On Wednesday 25 May 2022 at 16:54:43, aster...@phreaknet.org wrote:
> On 5/25/2022 10:41 AM, Antony Stone wrote:
> > On Wednesday 25 May 2022 at 15:27:38, aster...@phreaknet.org wrote:
> >>
> >> If I want to log something from the dialplan, I generally send it to a
> >> custom log level, as
On Wednesday 25 May 2022 at 15:27:38, aster...@phreaknet.org wrote:
> On 5/25/2022 8:11 AM, Antony Stone wrote:
> > On Tuesday 24 May 2022 at 01:12:46, Kevin Harwell wrote:
> >> So this turned out more complicated than I originally thought!
> >
> > Wow, thank you very much for:
> >
> > a) such
On Tuesday 24 May 2022 at 01:12:46, Kevin Harwell wrote:
> So this turned out more complicated than I originally thought!
Wow, thank you very much for:
a) such a comprehensive answer
b) confirming my findings
c) most of all, working out why and how all this stuff works (or, perhaps,
So this turned out more complicated than I originally thought!
My expectation:
Verbosity gets logged using an "at least" check against the current
system's verbose level, which if passed subsequently gets checked against
the logging channel's verbose level. Thus only verbose messages with a
Hi.
Does no-one else know either? I thought this was a simple question, and it
was just me being unable to find the appropriate documentation to explain how
these logging levels work.
Please, can anyone help?
On Friday 20 May 2022 at 15:33:45, Antony Stone wrote:
> Hi.
>
> I'm trying to
On Friday 20 May 2022 at 15:33:45, Antony Stone wrote:
> Hi.
>
> I'm trying to use different logging verbosity levels to get dialplan output
> into different log files, and there's clearly something I haven't
> understood about how Asterisk does this...
>
>
> I have the following in
Hi.
I'm trying to use different logging verbosity levels to get dialplan output
into different log files, and there's clearly something I haven't understood
about how Asterisk does this...
I have the following in /etc/asterisk/logger.conf:
[logfiles]
logtest.verbose.0 => verbose(0)
| LG Salzburg
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp
Gesendet: Donnerstag, 11. Januar 2018 16:35
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Logging ARI debug
On Thu, Jan 11, 2018, at 11:30 AM, Floimair Florian wrote:
> Hi there!
>
> Is there any way I can turn on debug for ARI and sending the output to a
> separate log file?
> So far I have only been able to turn on ARI debugging in the console
> which results in the debug output being logged in
Hi there!
Is there any way I can turn on debug for ARI and sending the output to a
separate log file?
So far I have only been able to turn on ARI debugging in the console which
results in the debug output being logged in /var/log/asterisk/messages
I would love to have ARI debug log messages in
Hi!
I am seeing a lot of warnings of these types:
res_pjsip_registrar.c: AOR '31' not found for endpoint 'anonymous'
I am guessing these are coming from a scanner trying to scan for the
extensions on the asterisk server.
Is there any way to print the IP address of the endpoint trying to
oun...@lists.digium.com] On Behalf Of er ic
Sent: Thursday, December 31, 2015 7:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Logging to CDR after call file Not Answered
If the end user does not pick up the phone, is there a way to log to the CDR
gards;
>
> John
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *er ic
> *Sent:* Thursday, December 31, 2015 7:05 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-user
If the end user does not pick up the phone, is there a way to log to the
CDR about the call file failing?
/var/log/asterisk/messages does show a NOTICE message
[2015-12-31 06:58:46] NOTICE[28059] pbx_spool.c: Call failed to go through,
reason (3) Remote end Ringing
[2015-12-31 06:58:46]
: [asterisk-users] Logging to CDR after call file Not Answered
I dont have this info in my cdr.conf.sample
Nor do I think this is what I need.
I believe that will log unanswered calls inside a dialplan. Meaning, if I call
directly from my extension to another extension/number, if they do
Hi again!
I just noticed, that my Asterisk (running on an OpenWRT-Switch) writes
the logs using GMT...
