On Fri, 24 Jul 2020 at 14:23, Tim Požár wrote:
> You can post process the logs with something like sed. See:
>
>
> https://superuser.com/questions/380772/removing-ansi-color-codes-from-text-stream
Yeah; we're injesting using filebeat and you can do sed style parsing on
lines into it using
You can post process the logs with something like sed. See:
https://superuser.com/questions/380772/removing-ansi-color-codes-from-text-stream
On 7/23/20 5:10 PM, Andrew Yager wrote:
Hi,
Is there a way to drop the ANSI colour strings from log files? In
particular, I've got JSON logging
Well I skipped reading the part where you say “only for log files” :-(
On Thu, Jul 23, 2020 at 21:06 Joel Serrano wrote:
> Have you tried starting asterisk with the "-n" param?
>
>-n Disable console colorization
>
> On Thu, Jul 23, 2020 at 5:11 PM Andrew Yager wrote:
>
>> Hi,
Have you tried starting asterisk with the "-n" param?
-n Disable console colorization
On Thu, Jul 23, 2020 at 5:11 PM Andrew Yager wrote:
> Hi,
>
> Is there a way to drop the ANSI colour strings from log files? In
> particular, I've got JSON logging throwing logs over to ES,
Hi,
Is there a way to drop the ANSI colour strings from log files? In
particular, I've got JSON logging throwing logs over to ES, but they have
the ANSI colour escape sequences.
Ideally I don't want to lose coloured logs from the console though, and I
can't "see" a way to do this.
Ast 16 at the
Check the footer at the bottom of this message for instructions on how
to unsubscribe :-)
Matthew Fredrickson
On Fri, Jun 1, 2018 at 12:11 PM, David Mutterer wrote:
>
> --
> _
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
On Fri, Jul 29, 2016 at 11:01 PM, Nabeel wrote:
> Hello,
>
> I would like to remove the 'Comedian Mail' name/message played when a user
> tries to access their voicemail. Please let me know how to do this.
>
> Nabeel
>
It plays a file
Hello,
I would like to remove the 'Comedian Mail' name/message played when a user
tries to access their voicemail. Please let me know how to do this.
Nabeel
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2013/2/21 akhilesh chand omakhileshch...@gmail.com
hello all,
i have two asterisk server for call transfer and one more asterisk server
for agent login(server_X) where agent take the call.
server_A and server_B
server_A is connected with pri and configure with 60 channel for call
In server_X, 60 agnet are login and call comes from server_A and server_B
which is connected with pri(total 90 channel), Supposed to two call
originate 02:25 PM from A B and hit into server_X goes to agent
2002(agnet extension) one of call is abandoned and another one pick up
agent 2002 .Both
Set QUEUE_PRIORITY in dialplan for server A. Queue will pick call with
higher priority first.
regards,
Bharat
On Thu, Feb 21, 2013 at 3:20 PM, akhilesh chand
omakhileshch...@gmail.comwrote:
In server_X, 60 agnet are login and call comes from server_A and server_B
which is connected with
On Thu, Feb 21, 2013 at 12:37 PM, akhilesh chand
omakhileshch...@gmail.comwrote:
hello all,
i have two asterisk server for call transfer and one more asterisk server
for agent login(server_X) where agent take the call.
server_A and server_B
server_A is connected with pri and configure
hello all,
i have two asterisk server for call transfer and one more asterisk server
for agent login(server_X) where agent take the call.
