Re: [asterisk-users] Touch tone stutter

[D'Arcy Cain]

On 2016-11-27 06:46 AM, Max Grobecker wrote:

Hi,

you could try switching the DTMF mode of the ATA's SIP peer (and also in the 
ATA itself) to INBAND transmission.
In this mode, the ATA doesn't need to recognise DTMF tones and your Asterisk 
can interpret it.
For this to work, the ATA needs to use a G.711 codec. Inband DTMF needs an 
uncompressed codec to work properly.

Another way is (if the ATA supports it) to switch DMTF mode to SIP INFO.
In this mode, DTMF is not interpreted out of the audio stream. For external 
peers which are not supporting this mode
Asterisk then generates the proper RTP messages or tones.

With SIP INFO mode I made my best results with all devices, sadly it's not very 
common used.


I have tried all of these ideas and it still does that stutter.  Can 
anyone suggest any other things to try?



--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk compatibility with SMS services

[Emiliano Vazquez]

i'm using gammu[1] with a 3g dongle and my own chip with my preffer
provider. It can send over 700 every our and receive to. I don't know if
you need asterisk and sms in the same way but with this tool you can make
everything. It has python tools to.


Best regards.

Emiliano.

[1] https://wammu.eu/gammu/

On Tue, Nov 29, 2016 at 3:07 PM, Brandon B.  wrote:

>
> Can anyone comment on using SMS in conjunction with VoIP service using one
> of these three VoIP providers: voip.ms, vitelity.com, flowroute.com? Are
> some SMS services more compatible with Asterisk (i.e. SMS over SIP works
> perfectly or not)? Is it best to use a different data channel for SMS
> messages (i.e. SMS via HTTP, SMS via XMPP) instead of Asterisk's built in SMS
> application MessageSend
> ? In order to develop
> a web application for sending and receives SMS messages for business users,
> are there any pitfalls in using Asterisk to handle the message exchanges?
>
>
> On 2016-11-29 09:01 AM, Sebastian Nielsen wrote:
>
> Im using SMS successfully over VoIP. No problems at all. You however need
> to use a good codec.
>
>
>
> However, I don’t use the MessageSend application, instead I use the raw
> SMS() application.
>
> This works by the SMS centre calling my fixed landline from a specific
> number, I detect the callerid, initiate a SMS reception and then the SMS is
> in the spool files.
>
> If I want to send a outgoing SMS, I push a SMS file in the spool folder,
> then initate a call to the SMS centre.
>
>
> That's pretty cool, thank you for the details. You are using the builtin
> SMS application that exchanges SMS data over SIP / PSTN connections. I
> don't believe I can get service like that in Canada. Does anyone use the
> SMS applications to send and receive SMS messages in North America? Who
> provides that service?
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Emiliano Vazquez | PcCentro S.R.L.
White 1611 | C.P. C1407IJG | C.A.B.A.
Office: +54 (11) 4635-7764
Celular: 15.6253.7165
Mail: emilianovazq...@gmail.com 
Web: http://www.pccentro.com.ar
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] app_queue ringall - 2 agents answer same time problem

[marek cervenka]

hmm. i think customer will not agree this is correct behavior

from pcap it looks like there is missing CANCEL to the second device



Dne 30/11/2016 v 19:42 Sam Basan napsal(a):


Your second call is not without sound, there is simply no call at all.
As the first answer the call his channel and the external call channel 
connected.
The second device simply off hook but his channel have no external 
channel to connect.


It's looks like a simple telephony glare.

Sam


בתאריך 30 בנוב' 2016 7:00 PM,‏ "marek cervenka" > כתב:


hi,

our customer reports problem when 2 agents answer the call in the
same time

faster operator (device) answer the call, but the second is showed
up (on device) and call is without sound

asterisk 13.9/app_queue with strategy ringall/operators via Local
channel with sip device (chan_sip)

do you have any tips/info before i will dig deep into logs/debug?

checked google&issues.asterisk.org 
without any clue

marek



-- 
_

-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at:
https://community.asterisk.org/ 

New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started


asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users






-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] inbound T38 to email

[Jeff LaCoursiere]

I have played around with iaxmodem and hylafax and have a few working 
installations where PRI's are involved.  I have a new customer that will 
be sending inbound fax calls via a new (SIP) DID provider we are working 
with (yup, same one from the last message I just sent), and they support 
T38.


I vaguely remembered a 't38modem' project on sourceforge and integration 
with hylafax, and started looking at that today, but t38modem hasn't 
been touched since 2009.


