Re: [asterisk-users] Logging different verbosity levels

[Antony Stone]

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 opposed to one of the built in ones.
> > 
> > How are you doing this?
> 
> See the custom_levels option in the sample logger.conf[1].
> You'll need a version of Asterisk from the last ~7 months or so, I think.

Ah.  I use the Debian packaged version, which for the current stable Debian 
release is Asterisk 16.16.1 dated Feb 2021.

custom_levels definitely isn't in there.

> It does sound like using custom levels might be better than your use case.
> You can also set up custom log files to log only the log levels you want to
> create the different "views" you want.

Indeed - thanks for the suggestion - looks like a nice feature.


Antony.

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Re: [asterisk-users] Logging different verbosity levels

[Antony Stone]

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 a comprehensive answer
> > 
> > b) confirming my findings
> > 
> > c) most of all, working out why and how all this stuff works (or,
> > perhaps, doesn't).
> > 
> > I wonder that nobody has discovered this before - do people not want
> > selective logging levels in their dialplans?
> 
> If I want to log something from the dialplan, I generally send it to a
> custom log level, as opposed to one of the built in ones.

How are you doing this?

> That way, it's not combined with a bunch of other stuff from Asterisk itself
> that I generally don't want. This also allows filtering on specific custom log
> levels. Verbose gets used for so much that if something got logged there
> it would just get lost.

Indeed - that was precisely my reasoning for wanting to use the different 
Verbose(N,message) levels and corresponding log files - so that I could keep 
specific things separate from each other.


Antony.

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Re: [asterisk-users] Logging different verbosity levels

[Antony Stone]

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, 
doesn't).

I wonder that nobody has discovered this before - do people not want selective 
logging levels in their dialplans?

Hm.


Thanks again,

Antony.

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Re: [asterisk-users] Logging different verbosity levels

[Kevin Harwell]

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
level less than or equal to the system's level AND the channel's level
would be logged to that channel. For example given a system verbose level
of 3, and your setup then the following should occur:

same => n,Verbose(0,Test message verbosity 0) ; gets logged to
logtest.verbose.3, logtest.verbose.2, logtest.verbose.1, and
logtest.verbose.0
same => n,Verbose(1,Test message verbosity 1) ; gets logged to
logtest.verbose.3, logtest.verbose.2, and logtest.verbose.1
same => n,Verbose(2,Test message verbosity 2) ; gets logged to
logtest.verbose.3 and logtest.verbose.2
same => n,Verbose(3,Test message verbosity 3) ; gets logged to
logtest.verbose.3
same => n,Verbose(4,Test message verbosity 4) ; nothing logged
same => n,Verbose(5,Test message verbosity 5) ; nothing logged
same => n,Verbose(6,Test message verbosity 6) ; nothing logged
same => n,Verbose(7,Test message verbosity 7) ; nothing logged
same => n,Verbose(8,Test message verbosity 8) ; nothing logged
same => n,Verbose(9,Test message verbosity 9) ; nothing logged

Reality:

What you saw in your output. As to why? Well it's a bit of a mess and it's
been that way for a while it seems. To start the app_verbose application
gets capped at verbose level 4. Meaning the following:

same => n,Verbose(7,Test message verbosity 7)

Essentially gets changed to:

same => n,Verbose(4,Test message verbosity 7)

Next, Asterisk sets the global verbose logging level according to the
following order with each level potentially overriding if greater than the
preceding:

1) core verbose level (set in asterisk.conf)
2) console level (set via system CLI, or asterisk CLI)
3) channel log level (set in logger.conf, e.g. verbose(9))

This means in asterisk.conf if you set verbose=2, and then on the CLI set
it to 3, and then for any channel in logger.conf specify verbose(9) then
the final level will be 9! The levels too are tracked via different
variables so given the preceding example the following will be output:

*CLI> core show settings
...
Root console verbosity:  2
...

Despite the actual level being 9.

Lastly, when the logger goes to output the actual log message to a channel
it does indeed first check if the channel's verbosity level is greater than
some "level" variable. But guess what? This variable is always equal to 0.
Thus as long as a channel's level is > 0 it always passes and the message
is output.

So you can see in your setup the final system level is indeed 9, and all
app_verbose messages being output are forced to 4 or less. Since the system
level is 9, and the channel's verbosity level is greater than zero then all
messages are output to all files.

Hope that makes sense!

Given all that, in my opinion there seems to be at least one or more bugs
with regards to verbose logging. Please file an issue at
https://issues.asterisk.org/. Feel free to copy/paste what's here as part
of the issue description.


On Mon, May 23, 2022 at 3:29 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> 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 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)
> > logtest.verbose.1 => verbose(1)
> > logtest.verbose.2 => verbose(2)
> > logtest.verbose.3 => verbose(3)
> > logtest.verbose.4 => verbose(4)
> > logtest.verbose.5 => verbose(5)
> > logtest.verbose.6 => verbose(6)
> > logtest.verbose.7 => verbose(7)
> > logtest.verbose.8 => verbose(8)
> > logtest.verbose.9 => verbose(9)
> >
> > I then put the following at a particular point in my dialplan:
> >
> > same => n,Verbose(0,Test message verbosity 0)
> > same => n,Verbose(1,Test message verbosity 1)
> > same => n,Verbose(2,Test message verbosity 2)
> > same => n,Verbose(3,Test message verbosity 3)
> > same => n,Verbose(4,Test message verbosity 4)
> > same => n,Verbose(5,Test message verbosity 5)
> > same => n,Verbose(6,Test message verbosity 6)
> > same => n,Verbose(7,Test message verbosity 7)
> > same => n,Verbose(8,Test message verbosity 8)
> > same => n,Verbose(9,Test message verbosity 9)
> >
> > I was expecting to get each message output into the respective filename,
> > but instead I got 10 files with the expected filenames, and all
> containing
> > every test message, no matter 

Re: [asterisk-users] Logging different verbosity levels

[Antony Stone]