server_A and server_B
server_A is connected with pri and configure with 60 channel for call
transfer into server_X
server_B is connected with pri and
We are bumping into an interesting issue. We have Dynamically created
parking lots. When the lots were created they had five positions when the
customer wants them changed to say 10 position. We update our database but
since they are already created the new range for PARKINGDYNPOS is being
Dumb question - you did module parkandannounce reload?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, March 19, 2012 8:25 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Remove
Relatively new to Asterisk and SIP and am trying to run a proof of
concept using Asterisk to make an outbound call through an Audiocodes
gateway via SIP using Asterisk version 1.6.1.12. The specific
requirements of the gateway in the configuration I am trying to use
specify that the Name part
On Wed, May 4, 2011 at 12:10 PM, John Hablitzel jjblitz...@gmail.comwrote:
Relatively new to Asterisk and SIP and am trying to run a proof of concept
using Asterisk to make an outbound call through an Audiocodes gateway via
SIP using Asterisk version 1.6.1.12. The specific requirements of the
On 5/4/2011 4:04 PM, Warren Selby wrote:
On Wed, May 4, 2011 at 12:10 PM, John Hablitzel jjblitz...@gmail.com
mailto:jjblitz...@gmail.com wrote:
Relatively new to Asterisk and SIP and am trying to run a proof of
concept using Asterisk to make an outbound call through an
Audiocodes
On 5/4/11 7:10 PM, John Hablitzel wrote:
exten = xxx,n,Set(CALLERID(name)=)
I'd either leave the name alone or do te following (haven't had the need
for removing it):
exten = xxx,n,Set(CALLERID(name)=)
--
Andreas Sikkema
--
Yup I just copy and paste to it but it shown not a known channel.
On Thu, Aug 28, 2008 at 6:47 PM, Steven Howes [EMAIL PROTECTED] wrote:
Did you tab complete it to make sure it was right?
On 28 Aug 2008, at 11:39, Rilawich Ango wrote:
I got the message below after I issue the soft
Hmm I would rather tab complete than copy and paste. It might do some
sort of escaping. Is there no time of day when all the queues are empty?
S
On 29 Aug 2008, at 07:14, Rilawich Ango wrote:
Yup I just copy and paste to it but it shown not a known channel.
On Thu, Aug 28, 2008 at
Hi ,
Actually, there are 3 queues in the server. Only one queue (2700)
has problem. I want to reset or remove the caller only in 2700
without affecting other queues or calls. Does it work for my case?
On Thu, Aug 28, 2008 at 11:49 AM, Andy Kuo [EMAIL PROTECTED] wrote:
Hi,
Try CLI soft
Try CLI soft hangup Local.
On 28 Aug 2008, at 09:01, Rilawich Ango wrote:
Hi ,
Actually, there are 3 queues in the server. Only one queue (2700)
has problem. I want to reset or remove the caller only in 2700
without affecting other queues or calls. Does it work for my case?
On
I got the message below after I issue the soft hangup.
sip01*CLI soft hangup Local/[EMAIL PROTECTED],2
Local/[EMAIL PROTECTED],2 is not a known channel
Any other way to kill the call without affecting other queues and calls?
On Thu, Aug 28, 2008 at 4:09 PM, Steven Howes [EMAIL PROTECTED] wrote:
Did you tab complete it to make sure it was right?
On 28 Aug 2008, at 11:39, Rilawich Ango wrote:
I got the message below after I issue the soft hangup.
sip01*CLI soft hangup Local/[EMAIL PROTECTED],2
Local/[EMAIL PROTECTED],2 is not a known channel
Any other way to kill the call without
Hi all,
I have the following queue and members. I found that there is a
call stuck in the queue so other call can't enter the queue. I want
to know whether we can remove the call (by CLI) to free the queue.
ango
2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
holdtime),
2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
holdtime), W:0, C:134, A:48, SL:88.8% within 120s
Members:
Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
Hi,
Try CLI soft hangup Local.
Andy
On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
I have the following queue and members. I found that there is a
call stuck in the queue so other call can't enter the queue. I want
to know whether we can remove the call (by CLI) to
Hi, I would like remove a Digium TDM2400P from Asterisk (version 1.2.13) box
but when I remove card from the PC after reboot Asterisk not started
correctly.
On box now with TDM Card I have:
[EMAIL PROTECTED]:~# lsmod
Module Size Used by
zaphfc
Let me quote oej:
Make sure that you remove unnecessary text when you reply
I don't need messages to tell me *5* times about Astricon,
who provides the bandwidth and how to unsubscribe.