Is there any new modern way to take t38 from a (SIP) DID provider and 
route to email?  Thanks for any insight :)


--
Jeff LaCoursiere
312 962 5250 desk
815 546 6599 cell   


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] new inbound DID provider... no auth?

[Jeff LaCoursiere]

We are trying to work with a new DID provider and I find myself 
confused.  Their standard integration is to send the call with no 
authentication.  I am expected to whitelist all their possible gateways, 
and accept their calls I guess with no peer definition.  I actually have 
it working this way; the calls land in our "public" context, I guess as 
"guest", and I am able to route them from there.  But that makes me nervous.


I would rather at least have them be associated with a defined peer, so 
I can set the right context and any other parameters I might want 
associated.  It is inbound only, no outbound.  I might try to set a 
host= in a peer definition with no secret, and see if that matches it, 
but I would rather avoid making a peer definition for every gateway they 
have.  Can anyone think of a way to define a single peer that might show 
from multiple potential addresses without authentication info?


Cheers,

--
Jeff LaCoursiere
312 962 5250 desk
815 546 6599 cell   


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Specify "name" for Resource in RLS

[Kevin Miller]

Is there are way to specify the display name of a resource in a resource
list?  I have setup a resource list in Asterisk 13 for 1234.  All is working
on the device, but I want to show "Joe User" instead of "1234".  Any
thoughts?


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

[Michael Maier]

Derek Bolichowski wrote:


HI Michael,
You can set this in sip.conf:
session-timers=refuse


I know of this option - it doesn't help, because the provider ignores it 
(on some calls) and the call is dropped anyway.


Normally, there is no problem with the timers. And the problem which 
occurred here is not just the timer, but the session which seams to be 
lost on both sides. But why?



Thanks,
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] app_queue ringall - 2 agents answer same time problem

[Sam Basan]

Your second call is not without sound, there is simply no call at all.
As the first answer the call his channel and the external call channel
connected.
The second device simply off hook but his channel have no external channel
to connect.

It's looks like a simple telephony glare.

Sam

בתאריך 30 בנוב' 2016 7:00 PM,‏ "marek cervenka"  כתב:

> hi,
>
> our customer reports problem when 2 agents answer the call in the same time
>
> faster operator (device) answer the call, but the second is showed up (on
> device) and call is without sound
>
> asterisk 13.9/app_queue with strategy ringall/operators via Local channel
> with sip device (chan_sip)
>
> do you have any tips/info before i will dig deep into logs/debug?
>
> checked google&issues.asterisk.org without any clue
>
> marek
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

[Derek Bolichowski]

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier
Sent: Wednesday, November 30, 2016 12:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Dropped call after 900s: 481 call/transaction does 
not exist and another anomaly during re-invite in timer - full anonymized trace 
attached

Hello all!

I can see a strange problem during invite in dialog in the context of timer 
handling.

Given is the following incoming call from provider at 8.195.88.234 (2@2) to my 
asterisk at 28.19.57.152 (1@1):

After 900s suddenly *asterisk* starts the timer reinvite - I would have 
expected the reinvite started by the provider as usual.

The expected reinvite by the provider is started during authentication of the 
reinvite started by asterisk and is answered immediately by asterisk with sip 
481.

The answer of the provider after the resend of the reinvite came about 0.5s 
later and is sip 481, too.

=> The session obviously isn't known on both sides!

Asterisk therefore now drops the call (bye).


Does anybody has any idea about the reason why both members don't recognize the 
existing session any more? I hope the attached sip trace can shed some light on 
the problem.


Thanks,
Michael


HI Michael,
You can set this in sip.conf:
session-timers=refuse

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached

[Michael Maier]

Hello all!

I can see a strange problem during invite in dialog in the context of
timer handling.

Given is the following incoming call from provider at 8.195.88.234 (2@2)
to my asterisk at 28.19.57.152 (1@1):

After 900s suddenly *asterisk* starts the timer reinvite - I would have
expected the reinvite started by the provider as usual.

The expected reinvite by the provider is started during authentication
of the reinvite started by asterisk and is answered immediately by
asterisk with sip 481.

The answer of the provider after the resend of the reinvite came about
0.5s later and is sip 481, too.

=> The session obviously isn't known on both sides!

Asterisk therefore now drops the call (bye).


Does anybody has any idea about the reason why both members don't
recognize the existing session any more? I hope the attached sip trace
can shed some light on the problem.