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 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)
> logtest.verbose.1 => verbose(1)
> logtest.verbose.2 => verbose(2)
> logtest.verbose.3 => verbose(3)
> logtest.verbose.4 => verbose(4)
> logtest.verbose.5 => verbose(5)
> logtest.verbose.6 => verbose(6)
> logtest.verbose.7 => verbose(7)
> logtest.verbose.8 => verbose(8)
> logtest.verbose.9 => verbose(9)
> 
> I then put the following at a particular point in my dialplan:
> 
> same => n,Verbose(0,Test message verbosity 0)
> same => n,Verbose(1,Test message verbosity 1)
> same => n,Verbose(2,Test message verbosity 2)
> same => n,Verbose(3,Test message verbosity 3)
> same => n,Verbose(4,Test message verbosity 4)
> same => n,Verbose(5,Test message verbosity 5)
> same => n,Verbose(6,Test message verbosity 6)
> same => n,Verbose(7,Test message verbosity 7)
> same => n,Verbose(8,Test message verbosity 8)
> same => n,Verbose(9,Test message verbosity 9)
> 
> I was expecting to get each message output into the respective filename,
> but instead I got 10 files with the expected filenames, and all containing
> every test message, no matter which verbosity level it was output at.
> 
> I'm sure there's just something basic which I haven't understaood from
> 
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Verbose
> 
> and
> 
> https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration
> 
> 
> Can someone please show me what I'm missing, so that I can get each
> Verbose(N,Message) dialplan command to send its message into the log file
> numbered N?
> 
> 
> Thanks,
> 
> 
> Antony.

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Re: [asterisk-users] Logging different verbosity levels

[Antony Stone]

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 /etc/asterisk/logger.conf:

Hm, the formatting of this mail seems to have got somewhat mangled on its way 
through the list server, I think - I'll edit it and try again, just so it's 
easier for people to see what I did:

> [logfiles]
> logtest.verbose.0 => verbose(0)
> logtest.verbose.1 => verbose(1)
> logtest.verbose.2 => verbose(2)
> logtest.verbose.3 => verbose(3)
> logtest.verbose.4 => verbose(4)
> logtest.verbose.5 => verbose(5)
> logtest.verbose.6 => verbose(6)
> logtest.verbose.7 => verbose(7)
> logtest.verbose.8 => verbose(8)
> logtest.verbose.9 => verbose(9)
> 
> I then put the following at a particular point in my dialplan:
> 
> same => n,Verbose(0,Test message verbosity 0)
> same => n,Verbose(1,Test message verbosity 1)
> same => n,Verbose(2,Test message verbosity 2)
> same => n,Verbose(3,Test message verbosity 3)
> same => n,Verbose(4,Test message verbosity 4)
> same => n,Verbose(5,Test message verbosity 5)
> same => n,Verbose(6,Test message verbosity 6)
> same => n,Verbose(7,Test message verbosity 7)
> same => n,Verbose(8,Test message verbosity 8)
> same => n,Verbose(9,Test message verbosity 9)
> 
> I was expecting to get each message output into the respective filename,
> but instead I got 10 files with the expected filenames, and all containing
> every test message, no matter which verbosity level it was output at.
> 
> I'm sure there's just something basic which I haven't understaood from
> 
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Verbose
> 
> and
> 
> https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration
> 
> 
> Can someone please show me what I'm missing, so that I can get each
> Verbose(N,Message) dialplan command to send its message into the log file
> numbered N?
> 
> 
> Thanks,
> 
> 
> Antony.

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[asterisk-users] Logging different verbosity levels

[Antony Stone]

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) 




logtest.verbose.1 => verbose(1) 




logtest.verbose.2 => verbose(2) 




logtest.verbose.3 => verbose(3) 




logtest.verbose.4 => verbose(4) 




logtest.verbose.5 => verbose(5) 




logtest.verbose.6 => verbose(6) 




logtest.verbose.7 => verbose(7) 




logtest.verbose.8 => verbose(8) 




logtest.verbose.9 => verbose(9)

I then put the following at a particular point in my dialplan:

same => n,Verbose(0,Test message verbosity 0)   




same => n,Verbose(1,Test message verbosity 1)
same => n,Verbose(2,Test message verbosity 2)
same => n,Verbose(3,Test message verbosity 3)
same => n,Verbose(4,Test message verbosity 4)
same => n,Verbose(5,Test message verbosity 5)
same => n,Verbose(6,Test message verbosity 6)
same => n,Verbose(7,Test message verbosity 7)
same => n,Verbose(8,Test message verbosity 8)
same => n,Verbose(9,Test message verbosity 9)

I was expecting to get each message output into the respective filename, but 
instead I got 10 files with the expected filenames, and all containing every 
test message, no matter which verbosity level it was output at.

I'm sure there's just something basic which I haven't understaood from

https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Verbose

and

https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration


Can someone please show me what I'm missing, so that I can get each 
Verbose(N,Message) dialplan command to send its message into the log 

Re: [asterisk-users] Logging ARI debug messages

[Floimair Florian]

Thanks for the quick reply Joshua!

I might dig into this and try an implementation.

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
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Security and Communication by Commend

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-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 messages

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 /var/log/asterisk/ 
> messages
> 
> I would love to have ARI debug log messages in /var/log/asterisk/debug 
> or even better in it's own ari-debug file.

That is not something anyone has implemented as of this time. The messages 
themselves just get raised as normal verbose messages.

--
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Re: [asterisk-users] Logging ARI debug messages

[Joshua Colp]

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 /var/log/asterisk/
> messages
> 
> I would love to have ARI debug log messages in /var/log/asterisk/debug 
> or even better in it's own ari-debug file.

That is not something anyone has implemented as of this time. The messages 
themselves just get raised as normal verbose messages.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Logging ARI debug messages

[Floimair Florian]

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 /var/log/asterisk/debug or even 
better in it's own ari-debug file.



With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

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[asterisk-users] Logging endpoint IP address when PJSIP registration fails

[Sree Harsha Totakura]

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
register an extension using PJSIP in asterisk 13?  I can then configure
fail2ban to temporarily hinder the scan.

Regards,
Sree

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Re: [asterisk-users] Logging to CDR after call file Not Answered

[Tech Support]

Here you go… Found this in cdr.conf.sample.

 

; Define whether or not to log unanswered calls. Setting this to "yes" will

; report every attempt to ring a phone in dialing attempts, when it was not

; answered. For example, if you try to dial 3 extensions, and this option is 
"yes",

; you will get 3 CDR's, one for each phone that was rung. Default is "no". Some

; find this information horribly useless. Others find it very valuable. Note, 
in "yes"

; mode, you will see one CDR, with one of the call targets on one side, and the 
originating

; channel on the other, and then one CDR for each channel attempted. This may 
seem

; redundant, but cannot be helped.