On the Switch the time is right configured and a date says me the
current LOCAL time.
I didn't found in logger.conf or other file an option to set the timezone.
Can someone help me?
Am 5. Juni 2015 16:29:21 MESZ, schrieb Luca Bertoncello lucab...@lucabert.de:
Hi again!
I just noticed, that my Asterisk (running on an OpenWRT-Switch) writes
the logs using GMT...
On the Switch the time is right configured and a date says me the
current LOCAL time.
I didn't found in
On Wed, 2013-04-10 at 11:06 -0700, Carlos Alvarez wrote:
On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards
asterisk@sedwards.com wrote:
dumpcap can capture all of the SIP (and RTP) packets into a
series of files without a huge performance hit.
Is anyone using something to log SIP results (connected/not, latency) that
they really like? We do some logging using simple scripts writing the
results of sip show peers to a text file if customers report issues, but it
would be nice to have a tool that logs all the time and lets us do some
On Wed, 10 Apr 2013, Carlos Alvarez wrote:
Is anyone using something to log SIP results (connected/not, latency)
that they really like? We do some logging using simple scripts writing
the results of sip show peers to a text file if customers report issues,
but it would be nice to have a tool
On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards
asterisk@sedwards.comwrote:
dumpcap can capture all of the SIP (and RTP) packets into a series of
files without a huge performance hit.
A cron job can pbzip2 the files and delete if over x days old.
That's completely different. We
http://www.artifact-software.com/?page_id=1666
Would this help?
Put a JasperReport graph or two in a report step.
Ron
On 10/04/2013 2:02 PM, Steve Edwards wrote:
On Wed, 10 Apr 2013, Carlos Alvarez wrote:
Is anyone using something to log SIP results (connected/not, latency)
that they really
On On Wed, 10 Apr 2013, Carlos Alvarez wrote:
Is anyone using something to log SIP results (connected/not, latency) that
they really like? We do some logging using simple scripts writing the
results of sip show peers to a text file if customers report issues, but it
would be nice to
Un-top-posting and de-crufting...
On Mon, Nov 14, 2011 at 3:19 AM, Tristram Cheer wrote:
I'm using DumpChan(1001)...
I would like to dump this output to a file specifically for DumpChan...
On Mon, 14 Nov 2011, Warren Selby wrote:
If you call DumpChan from an AGI you should be able to
Hi All,
Hopefully this is considered on-topic for this list.
I'm using DumpChan(1001) in a Macro called debug in order to debug issues
within the dialplan, I would like to dump this output to a file
specifically for DumpChan output but I'm having issues with figuring out
how to do this under
Hello,
Reading about the application DumpChan() shows this:
[Synopsis]
Dump Info About The Calling Channel.
[Description]
Displays information on channel and listing of all channel variables. If
level is specified, output is only displayed when the verbose level is
currently set to that number
Maybe this can help
http://www.russellbryant.net/blog/2011/03/04/debugging-the-asterisk-dialplan-with-verbose/
2011/11/14 Tristram Cheer trist...@tristramcheer.com
Hi All,
Hopefully this is considered on-topic for this list.
I'm using DumpChan(1001) in a Macro called debug in order to debug
Hi Sammy,
It's a good start, Atleast being split it is handy, Ideally I'd to be able
to spit DumpChan output direct to JabberSend or func_ODBC but I fear this
will require someone who know's C to alter the module. I think i'm going to
have to just use JabberSend for each variable I use and the
If you call DumpChan from an AGI you should be able to read the response
programmatically and then dump the data into a database. Cleans up your
dialplan but requires some scripting or programming knowledge (php, perl, bash
or even C) in order to write the AGI.