Thanks,
Michael


sip481.pcap.gz
Description: GNU Zip compressed data
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] app_queue ringall - 2 agents answer same time problem

[marek cervenka]

hi,

our customer reports problem when 2 agents answer the call in the same time

faster operator (device) answer the call, but the second is showed up 
(on device) and call is without sound


asterisk 13.9/app_queue with strategy ringall/operators via Local 
channel with sip device (chan_sip)


do you have any tips/info before i will dig deep into logs/debug?

checked google&issues.asterisk.org without any clue

marek



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] _FAX_. extension refuses to work !

[A J Stiles]

On Wednesday 30 Nov 2016, Michele Pinassi wrote:
>[stuff deleted]
> but on a call directed to, es. FAX_3700 i got:
> 
> [Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309
> handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension
> 'FAX_3700' rejected because extension not found in context 'from-voip'.
> [Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309
> handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension
> 'FAX_3700' rejected because extension not found in context 'from-voip'.
> 
> Other extension like _IVR_ or _VMR_ works perfeclty and are defined in
> the same manner.
> 
> Maybe _FAX was a reserved keyword ?

Almost.  The letter X is a reserved *character*; it gets translated internally 
to [0-9]. So your extension pattern
_FAX_.
actually matches the pattern
FA[0-9]_.
and something like "FAX_3700" is not going to match this pattern  (because 
there is now a letter X in the place where the extension pattern is expecting 
for there to be a digit 0-9).

Try a backslash before the X?

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] _FAX_. extension refuses to work !

[Jean Aunis]

Hello,

The letter "X" is reserved for dialplan patterns. You should escape it 
this way :


_FA[X]_

Best regards

Jean Aunis


Le 30/11/2016 à 11:45, Michele Pinassi a écrit :


Hi all,

my dialplan is:

/; 
==//

//; FROM VOIP//
//; 
==//

//
//[from-voip]//
//include => default//
//
//[default]
/

/; FAXes//
//exten => _FAX_.,1,Noop("from-voip: FAX ${CALLERID(num)} ${EXTEN}")//
//exten => _FAX_.,n,Set(DID=${EXTEN:4})//
//exten => _FAX_.,n,Goto(fax-services,s,1)/

but on a call directed to, es. FAX_3700 i got:

[Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309 
handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension 
'FAX_3700' rejected because extension not found in context 'from-voip'.
[Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309 
handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension 
'FAX_3700' rejected because extension not found in context 'from-voip'.


Other extension like _IVR_ or _VMR_ works perfeclty and are defined in 
the same manner.


Maybe _FAX was a reserved keyword ?

Michele

--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 -central...@unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo,http://www.faq.unisi.it  





-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] _FAX_. extension refuses to work !

[Michele Pinassi]

Hi all,

my dialplan is:

/;
==//
//; FROM VOIP//
//;
==//
//
//[from-voip]//
//include => default//
//
//[default]
/

/; FAXes//
//exten => _FAX_.,1,Noop("from-voip: FAX ${CALLERID(num)} ${EXTEN}")//
//exten => _FAX_.,n,Set(DID=${EXTEN:4})//
//exten => _FAX_.,n,Goto(fax-services,s,1)/

but on a call directed to, es. FAX_3700 i got:

[Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309
handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension
'FAX_3700' rejected because extension not found in context 'from-voip'.
[Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309
handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension
'FAX_3700' rejected because extension not found in context 'from-voip'.

Other extension like _IVR_ or _VMR_ works perfeclty and are defined in
the same manner.

Maybe _FAX was a reserved keyword ?

Michele

-- 
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di 
Ateneo, http://www.faq.unisi.it 



signature.asc
Description: OpenPGP digital signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

[Michele Pinassi]

Yes, it works !

Thanks :-)

Michele

On 30/11/2016 10:19, Jonathan H wrote:
> I think it might be related to this?
> https://issues.asterisk.org/jira/browse/ASTERISK-26391
>
> I think I remember having to edit logger.conf - this is what mine
> looks like now:
> console => notice,warning,error
> messages => notice,warning,error
>
> Try that, restart asterisk and see if it works :)

-- 
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di 
Ateneo, http://www.faq.unisi.it 




signature.asc
Description: OpenPGP digital signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

[Jonathan H]

I think it might be related to this?
https://issues.asterisk.org/jira/browse/ASTERISK-26391

I think I remember having to edit logger.conf - this is what mine
looks like now:
console => notice,warning,error
messages => notice,warning,error

Try that, restart asterisk and see if it works :)