I'm sure this has been posted a dozen times but please
http://learn.to/quote
Thanks,
Philipp Kempgen
--
On Sep 5, 2007, at 7:42 PM, Philipp Kempgen wrote:
I don't need messages to tell me *5* times about Astricon,
who provides the bandwidth and how to unsubscribe.
You sure about that unsubscribe part? People do seem to miss it :P
/b
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Brian West wrote:
On Sep 5, 2007, at 7:42 PM, Philipp Kempgen wrote:
I don't need messages to tell me *5* times about Astricon,
who provides the bandwidth and how to unsubscribe.
You sure about that unsubscribe part? People do seem to miss it :P
Good point. :)
Regards,
Philipp
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, June 09, 2007 5:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to tell what codec is used for each end of a
call MD110-H323-SIP
Hi.
Ihave an older version (0.9.0)of
Asterisk on my linux box. Do I need to remove it before I install version 1.2?
How do I remove it? Does Asterisk make file contain the uninstall process? Or I
have to manully remove all the directory structure.
Gary
On 11/18/05, gc [EMAIL PROTECTED] wrote:
I have an older version (0.9.0) of Asterisk on my linux box. Do I need to
remove it before I install version 1.2? How do I remove it? Does Asterisk
make file contain the uninstall process? Or I have to manully remove all the
directory structure.
It
Is there a command to remove completely asterisk?
I want clean the server before the installation of 1.2 version.
Matteo
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make uninstall?
Matteo Piazza wrote:
Is there a command to remove completely asterisk?
I want clean the server before the installation of 1.2 version.
Matteo
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rm -rf for the following dirs (taken from
http://www.oinko.net/astrecipes/index.php?n=93 ):
/etc/asterisk
/var/log/asterisk
/var/lib/asterisk
/var/spool/asterisk
/usr/lib/asterisk
any other?
l.
On Tue, 15 Nov 2005 11:02:08 +0100, Matteo Piazza
[EMAIL PROTECTED] wrote:
Is there a
Are you sure that with make unistall all asterisk's three is cancelled?
Martin Vit wrote:
make uninstall?
Matteo Piazza wrote:
Is there a command to remove completely asterisk?
I want clean the server before the installation of 1.2 version.
Matteo
Ive got a small office setup with a TDM400 and 3 FXO
cards. Id like to take away the ability of the 3rd FXO to
receive calls (as this line runs through our old-fashioned fax machine) BUT
still be able to use it for outbound calls. With our original, and very basic
PBX we could modify the
On Mon, 2005-03-28 at 12:02 -0500, Bob Sowers wrote:
Ive got a small office setup with a TDM400 and 3 FXO cards. Id like
to take away the ability of the 3rd FXO to receive calls (as this line
runs through our old-fashioned fax machine) BUT still be able to use
it for outbound calls. With our
Richard wrote:
Hi,
I have some stall calls in *. Is there any command to clear the call?
1. Yes, in the console type soft hangup and then press the tab key to
auto complete to the channel you want.
2. Usually the only need for this is incorrect set-up of
indications.conf etc. Do you want to
Title: remove channels
Hi,
I have some stall calls in *. Is there any command to clear the call?
Thanks,
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To UNSUBSCRIBE or update options
rm -rf reggie
On Tue, 2003-12-02 at 22:35, reginald huey wrote:
Please
Remove me from the list
Reggie
Reginald Huey
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Free Pop-Up Blocker - Get it now
Please
Remove me from the list
ReggieReginald Huey
Do you Yahoo!?
Free Pop-Up Blocker - Get it now
On Tue, 2003-12-02 at 16:35, reginald huey wrote:
Please
Remove me from the list
There should be a link to unsubscribe yourself at the bottom of this
message. Read ANY other message, and they are there. This is a
self-serve mailing list.