Thanks,
--Warren Selby, dCAP
We have queuemetrics, qloaderd and mysql running on our asterisk server in
order to streamline call reporting. Now in order to get internal call logs,
I need to get thirdlane Master.csv file. I see there is an option in
thirdlane to import this into mysql which would make it easier to work with
Quoting Matt Riddell li...@venturevoip.com:
Maybe you could do:
Set(CDR(userfield)=${CALLERID(num)})
Before dialing SIP/1000
That looks so simple -- and it actually works! -- although exactly not
in the way that I was expecting. Instead of replacing the contents of
one of the existing
Hi folks,
My v1.6 Asterisk system logs all Call Detail Records to a PostgreSQL
database, including those handled by the Privacy Manager.
Unfortunately, even though I can use the CLI to see the information
being submitted by anonymous callers to satisfy the demands of the the
Privacy
On 1/09/10 11:27 AM, Jaap Winius wrote:
exten = jw,1,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jw,n(true),Set(CALLERID(num)=)
exten = jw,n(false),NoOp()
exten = jw,n,Verbose(-- CID is${CALLERID(num)})
exten =
Can asterisk log the registration date/time in a database? Is there a
standard option to do this?
I know it being logged in the asterisks 'full' (debug) log and we are
probably able to script something with the API interface but there might
be somewhat easier if there is a option to make
I am not aware with any logging option, but If you want to monitor
registration status. Asterisk Realitme can help you.
For example if you are using Realtime SIP configuration then you can find
registration info at regserver and regseconds fields
On Sun, Jul 18, 2010 at 7:28 PM, Bram Bosboom
Happy Friday everyone,
Is there a way to log the negotiated codec that was used for each call
in CDR or in a separate log file?
This is for SIP-based calls, if that matters.
Perhaps there is some variable that can be queried as part of the
dialing script;
Or is it possible to grab the codec
Hi!
Is there a way to log the negotiated codec that was used for each call
in CDR or in a separate log file?
Use CHANNEL(audionativeformat) - and do the same with the help of the M
option to Dial() for the remote call leg. Store that info in the CDR
userfield, or create your own field if you
Hello list,
I noticed today that the last logfiles dates 3 days ago !
The logfiles are rotated every night. The logfiles of 2 days ago, 1 day
ago and today are empty !
vps*CLI module show like logger
Module Description
Use Count
0
On 14/06/10 11:04, Jonas Kellens wrote:
Hello list,
I noticed today that the last logfiles dates 3 days ago !
The logfiles are rotated every night. The logfiles of 2 days ago, 1
day ago and today are empty !
vps*CLI module show like logger
Module
Description
I don't think it's a disk space issue :
bash-3.2# df -h
FilesystemSize Used Avail Use% Mounted on
/dev/sda1 25G 5.0G 19G 21% /
tmpfs 256M 0 256M 0% /dev/shm
bash-3.2# df -h /var/log/
FilesystemSize Used Avail Use% Mounted on
Hello,
I'm using Asterisk 1.6.2.0-beta3 with asterisk-addons-1.6.2.0-rc1 to
write cdr into mysql. I followed:
http://www.spiration.co.uk/post/1327/asterisk-addons%20setting%20up%20mysql%20cdr%20for%20Asterisk,
and it is mostly working however CLID is not written to mysql,
although it
Greetings all, I have an interesting problem I am trying to work around.
I currently have 2 * servers running in separate offices, using IAX2 to trunk
between them, and queues in our main office. I'll call them Office_A and
Office_B. I use Polycom 501s with a primary and secondary server, the
Hi, I'm in the process for setting up an asterisk server for four
organisations sharing a SIP trunk. In order to split the costs according
to usage, it would be nice to log all incoming, outgoing and missed
calls.
Is there a simple way of doing this, preferrably in a database? Perhaps
someone has
Hi everybody
Did anybody by any chance ever work out how to log in and out agents on
Asterisk 6+?
I used to have it working perfect in Asterisk 1.2 but since I upgraded to 6
the agent login functions are gone and the readme file that came with it
made no sense to me.