On 30 November 2016 at 09:09, Michele Pinassi  wrote:
> Hi all,
>
> after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show
> what's happens. I've trying setting debug and verbose to 100 but
> nothing, no show. All commands works as expected but i can't what's
> happens on my asterisk server.
>
> asterisk*CLI> core show settings
>
> PBX Core settings
> -
>   Version: 14.2.0
>   Build Options:   LOADABLE_MODULES, BUILD_NATIVE, OPTIONAL_API
>   Maximum calls:   30 (Current 0)
>   Maximum open file handles:   1024
>   Root console verbosity:  100
>   Current console verbosity:   100
>   Debug level: 100
>   Maximum load average:0.90
>   Minimum free memory: 1 MB
>   Startup time:09:07:33
>   Last reload time:09:07:33
>   System:  Linux/3.16.0-4-686-pae built by root on
> i686 2016-11-28 14:50:24 UTC
>   System name:
>   Default language:en
>   Language prefix: Enabled
>   User name and group: /
>   Executable includes: Disabled
>   Transcode via SLIN:  Enabled
>   Transmit silence during rec: Disabled
>   Generic PLC: Enabled
>   Min DTMF duration::  80
>   RTP dynamic payload types:   96-127
>
> * Subsystems
>   -
>   Manager (AMI):   Disabled
>   Web Manager (AMI/HTTP):  Disabled
>   Call data records:   Disabled
>   Realtime Architecture (ARA): Disabled
>
> * Directories
>   -
>   Configuration file:  /etc/asterisk/asterisk.conf
>   Configuration directory: /etc/asterisk
>   Module directory:/usr/lib/asterisk/modules
>   Spool directory: /var/spool/asterisk
>   Log directory:   /var/log/asterisk
>   Run/Sockets directory:   /var/run/asterisk
>   PID file:/var/run/asterisk/asterisk.pid
>   VarLib directory:/var/lib/asterisk
>   Data directory:  /var/lib/asterisk
>   ASTDB:   /var/lib/asterisk/astdb
>   IAX2 Keys directory: /var/lib/asterisk/keys
>   AGI Scripts directory:   /var/lib/asterisk/agi-bin
>
> Any hint ?
>
> Michele
>
> --
> Michele Pinassi
> Responsabile Telefonia di Ateneo
> Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di 
> Siena
> tel: 0577.(23)5000 - central...@unisi.it
>
> Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di 
> Ateneo, http://www.faq.unisi.it
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

[Michele Pinassi]

Hi all,

after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show
what's happens. I've trying setting debug and verbose to 100 but
nothing, no show. All commands works as expected but i can't what's
happens on my asterisk server.

asterisk*CLI> core show settings

PBX Core settings
-
  Version: 14.2.0
  Build Options:   LOADABLE_MODULES, BUILD_NATIVE, OPTIONAL_API
  Maximum calls:   30 (Current 0)
  Maximum open file handles:   1024
  Root console verbosity:  100
  Current console verbosity:   100
  Debug level: 100
  Maximum load average:0.90
  Minimum free memory: 1 MB
  Startup time:09:07:33
  Last reload time:09:07:33
  System:  Linux/3.16.0-4-686-pae built by root on
i686 2016-11-28 14:50:24 UTC
  System name:
  Default language:en
  Language prefix: Enabled
  User name and group: /
  Executable includes: Disabled
  Transcode via SLIN:  Enabled
  Transmit silence during rec: Disabled
  Generic PLC: Enabled
  Min DTMF duration::  80
  RTP dynamic payload types:   96-127

* Subsystems
  -
  Manager (AMI):   Disabled
  Web Manager (AMI/HTTP):  Disabled
  Call data records:   Disabled
  Realtime Architecture (ARA): Disabled

* Directories
  -
  Configuration file:  /etc/asterisk/asterisk.conf
  Configuration directory: /etc/asterisk
  Module directory:/usr/lib/asterisk/modules
  Spool directory: /var/spool/asterisk
  Log directory:   /var/log/asterisk
  Run/Sockets directory:   /var/run/asterisk
  PID file:/var/run/asterisk/asterisk.pid
  VarLib directory:/var/lib/asterisk
  Data directory:  /var/lib/asterisk
  ASTDB:   /var/lib/asterisk/astdb
  IAX2 Keys directory: /var/lib/asterisk/keys
  AGI Scripts directory:   /var/lib/asterisk/agi-bin

Any hint ?

Michele

-- 
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di 
Ateneo, http://www.faq.unisi.it 




signature.asc
Description: OpenPGP digital signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users