;

; In brief, this option controls the reporting of unanswered calls which only 
have an A

; party. Calls which get offered to an outgoing line, but are unanswered, are 
still

; logged, and that is the intended behaviour. (It also results in some B side 
CDRs being

; output, as they have the B side channel as their source channel, and no 
destination

; channel.)

;unanswered = no

 

 

Regards;

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-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 
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] NOTICE[28059] pbx_spool.c: Queued call to SIP/102 expired 
without completion after 0 attempts

 

but I would rather not have to run a script and search the call log for those 
two strings, extrapolate the reason code and then compare it to my custom 
php/database driven scheduler. 

 

Is there a way to have it logged to the server?

 

Also, both chanunavail attempts and NA attempts both register the status as 
expried. Is this correct or should one of them be failed? I am trying to be 
able to distinguish the two conditions by looking at the archived message.

 

Thanks,

--E

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Re: [asterisk-users] Logging to CDR after call file Not Answered

[er ic]

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 not
pick up, it will log; as it currently does now.

With call files, it is not logging. There is no execution of a dialplan
inside my assigned context.

On Thu, Dec 31, 2015 at 9:29 AM, Tech Support <aster...@voipbusiness.us>
wrote:

> Here you go… Found this in cdr.conf.sample.
>
>
>
> ; Define whether or not to log unanswered calls. Setting this to "yes" will
>
> ; report every attempt to ring a phone in dialing attempts, when it was not
>
> ; answered. For example, if you try to dial 3 extensions, and this option
> is "yes",
>
> ; you will get 3 CDR's, one for each phone that was rung. Default is "no".
> Some
>
> ; find this information horribly useless. Others find it very valuable.
> Note, in "yes"
>
> ; mode, you will see one CDR, with one of the call targets on one side,
> and the originating
>
> ; channel on the other, and then one CDR for each channel attempted. This
> may seem
>
> ; redundant, but cannot be helped.
>
> ;
>
> ; In brief, this option controls the reporting of unanswered calls which
> only have an A
>
> ; party. Calls which get offered to an outgoing line, but are unanswered,
> are still
>
> ; logged, and that is the intended behaviour. (It also results in some B
> side CDRs being
>
> ; output, as they have the B side channel as their source channel, and no
> destination
>
> ; channel.)
>
> ;unanswered = no
>
>
>
>
>
> Regards;
>
> 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-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 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] NOTICE[28059] pbx_spool.c: Queued call to SIP/102
> expired without completion after 0 attempts
>
>
>
> but I would rather not have to run a script and search the call log for
> those two strings, extrapolate the reason code and then compare it to my
> custom php/database driven scheduler.
>
>
>
> Is there a way to have it logged to the server?
>
>
>
> Also, both chanunavail attempts and NA attempts both register the status
> as expried. Is this correct or should one of them be failed? I am trying to
> be able to distinguish the two conditions by looking at the archived
> message.
>
>
>
> Thanks,
>
> --E
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Logging to CDR after call file Not Answered

[er ic]

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] NOTICE[28059] pbx_spool.c: Queued call to SIP/102
expired without completion after 0 attempts

but I would rather not have to run a script and search the call log for
those two strings, extrapolate the reason code and then compare it to my
custom php/database driven scheduler.

Is there a way to have it logged to the server?

Also, both chanunavail attempts and NA attempts both register the status as
expried. Is this correct or should one of them be failed? I am trying to be
able to distinguish the two conditions by looking at the archived message.

Thanks,
--E
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Re: [asterisk-users] Logging to CDR after call file Not Answered

[Tech Support]

Why don’t you try it and see what happens.

Regards;

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 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [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 not pick up, 
it will log; as it currently does now.

 

With call files, it is not logging. There is no execution of a dialplan inside 
my assigned context.

 

On Thu, Dec 31, 2015 at 9:29 AM, Tech Support <aster...@voipbusiness.us> wrote:

Here you go… Found this in cdr.conf.sample.

 

; Define whether or not to log unanswered calls. Setting this to "yes" will

; report every attempt to ring a phone in dialing attempts, when it was not

; answered. For example, if you try to dial 3 extensions, and this option is 
"yes",

; you will get 3 CDR's, one for each phone that was rung. Default is "no". Some

; find this information horribly useless. Others find it very valuable. Note, 
in "yes"

; mode, you will see one CDR, with one of the call targets on one side, and the 
originating

; channel on the other, and then one CDR for each channel attempted. This may 
seem

; redundant, but cannot be helped.

;

; In brief, this option controls the reporting of unanswered calls which only 
have an A

; party. Calls which get offered to an outgoing line, but are unanswered, are 
still

; logged, and that is the intended behaviour. (It also results in some B side 
CDRs being

; output, as they have the B side channel as their source channel, and no 
destination

; channel.)

;unanswered = no

 

 

Regards;

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-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 
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] NOTICE[28059] pbx_spool.c: Queued call to SIP/102 expired 
without completion after 0 attempts

 

but I would rather not have to run a script and search the call log for those 
two strings, extrapolate the reason code and then compare it to my custom 
php/database driven scheduler. 

 

Is there a way to have it logged to the server?

 

Also, both chanunavail attempts and NA attempts both register the status as 
expried. Is this correct or should one of them be failed? I am trying to be 
able to distinguish the two conditions by looking at the archived message.

 

Thanks,

--E


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[asterisk-users] Logging in local time

[Luca Bertoncello]

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?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] Logging in local time

[Sebastian Kemper]

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 logger.conf or other file an option to set the
timezone.
Can someone help me?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

Hi Luca,

set up a proper /etc/timezone, see 
http://wiki.openwrt.org/doc/howto/voip.asterisk.

Regards,
Sebastian

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Re: [asterisk-users] Logging SIP connection status for review

[Ishfaq Malik]

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.
 
 A cron job can pbzip2 the files and delete if over x days old.
 
 
 That's completely different.  We already run a good packet capture
 system.  What I want to see is SIP registration statuses and latency
 logged about once a minute.  We do that now by doing a 'sip show peers
 like x' and putting it in a text file.  I can then correlate issues
 with times of high latency or unreachable phones.  I'd just like to
 see more reporting and the ability to correlate times and such.
 