I noticed somebody on the
Agentcallbacklogin was deprecated in Asterisk 1.4 and eliminated from
1.6 so you now need to use Dynamic Agents. Although they claim that is
is simple enough to replace that functionality with dial plan code I
have yet to see a one line example that replaces everything the
Discussion
asterisk-users@lists.digium.com
Date: Fri, 15 May 2009 22:40:55 +0100
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Logging In / Out Agents on Asterisk 6 ???
Hi everybody
Did anybody by any chance ever work out how to log in and out agents on
Asterisk 6+?
I used to have
On Tue, 2009-04-07 at 15:21 +0200, Marco Sambo wrote:
Hi Enrico,
I do that by modifying logger.conf
[logfiles]
logpro = notice,warning,error,debug,verbose
and modifying asterisk.conf
[directories]
astetcdir = /etc/asterisk
astmoddir = /usr/lib/asterisk/modules
astvarlibdir =
Hi all, in witch way can I put in a log file the asterisk console?
I have tried with some settings in file logger.conf but the log not
contain the same debug that I can see with asterisk -rvvv.
I need it in debugging purpose for tracking some bug.
Thanks Enrico.
smime.p7s
Description: S/MIME
2009/4/7 Enrico Pasqualotto enr...@pasqualotto.org:
Hi all, in witch way can I put in a log file the asterisk console?
I have tried with some settings in file logger.conf but the log not
contain the same debug that I can see with asterisk -rvvv.
I need it in debugging purpose for tracking some
: Tuesday, April 07, 2009 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Logging Asterisk console
Hi all, in witch way can I put in a log file the asterisk console?
I have tried with some settings in file logger.conf but the log not
contain the same debug
Hi Enrico,
I do that by modifying logger.conf
[logfiles]
logpro = notice,warning,error,debug,verbose
and modifying asterisk.conf
[directories]
astetcdir = /etc/asterisk
astmoddir = /usr/lib/asterisk/modules
astvarlibdir = /var/lib/asterisk
astdatadir = /var/lib/asterisk
astagidir =
and what about add a custome field or setup a variable on outgoing calls and
use the common cdr and then filtering by that field.
David
2009/1/24 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Friday 23 January 2009 18:22:16 Pascal Bruno wrote:
Is it possible to log just the outgoing
That is a good idea too, where would I configure asterisk to log the channel
status on that custom field?
On Sat, Jan 24, 2009 at 8:27 AM, David fire ddf...@gmail.com wrote:
and what about add a custome field or setup a variable on outgoing calls
and use the common cdr and then filtering
Is it possible to log just the outgoing calls using cdr_odbc into a custom
mysql database table?
my table will look like this:
| call_status |
|-- --|
| · id |
| · destination |
| · status |
||
I just need to store the
On Friday 23 January 2009 18:22:16 Pascal Bruno wrote:
Is it possible to log just the outgoing calls using cdr_odbc into a custom
mysql database table?
my table will look like this:
| call_status |
|-- --|
| · id |
| · destination |
| ·
Is it possible in Asterisk 1.4 to log by somehow the estimated roundtrip
time (RTT) between server and some peer, which Asterisk computes based on
the sending of OPTIONS and the receiving of the responses to those OPTIONS?
Regards,
Ricardo Carvalho.
___
Is it possible to implement in the Asterisk dialplan some way to
authenticate a user with a dialed passcode which opens session that stays
active enabling the user to make and receive calls, until the user logs off
with another dialed passcode?
I am aware of the Asterisk application
On Thu, 6 Dec 2007, Ricardo Carvalho wrote:
Is it possible to implement in the Asterisk dialplan some way to
authenticate a user with a dialed passcode which opens session that stays
active enabling the user to make and receive calls, until the user logs off
with another dialed passcode?
The
Thanks Gordon, I'll give it a try with astDB.
Regards,
Ricardo Carvalho.
___
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asterisk-users mailing list
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Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can
pick up CallerID. How can I redirect the log output of asterisk to
/dev/ttyS0 or /dev/console?
