.
Richard Reina.
--- Brent Franks [EMAIL PROTECTED] wrote:
-Original Message
I have purchased an Adit 600 but with 6 FXS 8 channel
cards. Can somone tell me where I plug analog phones
in. The cards do not have any ports on them.
Thanks
Richard
__
Do you Yahoo!?
Check out the new Yahoo! Front Page.
www.yahoo.com
I am interested in implementing Asterisk and someday
hope to have it replace my 8 x 24 Nortel switch.
However, I was told by a Telcom friend that my multi
line phones (Nortel 7208s) may not work with Asterisk.
This is a huge concern because in my business we are
constantly jumping back from one
?
Thanks Again,
Richard
--- Walt Reed [EMAIL PROTECTED] wrote:
On Thu, Nov 04, 2004 at 07:18:38AM -0800, Richard
Reina said:
I am interested in implementing Asterisk and
someday
hope to have it replace my 8 x 24 Nortel switch.
However, I was told by a Telcom friend that my
multi
line
Thank you very much for that clarification.
--- Scott Laird [EMAIL PROTECTED] wrote:
On Nov 4, 2004, at 9:23 AM, Richard Reina wrote:
Thank you very much for your thoghtful and
thorough
response.
I guess I don't wan't to set up * to behave like a
key
system, thank godness, I
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