 
How about using your current scripts and then pushing the data into
Graphite?

http://kaivanov.blogspot.co.uk/2012/02/how-to-install-and-use-graphite.html

Ish

-- 
Ishfaq Malik i...@pack-net.co.uk
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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[asterisk-users] Logging SIP connection status for review

[Carlos Alvarez]

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
better reporting.  For example, graphs of latency in a time range, or a
list of unreachable phones within a range, etc.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Logging SIP connection status for review

[Steve Edwards]

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 that logs all the time and lets us 
do some better reporting.  For example, graphs of latency in a time 
range, or a list of unreachable phones within a range, etc.


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.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000--
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Re: [asterisk-users] Logging SIP connection status for review

[Carlos Alvarez]

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 already run a good packet capture system.
 What I want to see is SIP registration statuses and latency logged about
once a minute.  We do that now by doing a 'sip show peers like x' and
putting it in a text file.  I can then correlate issues with times of high
latency or unreachable phones.  I'd just like to see more reporting and the
ability to correlate times and such.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Logging SIP connection status for review

[Ron Wheeler]

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 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 better reporting.  For example, graphs of 
latency in a time range, or a list of unreachable phones within a 
range, etc.


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.



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--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Logging SIP connection status for review

[Duncan Turnbull]

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 have a tool that logs all the time and lets us do some 
 better reporting.  For example, graphs of latency in a time range, or a 
 list of unreachable phones within a range, etc. 
 

How about munin

http://munin-monitoring.org/

With these plugins
http://www.venturevoip.com/news.php?rssid=2322

You could easily adapt one to do registrations. I find the sip peers and calls 
in the last hour quite interesting

Cheers Duncan

 
 
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 -- 
 Ron Wheeler
 President
 Artifact Software Inc
 email: rwhee...@artifact-software.com
 skype: ronaldmwheeler
 phone: 866-970-2435, ext 102
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Re: [asterisk-users] Logging Specific Verbose Level To Seperate File

[Steve Edwards]

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 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.


If you execute the dumpchan() application from an AGI, the output is still 
via Asterisk's logging mechanism.


If your logging level is high enough, that output will be logged. If you 
log to syslogd, it will be as a single line. I don't know what it looks 
like if you log to files.


In any case, I'd say this is a bad approach because the 'verbosity' level 
is too volatile. Some people (me), automagically* bump up verbosity when 
they access the Asterisk console and then automagically set it back to 0 
when they exit the console.


Perhaps executing dumpchan() via AMI would be more promising.

*) Via shell scripting.

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-
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Newline  Fax: +1-760-731-3000

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[asterisk-users] Logging Specific Verbose Level To Seperate File

[Tristram Cheer]

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 logger.conf. Ideally I would like to put DumpChan into
SQL using func_ODBC but it seems that you can't do this so runner up is a
file.

Anyone have any pointers on how to do this? I would like to log DumpChan
output and only DumpChan output to a separate file.


Cheers!
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Re: [asterisk-users] Logging Specific Verbose Level To Seperate File

[Sammy Govind]

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 or greater.

[Syntax]
DumpChan([level])

So in theory its just another Verbose output on CLI, you can separate
Verbose logging to another file in logger.conf. Your verbose level is 1001
so whenever you set core set verbose 1001 this DumpChan() application
will start dumping output in CLI and then fro there be logged in the
Verbose logging file.

I don't think this is exactly what you require.

--
Regards,
Sammy

On Mon, Nov 14, 2011 at 3:19 AM, Tristram Cheer
trist...@tristramcheer.comwrote:

 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 logger.conf. Ideally I would like to put DumpChan into
 SQL using func_ODBC but it seems that you can't do this so runner up is a
 file.

 Anyone have any pointers on how to do this? I would like to log DumpChan
 output and only DumpChan output to a separate file.


 Cheers!

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Re: [asterisk-users] Logging Specific Verbose Level To Seperate File

[Anton Kvashenkin]

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 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 logger.conf. Ideally I would like to put DumpChan into
 SQL using func_ODBC but it seems that you can't do this so runner up is a
 file.

 Anyone have any pointers on how to do this? I would like to log DumpChan
 output and only DumpChan output to a separate file.


 Cheers!

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Re: [asterisk-users] Logging Specific Verbose Level To Seperate File

[Tristram Cheer]

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 channel details
which is going to blow out the size of the dialplan a bit but I cant see
another way around it


Cheers

On 14 November 2011 18:07, Sammy Govind govoi...@gmail.com wrote:

 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 or greater.

 [Syntax]
 DumpChan([level])

 So in theory its just another Verbose output on CLI, you can separate
 Verbose logging to another file in logger.conf. Your verbose level is 1001
 so whenever you set core set verbose 1001 this DumpChan() application
 will start dumping output in CLI and then fro there be logged in the
 Verbose logging file.

 I don't think this is exactly what you require.

 --
 Regards,
 Sammy

 On Mon, Nov 14, 2011 at 3:19 AM, Tristram Cheer 
 trist...@tristramcheer.com wrote:

 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 logger.conf. Ideally I would like to put DumpChan into
 SQL using func_ODBC but it seems that you can't do this so runner up is a
 file.

 Anyone have any pointers on how to do this? I would like to log DumpChan
 output and only DumpChan output to a separate file.


 Cheers!

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Re: [asterisk-users] Logging Specific Verbose Level To Seperate File

[Warren Selby]

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

On Nov 13, 2011, at 11:14 PM, Tristram Cheer trist...@tristramcheer.com wrote:

 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 channel details which 
 is going to blow out the size of the dialplan a bit but I cant see another 
 way around it
 
 
 Cheers
 
 On 14 November 2011 18:07, Sammy Govind govoi...@gmail.com wrote:
 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 or greater.
 
 [Syntax]
 DumpChan([level])
 
 So in theory its just another Verbose output on CLI, you can separate Verbose 
 logging to another file in logger.conf. Your verbose level is 1001 so 
 whenever you set core set verbose 1001 this DumpChan() application will 
 start dumping output in CLI and then fro there be logged in the Verbose 
 logging file.
 
 I don't think this is exactly what you require.
 
 --
 Regards,
 Sammy
 
 On Mon, Nov 14, 2011 at 3:19 AM, Tristram Cheer trist...@tristramcheer.com 
 wrote:
 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 logger.conf. Ideally I would like to put DumpChan into SQL using 
 func_ODBC but it seems that you can't do this so runner up is a file.
 
 Anyone have any pointers on how to do this? I would like to log DumpChan 
 output and only DumpChan output to a separate file.
 
 
 Cheers!
 