Many thanks,
Neil
safeharbour IT Ltd
Your IT Department
fax: 0845 867 2891
mob: 07812 114784
voip: [EMAIL PROTECTED]
Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can
pick up CallerID. How can I redirect the log output of asterisk to
/dev/ttyS0 or /dev/console?
I think you might be better off with a System() call in your dial plan such as:
System(echo ${CALLERIDNUM} /dev/ttyS0)
On Thu, Feb 01, 2007 at 03:32:22PM -, Neil Tancock wrote:
Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can
pick up CallerID. How can I redirect the log output of asterisk to
/dev/ttyS0 or /dev/console?
If you can't simply put /dev/ttyS0 or /dev/console as a log
If you really need it, save it on a remote server (nfs or so), that
should minimize the problems
Zoa
---
www.asteriskguru.com
Simone Cittadini wrote:
Moises Silva ha scritto:
How important is the impact i could have if I have a single entry log
file in /etc/asterisk/logger.conf wich loggs
How important is the impact i could have if I have a single entry log
file in /etc/asterisk/logger.conf wich loggs everything, even debug
level. This is sometimes important to us because it helps us to make a
track of the issues some times we have with the system. I just want to
know if there is a
Moises Silva ha scritto:
How important is the impact i could have if I have a single entry log
file in /etc/asterisk/logger.conf wich loggs everything, even debug
level. This is sometimes important to us because it helps us to make a
track of the issues some times we have with the system. I
i have registered on teliax service and i m using a hathway internet connection.with X-lite phone it is not logging in .it says login timed out whereas the phone with same X-lite and service settings i m getting logged in all other internet connection and the phone also works perfectly. the
Had not seen a response on the following question - wondering if
anyone may have any insight on this?
Original Question-
Is there a way to log SIP response codes without enabling verbose
logging? Reason being is that from time to time I see a call fail on
our primary provider and
Is there a way to log SIP response codes without enabling verbose
logging? Reason being is that from time to time I see a call fail on
our primary provider and roll-over to our backup providers. If I
happen to catch it on the console I can see the code 484 or similar.
It would really help in
I have searched and searched, but been unable to find a
method to log agents in and out other than through the phone and
AgentCallbackLogin or AgentLogin. Anyone know if there is another way? I have
looked through the manager API but was unable to find an alternate way to do
this.
i want the to find out the delay between two events:
1) the instance a call is recieved on an FXO port and the
2) the instance a SIP INVITE is sent to the SIP destination.
i need to attach timestamps to the events before logging them. How can i:
1) log `ALL' events.
2) Attach timestamps to them?
hi
Is it possible to log the codec used in CDR?
Today, I have an AGI script logging the ipaddr of the sip client to the
userfield. how can I find the current codec as reported on the console:
-- Format for call is g726
roy
___
Asterisk-Users mailing
-Original Message-
From: Shad Mortazavi [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 9:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Logging into Multiple Call Queues on two *
Servers and Voice Mail option.
Dear All,
I have two objectives that I need to meet;
1. I
Title: Logging into Multiple Call Queues on two * Servers and Voice Mail option.
Dear All,
I have two objectives that I need to meet;
1. I need to be able to log into two separate call queues on two different Asterisk servers, servicing two data centers. I seem to have problems
Hi,
is there a way to log errors to syslog ?
Currently logger.conf is logging everything in /var/log/asterisk/messages
but I would like to see the errors in my /dev/tty12 console on my server.
[logfiles]
console = notice,warning,error
messages = notice,warning,error
Thanks,
Jean-Pierre Denis
As far as I know, there is not currently a way to do this. I've toyed
with the idea of adding support myself, but 1) its probably a bit over
my head and 2) I really don't have that much *spare* time anyway.
But its a feature I'd love to see!
Anyone want to collaborate on this?
-BAK
On Thu,
Yes, you can:
asterisk -vvvgcn|tee /tmp/log
regards
Martin
On Thu, 27 Feb 2003, Roy Sigurd Karlsbakk wrote:
hi
can I log all console output while having console access as with
asterisk -vvvgc
?
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801
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