--
Leif Madsen [EMAIL PROTECTED]
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 9:59 AM
Subject: Re: [Asterisk-Users] X100P and PSTN caller id
Gary sais some time ago:
Have you got the number of rings for caller-id set right ??
its in chan_zap somewhere from
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 9:12 AM
Subject: Re: [Asterisk-Users] X100P and PSTN caller id
As everything is under software control in X100P, then for sure must be a
way to solve it...
Even in UK they are several
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 10:19 AM
Subject: Re: [Asterisk-Users] Asterisk, IAX and NAT issue
Hi Simon,
Is your firewall redirecting incoming connections on n.n.n.n:5036 to
the Internal Asterisk instance?
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 10:25 AM
Subject: Re: [Asterisk-Users] Asterisk, IAX and NAT issue
SOLVE IT!!!
Just need to comment the line:
permit=0.0.0.0/0.0.0.0
for the pspbx IAX user defined in iax.conf file
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 10:26 AM
Subject: Re: [Asterisk-Users] X100P and PSTN caller id
I think the problem is more fundamental than this. The state machine in
the X100P assumes that nothing at
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 10:22 AM
Subject: Re: [Asterisk-Users] Asterisk, IAX and NAT issue
I may be wrong here but wouldn't a register line in the iax.conf on the
systems behind the NAT keep the port
- Original Message -
From: Simon J Mudd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 9:47 AM
Subject: Re: [Asterisk-Users] Asterisk, IAX and NAT issue
[EMAIL PROTECTED] (Dan) writes:
[snip]
Now, on the other PBX I have the following in extensions.conf
- Original Message -
From: K. C. Li [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 9:15 AM
Subject: Re: [Asterisk-Users] X100P and PSTN caller id
On Thu, 26 Jun 2003, Andy Powell wrote:
I don;'t know what else to try, I've had callerid turned on here but
- Original Message -
From: Stephen Davies [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 8:12 AM
Subject: Re: [Asterisk-Users] X100P and PSTN caller id
On Thu, 26 Jun 2003, Dan wrote:
There is nobody with an X100P in Europe having this issue related to the
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 9:07 AM
Subject: Re: [Asterisk-Users] X100P and PSTN caller id
Hi Andy,
Did you get something as callerid?
I get allways my own PSTN number.
Talking with a friend who sells CallerID
can someone remove this jack... already please
At 01:25 PM 6/26/2003 -0300, you wrote:
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 10:22 AM
Subject: Re: [Asterisk-Users] Asterisk, IAX and NAT issue
I may be wrong here but
Spaming the list with multiple inane REMOVE messages will not get you
removed from the list. There is a link to the mail list interface at the
bottom of every single email you receive from the list. If you are unable to
figure out how to unsubscribe that way, then *ask* the list. Quit spaming
I wonder HOW MANY TIMES YOU NEED TO SEND THE SAME MAIL TO SEE THAT YOU
CANNOT UNSUBSCRIBE IN THIS WAY
D
- Original Message -
From: cisb [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 7:25 PM
Subject: [Asterisk-Users] REMOVE
- Original Message
He has been officially removed and blocked. Have a nice day! :)
On Thu, 26 Jun 2003, Jon Pounder wrote:
can someone remove this jack... already please
At 01:25 PM 6/26/2003 -0300, you wrote:
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
--- cisb [EMAIL PROTECTED] wrote:
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 10:19 AM
Subject: Re: [Asterisk-Users] Asterisk, IAX and NAT
issue
Hi Simon,
Is your firewall redirecting incoming
connections on
--- Gary [EMAIL PROTECTED] wrote:
I just had a thought, but haven't tried it yet
we are able to include one conf file into another...
this leads to the thought can we use the same file
to include into
iax.conf as well as sif.conf ?
does the various differences actually have an
No! You're outnumbered and trapped!
On Wed, 28 May 2003 10:22:33 -0700 (PDT)
Patrick Tabor [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
--- Gary [EMAIL PROTECTED] wrote:
I just had a thought, but haven't tried it yet
we are able to
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