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[asterisk-users] logging

[Nicholas Hart]

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
across multiple locations.  Is anyone already doing this... is this
recommended?  Also, any general suggestions on handling all call data both
external and internal would be appreciated.

Thanks,
Nick
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Re: [asterisk-users] Logging the CID from the Privacy Manager

[Jaap Winius]

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 fields, a new field, userfield, appeared at the  
end of the record containing the number submitted by the caller.

I did try to use the same method to change one of the existing fields,  
e.g. src, like this:

Set(CDR(src)=${CALLERID(num)})

But, then I received this error:

[Sep  1 12:26:15] ERROR[12562]: cdr.c:303 ast_cdr_setvar:
Attempt to set the 'src' read-only variable!.

That doesn't seem to be possible. So, I'm happy with your solution.

Thanks, Matt!

Cheers,

Jaap


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[asterisk-users] Logging the CID from the Privacy Manager

[Jaap Winius]

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 Manager, that information is not recorded in the database.  
Instead, all that is written to it:

clid:   Privacy Manager anonymous
src:anonymous

Can the number submitted to the Privacy Manager somehow be recorded in  
the database, instead of anonymous?

Thanks,

Jaap

PS -- Currently, the configuration I'm using in the dialplan for the  
Privacy Manager looks like this:

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 ${CALLERID(num)})
exten = jw,n,PrivacyManager(3,10)
exten = jw,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jw,n,Verbose(-- CID is ${CALLERID(num)})
exten = jw,n,Dial(SIP/1000,60,w)
exten = jw,n(bad),Playback(im-sorry)
exten = jw,n,Playback(vm-goodbye)
exten = jw,n,Hangup()


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Re: [asterisk-users] Logging the CID from the Privacy Manager

[Matt Riddell]

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 =  jw,n,PrivacyManager(3,10)
 exten =  jw,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
 exten =  jw,n,Verbose(-- CID is${CALLERID(num)})
 exten =  jw,n,Dial(SIP/1000,60,w)

Maybe you could do:

Set(CDR(userfield)=${CALLERID(num)})

Before dialing SIP/1000

-- 
Cheers,

Matt Riddell
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[asterisk-users] Logging registration/unregistration of peers/extensions in database

[Bram Bosboom]

  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 asterisk log this 
information directly into a database.

Thanks in advance,
Bram

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Re: [asterisk-users] Logging registration/unregistration of peers/extensions in database

[Nasir Iqbal]

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 b.bosb...@prompt.nl wrote:

  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 asterisk log this
 information directly into a database.

 Thanks in advance,
 Bram

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-- 
Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/
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[asterisk-users] Logging codec used in CDR

[Steve Johnson]

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 name using the exten =h, after
the call completes...

Thanks in advance for all suggestions.

S

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Re: [asterisk-users] Logging codec used in CDR

[Philipp von Klitzing]

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 are on Asterisk 1.6 with the 
adaptive CDR columns.

Note: In principle the codec can be changed in the middle of the call, 
however in practice this very rarely (never) happens.

And while you are at it also look at RTCP stats as well. If you are still 
on Asterisk 1.4 then consider to apply bug/patch #10590.

More details:
http://www.voip-info.org/wiki/index.php?page=Asterisk+RTCP

Philipp

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[asterisk-users] logging stopped suddenly

[Jonas Kellens]

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 modules loaded
vps*CLI logger reload
[Jun 14 11:57:19]   == Parsing '/etc/asterisk/logger.conf': [Jun 14 
11:57:19] Found

[Jun 14 11:57:19] Asterisk Queue Logger restarted
vps*CLI module show like logger
Module Description  
Use Count

0 modules loaded

ls -l /var/log/asterisk :

-rw-rw-r-- 1 root root 0 Jun 14 02:02 messages.vps.hosting.net
-rw-rw-r-- 1 root root 0 Jun 13 02:02 messages.vps.hosting.net.1
-rw-rw-r-- 1 root root592918 Jun  5 02:30 messages.vps.hosting.net.10
-rw-rw-r-- 1 root root 145037679 Jun  4 02:30 messages.vps.hosting.net.11
-rw-rw-r-- 1 root root866083 Jun  3 02:30 messages.vps.hosting.net.12
-rw-rw-r-- 1 root root   1395911 Jun  2 02:30 messages.vps.hosting.net.13
-rw-rw-r-- 1 root root   2151387 Jun  1 02:30 messages.vps.hosting.net.14
-rw-rw-r-- 1 root root 0 Jun 12 02:02 messages.vps.hosting.net.2


How come my logging suddenly stopped and how do I reactivate the logging ??


Jonas.
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Re: [asterisk-users] logging stopped suddenly

[Ishfaq Malik]

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  Use Count

0 modules loaded
vps*CLI logger reload
[Jun 14 11:57:19]   == Parsing '/etc/asterisk/logger.conf': [Jun 14 
11:57:19] Found

[Jun 14 11:57:19] Asterisk Queue Logger restarted
vps*CLI module show like logger
Module 
Description  Use Count

0 modules loaded

ls -l /var/log/asterisk :

-rw-rw-r-- 1 root root 0 Jun 14 02:02 messages.vps.hosting.net
-rw-rw-r-- 1 root root 0 Jun 13 02:02 messages.vps.hosting.net.1
-rw-rw-r-- 1 root root592918 Jun  5 02:30 messages.vps.hosting.net.10
-rw-rw-r-- 1 root root 145037679 Jun  4 02:30 messages.vps.hosting.net.11
-rw-rw-r-- 1 root root866083 Jun  3 02:30 messages.vps.hosting.net.12
-rw-rw-r-- 1 root root   1395911 Jun  2 02:30 messages.vps.hosting.net.13
-rw-rw-r-- 1 root root   2151387 Jun  1 02:30 messages.vps.hosting.net.14
-rw-rw-r-- 1 root root 0 Jun 12 02:02 messages.vps.hosting.net.2


How come my logging suddenly stopped and how do I reactivate the 
logging ??



Jonas.


Hi

Could it be a disk space issue?

Ish

--
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Software Developer
PackNet Ltd

Office:   0161 660 3062
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Re: [asterisk-users] logging stopped suddenly

[Jonas Kellens]

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
/dev/sda1  25G  5.0G   19G  21% /


Jonas


On 06/14/2010 12:22 PM, Ishfaq Malik wrote:


Hi

Could it be a disk space issue?

Ish

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[asterisk-users] logging cdr to mysql does not fill clid field

[pepesz76]

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 appears in Master.csv.  What do I possibly missing? TIA

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Best regards,
 pepesz76  mailto:pepes...@o2.pl





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[asterisk-users] Logging into queue homed off remote system

[Ekelund, Bryan]

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 primary 
being whichever server is local to them. What I'm finding is that when a user 
logs into the queue on Office_A, if their primary server is set to Office_B, 
the IAX trunk ends up getting logged into the queue, which is no good. I tried 
using a SIP trunk to make the connection, but I either end up with a digest 
user mismatch or a Forbidden Auth.  After playing around with 
insecure=port,invite and fromuser=user I was able to log into the queue being 
homed off Office_B, but the agent logged into the queue was the SIP trunk. 
Below is the relevant parts of sip.conf, any suggestions are welcome.

Office_A sip.conf

[general]
Register = office_a:sec...@10.10.40.118/office_b

[office_b]
fromuser=office_a
username=office_a
type=peer
secret=secret
context=incoming
host=dynamic
insecure=port,invite


Office_B sip.conf

[general]
Register = office_b:sec...@10.10.40.118/office_a

[office_a]
fromuser=office_b
username=office_b
type=peer
secret=secret
context=incoming
host=dynamic
insecure=port,invite



Thanks,

Bryan


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[asterisk-users] Logging calls made/lost

[Andreas-Johann Ulvestad]

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 made a solution with a simple web interface already?

Any suggestions would be welcome :-)

-- 
Andreas-Johann Ulvestad
Dagleg leiar, Unicornis


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[asterisk-users] Logging In / Out Agents on Asterisk 6 ???

[David Anthony O Reilly]

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 net posted that they had the same problem but used
Voicemail to authenticate users, but that seemed a nightmare as I don't use
voicemail.

As a work around I have all agents online from the conf files and I use Do
Not Disturb on the phones but this isn't a nice function as it means other
calls outside of the queue cannot come in as all are blocked so not a great
login/logout function.

If anybody could help provide a sample of how they did it on 6 I would be
extremely grateful and will create a WIKI page on it for others as I have
been very unlucky trying to work this out.

Many thanks
David O'Reilly

note-I use extensions.ael but I am sure any code that is for extensions.conf
will be easily convertable as I love AEL
-- 
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Mr. David Anthony O'Reilly, M.Sc (Mob), B.Sc Comp (Hons)

M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009

Computer Science Graduate of The University of Dublin, Trinity College -
B.Sc (Comp) 2008

Email: oreil...@tcd.ie/d...@student.cs.ucc.ie
Tel: +353 (0) 86 030 60 32
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Re: [asterisk-users] Logging In / Out Agents on Asterisk 6 ???

[Carlos Chavez]

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
agentcallbacklogin command did.

Now you have to use the Addqueuemember command to insert a new agent
into a queue.  Go to http://www.voip-info.org/wiki/view/Asterisk+cmd
+AddQueueMember to see how that command works and the the other commands
to remove and pause agents.

On Fri, 2009-05-15 at 22:40 +0100, David Anthony O Reilly wrote:
 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 net posted that they had the same problem
 but used Voicemail to authenticate users, but that seemed a nightmare
 as I don't use voicemail.
 
 As a work around I have all agents online from the conf files and I
 use Do Not Disturb on the phones but this isn't a nice function as it
 means other calls outside of the queue cannot come in as all are
 blocked so not a great login/logout function.
 
 If anybody could help provide a sample of how they did it on 6 I would
 be extremely grateful and will create a WIKI page on it for others as
 I have been very unlucky trying to work this out.
 
 Many thanks
 David O'Reilly
 
 note-I use extensions.ael but I am sure any code that is for
 extensions.conf will be easily convertable as I love AEL
 -- 
 _
 
 Mr. David Anthony O'Reilly, M.Sc (Mob), B.Sc Comp (Hons)
 
 M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob)
 - 2009
 
 Computer Science Graduate of The University of Dublin, Trinity College
 - B.Sc (Comp) 2008
 
 Email: oreil...@tcd.ie/d...@student.cs.ucc.ie
 Tel: +353 (0) 86 030 60 32
 _
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-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Logging In / Out Agents on Asterisk 6 ???

[Jim Dickenson]

Here is what I use:

;  Agent login logout 
exten = *20,1,Answer()
exten = *20,n,wait(.0.5)
exten = *20,n,Read(AgentNumber,agent-user)
exten = *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = *20,n,GotoIf($[${UserID}=]?NOUSER)
exten = *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten = *20,n,GotoIf($[${AgentStatus}=1]?VERIFY)
exten = *20,n,GotoIf($[${AgentStatus}=2]?VERIFY)
exten = *20,n(NOUSER),Playback(cfmc/bad-agent)
exten = *20,n,Playback(vm-goodbye)
exten = *20,n,Hangup()
exten = *20,n(VERIFY),VMAuthenticate(${agentnumb...@ourvm)
exten = *20,n,GotoIf($[${AgentStatus}=2]?AGENTOFF)
exten = *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten = *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten = 
*20,n,AddQueueMember(support,Local/queue${agentnumb...@ansqueue${CUT(CHA
NNEL,-,1)})
;   AQMSTATUS can be  ADDED | MEMBERALREADY | NOSUCHQUEUE
exten = *20,n,Playback(agent-loginok)
exten = *20,n,Verbose(2,Agent ${AgentNumber} added
${DB(users/${UserID}/AgentDevice)})
exten = *20,n,HangUp()
exten = *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten = *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten = *20,n,RemoveQueueMember(support,Local/queue${agentnumb...@ansqueue)
exten = *20,n,Playback(agent-loggedoff)
exten = *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten = *20,n,Hangup()


[ansqueue]
exten = _Queue.,1,Set(AgentNumber=${EXTEN:5})
exten = _Queue.,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = _Queue.,n,Set(AgentDevice=${DB(users/${UserID}/AgentDevice)})
exten = _Queue.,n,Verbose(2,Agent ${AgentNumber} status is
${DEVICE_STATE(${AgentDevice})})
exten = 
_Queue.,n,GotoIf($[${DEVICE_STATE(${AgentDevice})}=NOT_INUSE]?DIALIT)
exten = _Queue.,n,Busy()
exten = _Queue.,n,HangUp()
exten = _Queue.,n(DIALIT),Dial(${AgentDevice},,g)
exten = _Queue.,n,HangUp()


exten = 201,1,Answer()
exten = 201,n,Wait(0.5)
exten = 201,n,Set(qac=${QUEUE_MEMBER(support,free)})
exten = 201,n,GotoIf($[${qac}  0]?HAVEAGNT)
exten = 201,n,Verbose(2,No agents free in support queue)
exten = 201,n,Playback(cfmc/support-no-agent)
exten = 201,n,Voicemail(2...@ourvm,u)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
exten = 201,n(HAVEAGNT),Playback(cfmc/support-intro)
exten = 201,n,Verbose(2,Queuing caller for support agent)
exten = 201,n,Queue(support,nrt,,,120)
exten = 201,n,Verbose(2,Support agent did not answer call)
exten = 201,n,Voicemail(2...@ourvm,b)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/




From: David Anthony O Reilly oreil...@tcd.ie
Reply-To: Asterisk Users Mailing List - Non-Commercial 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 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 net posted that they had the same problem but used
Voicemail to authenticate users, but that seemed a nightmare as I don't use
voicemail.

As a work around I have all agents online from the conf files and I use Do
Not Disturb on the phones but this isn't a nice function as it means other
calls outside of the queue cannot come in as all are blocked so not a great
login/logout function.

If anybody could help provide a sample of how they did it on 6 I would be
extremely grateful and will create a WIKI page on it for others as I have
been very unlucky trying to work this out.

Many thanks
David O'Reilly

note-I use extensions.ael but I am sure any code that is for extensions.conf
will be easily convertable as I love AEL
-- 
_

Mr. David Anthony O'Reilly, M.Sc (Mob), B.Sc Comp (Hons)

M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009

Computer Science Graduate of The University of Dublin, Trinity College -
B.Sc (Comp) 2008

Email: oreil...@tcd.ie    /    d...@student.cs.ucc.ie
Tel: +353 (0) 86 030 60 32
_


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Re: [asterisk-users] Logging Asterisk console

[Enrico Pasqualotto]

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 = /var/lib/asterisk
 astdatadir = /var/lib/asterisk
 astagidir = /var/lib/asterisk/agi-bin
 astspooldir = /var/spool/asterisk
 astrundir = /var/run/asterisk
 astlogdir = /var/log/asterisk
 
 [options]
 verbose = 3
 
 and so I find into /var/log/asterisk the logpro file with the output
 of CLI (verbose) and notice, warning, error, debug message of
 Asterisk.
 

Yes, but with this log I can't see the list of application called.

Enrico.


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[asterisk-users] Logging Asterisk console

[Enrico Pasqualotto]

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.


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Re: [asterisk-users] Logging Asterisk console

[Giancarlo Rubio]

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 bug.

asterisk -rvc |tee /tmp/my_log.txt


 Thanks Enrico.

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-- 
Giancarlo Rubio

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Re: [asterisk-users] Logging Asterisk console

[Danny Nicholas]

Here is a suggestion from the Digium Bug site
asterisk -cvvvgn | tee /tmp/my_log_file.txt

http://bugs.digium.com/view.php?id=14255

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Enrico
Pasqualotto
Sent: 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 that I can see with asterisk -rvvv.
I need it in debugging purpose for tracking some bug.

Thanks Enrico.


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Re: [asterisk-users] Logging Asterisk console

[Marco Sambo]

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 = /var/lib/asterisk/agi-bin
astspooldir = /var/spool/asterisk
astrundir = /var/run/asterisk
astlogdir = /var/log/asterisk

[options]
verbose = 3

and so I find into /var/log/asterisk the logpro file with the output of CLI
(verbose) and notice, warning, error, debug message of Asterisk.


Ciao
Marco


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 bug.

 Thanks Enrico.

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Re: [asterisk-users] Logging outgoing calls

[David fire]

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 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 destination number and the status of the channel
  for example BUSY, UNAVAILABLE etc...

 Yes, if you install the cdr_adaptive_odbc backport.  See the sample config
 file for more information.

 Web:  http://svncommunity.digium.com/view/tilghman/branches/1.4
 SVN:  http://svncommunity.digium.com/svn/tilghman/branches/1.4

 --
 Tilghman

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-- 
(\__/)
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()_()signature to help him gain world domination.
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Re: [asterisk-users] Logging outgoing calls

[Pascal Bruno]

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 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 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 destination number and the status of the
 channel
  for example BUSY, UNAVAILABLE etc...

 Yes, if you install the cdr_adaptive_odbc backport.  See the sample config
 file for more information.

 Web:  http://svncommunity.digium.com/view/tilghman/branches/1.4
 SVN:  http://svncommunity.digium.com/svn/tilghman/branches/1.4

 --
 Tilghman

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[asterisk-users] Logging outgoing calls

[Pascal Bruno]

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 destination number and the status of the channel
for example BUSY, UNAVAILABLE etc...
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Re: [asterisk-users] Logging outgoing calls

[Tilghman Lesher]

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  |
 | · status |
 ||

 I just need to store the destination number and the status of the channel
 for example BUSY, UNAVAILABLE etc...

Yes, if you install the cdr_adaptive_odbc backport.  See the sample config
file for more information.

Web:  http://svncommunity.digium.com/view/tilghman/branches/1.4
SVN:  http://svncommunity.digium.com/svn/tilghman/branches/1.4

-- 
Tilghman

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[asterisk-users] logging the estimated RTT using SIP

[Ricardo Carvalho]

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.
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[asterisk-users] Logging in and off sessions in the dialplan

[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 'Authenticate', but as far as I know,
with this application the user meeds to dial his pin at each call he whats
to make, and that not what I need!

Some ideas?

Thanks,
Ricardo Carvalho.
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Re: [asterisk-users] Logging in and off sessions in the dialplan

[Gordon Henderson]

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 easy answer is yes.

 I am aware of the Asterisk application 'Authenticate', but as far as I know,
 with this application the user meeds to dial his pin at each call he whats
 to make, and that not what I need!

The harder answer is that you need to do a few extra things. Authenticate 
is indeed one of those things, so create an extension the user can dial, 
enter a pin code  authenticate, then store a value in the astDB connected 
with that extenstion... Then you need to check this variable every time 
the user dials a number.

Then finally, you'll need another bit of dialplan to clear the number (or 
maybe an external script that clears it after a certian time)


Gordon

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Re: [asterisk-users] Logging in and off sessions in the dialplan

[Ricardo Carvalho]

Thanks Gordon, I'll give it a try with astDB.

Regards,
Ricardo Carvalho.
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[asterisk-users] Logging to /dev/ttyS0

[Neil Tancock]

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]
email:  mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
web:  http://www.safeharbourit.co.uk/ www.safeharbourit.co.uk
 
 The information in this e-mail is confidential and may be legally
privileged. It is intended solely for the addressee. Access to this e-mail
by anyone else is unauthorised. If you are not the intended recipient, any
disclosure, copying, distribution or any action taken or omitted to be taken
in reliance on it, is prohibited and may be unlawful. When addressed to our
clients, any opinions or advice contained in this e-mail are subject to the
terms and conditions expressed in any applicable governing terms of
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Re: [asterisk-users] Logging to /dev/ttyS0

[Luki]

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)

That will send the callerID number followed by a new line. You can of
course change the format to your desire. Make sure /dev/ttyS0 is
writable by the asterisk user, and is also properly set up (baud rate,
bits, ...).

--Luki
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Re: [asterisk-users] Logging to /dev/ttyS0

[Tzafrir Cohen]

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 file in
logger.conf (make sure that no harm is done on log rotation), log things
to syslog and from syslog to those devices.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [Asterisk-Users] logging performance, important impact?

[Zoa]

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 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 considerable impact in performance because of 
the writing of the logs.




I haven't made benchmarks, but speaking out of my experience and 
knowing that asterisk debug level is very verbose I think it will have 
a sensible impact.
I can remember a very slow samba installation due to the sysadmin 
forgetting to turn off the debug level of logging, it made the 
difference between we can use it and we switch back to windows, 
and I'm talking about a dozen of users, not big numbers.
Are you sure debug level will help you tracking the issues ? Usually 
debug level info is for debug like what is the bottleneck ?, why my 
prepaid agi isn't doing the update on hangup ?, nothing you need to 
keep tracking once you are in production.


imho
Is better to log as few expected stuff as possible and as much 
unexpected stuff as possible.

/imho

Anyway autoanswering your question is pretty simple, put an agi which 
timestamps the first line of each extension and one for the last one, 
send a lot of calls in the system with and without debugging and look 
at the results.

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[Asterisk-Users] logging performance, important impact?

[Moises Silva]

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 considerable impact in performance because of the
writing of the logs.

Best Regards-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org
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Re: [Asterisk-Users] logging performance, important impact?

[Simone Cittadini]

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 just want to 
know if there is a considerable impact in performance because of the 
writing of the logs.



I haven't made benchmarks, but speaking out of my experience and knowing 
that asterisk debug level is very verbose I think it will have a 
sensible impact.
I can remember a very slow samba installation due to the sysadmin 
forgetting to turn off the debug level of logging, it made the 
difference between we can use it and we switch back to windows, and 
I'm talking about a dozen of users, not big numbers.
Are you sure debug level will help you tracking the issues ? Usually 
debug level info is for debug like what is the bottleneck ?, why my 
prepaid agi isn't doing the update on hangup ?, nothing you need to 
keep tracking once you are in production.


imho
Is better to log as few expected stuff as possible and as much 
unexpected stuff as possible.

/imho

Anyway autoanswering your question is pretty simple, put an agi which 
timestamps the first line of each extension and one for the last one, 
send a lot of calls in the system with and without debugging and look at 
the results.

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[Asterisk-Users] logging in problem

[prashant yadav]

 
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 issue only remains with hathway internet connection.what is the nature of problem i need immediate help regarding it 
thanking you 
prashant



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[Asterisk-Users] Logging SIP response codes

[Pedro]

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 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 troubleshooting with our primary provider if
I could log those types of codes.  Verbose just saves way to much
stuff in the log files.
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[Asterisk-Users] Logging SIP response codes

[Pedro]

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 troubleshooting with our primary provider if
I could log those types of codes.  Verbose just saves way to much
stuff in the log files.
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[Asterisk-Users] Logging agents in and out via manager API or other utility. Is it possible?

[Senyo Gualt-Williams]

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.



Thanks,

~Senyo






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[Asterisk-Users] logging events with time stamps

[M.N.A.Smadi]

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?
thanks
m.smadi
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[Asterisk-Users] Logging codec in cdr?

[Roy Sigurd Karlsbakk]

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
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RE: [Asterisk-Users] Logging into Multiple Call Queues on two * Servers and Voice Mail option.

[Robert Jackson]

-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 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 configuring my SNOM phone to actively register with both
servers. Has anyone got a working configuration for this?

I have not implemented this exact scenario, but I have setup agents in
multiple queues.  In queues.conf you can either specify the agents that
belong to the group or use agent groups.  Since each agent can be in
multiple queues all you have to do is specify that agent in the second
queues section.  As far as getting it working with two servers all you
should have to do is use the switch = statement to forward the calls
back and forth.  Basically the agent would have to log in to both
servers, but once finished he/she could receive calls from both.  

2. I need to have an option for the user to press a button when in the
call queue to go to voicemail. Has anyone got a working configuration
for this?

You need to specify the context option in your queues.conf for that
queue.  When you have a context defined whatever buttons the caller hits
is sent directly to that context.  This can work very similarly to an
auto attendant where you can have them hit one to leave a message or two
to receive a callback, etc...

I appreciate all the help. 

No Problem, I hope this qualifies.

Warm Regards 
Shad Mortazavi 


Robert Jackson
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[Asterisk-Users] Logging into Multiple Call Queues on two * Servers and Voice Mail option.

[Shad Mortazavi]

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 configuring my SNOM phone to actively register with both servers. Has anyone got a working configuration for this?

2. I need to have an option for the user to press a button when in the call queue to go to voicemail. Has anyone got a working configuration for this?

I appreciate all the help.


Warm Regards


Shad Mortazavi
---
Nexus Technical Manager
n|m Nexus Management Inc 
Sydney

[Asterisk-Users] Logging errors to syslog

[Jean-Pierre Denis]

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
jp at msfree dot ca


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Re: [Asterisk-Users] Logging errors to syslog

[Ben Klang]

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, 2003-03-13 at 17:32, Jean-Pierre Denis wrote:
 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
 jp at msfree dot ca
 
 
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Re: [Asterisk-Users] logging all console output?

[Martin Pycko]

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 3356

 Computers are like air conditioners.
 They stop working when you open Windows.


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