Re: [asterisk-users] PBX selection

2017-04-21 Thread Roamer2998 
Hi all,

Finally we make decision that go with PortSIP, the reasons are below:

1. Support the easy cluster deployment for handle large concurrent calls
and provide
2. All REST API(this is very important to us for integrate the PBX with our
current system), and also offer the rebrand app for free.
3. The multi-tenant arch.

Thanks all for your suggestions, we have learned a lot of !

BR

On Mon, Apr 17, 2017 at 10:57 PM, Speed Boy  wrote:

>  Hi all, I'm new to VoIP, now we have a project that needs a
>  PBX with client APPs.
> In our team we have argument for choosing PBX. By so far, we
>  have following candidates:
>
> A: Open source
>
>  1) Asterisk PBX (http://www.asterisk.org) (with longest
>  history that almost every one knows it, now the last version using the
> PJSIP stack)
>  2) FreeSwitch (http://www.freeswitch.org) (A lot people
>  recommended it to us)
>
>
> B: Commercial
>
> 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now
> acquired by a HongKong company now
> 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It
> also includes VoIP SDK, WebRTC and offer rebranding app for free.
>
> My boss prefers the Open Source PBX since they are free, but
>  our CTO prefers the commercial editions, according to whom
> the business PBX has better support, and the performance
> is good, and easy to use - considering our team all are new to VoIP/PBX.
>
> We have did some searching of Asterisk, here are my questions:
>
> 1. Does the last Asterisk using PJSIP stack ?
> 2. Does there has the comparison of PJSIP and reSIProcate, sofia(using by
> FreeSwicth) ?
> 3. Is it easy to compile and setup Asterisk?
> 4. Which Asterisk version is recommended? And does Asterisk support
> Windows ?
>
> Thanks in advance .
>
>
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> org/
>
> New to Asterisk? Start here:
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Re: [asterisk-users] log incoming calls without answering

2017-04-21 Thread Tzafrir Cohen 
On Thu, Apr 20, 2017 at 05:51:59PM -0300, Fabio Moretti wrote:
> Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
> >
> > This gets kinda Rube Golberg-ish, but convert the incoming analog line
> > to sip, route it through asterisk and have asterisk do its thing
> > before converting it back to analog to send to the phone. Only problem
> > is you get a lot of extra hardware involved in the mix to make it
> > work. It will be a lot of expense and trouble, so you need to make
> > sure that whatever part you want asterisk to play is worth that
> > effort. Also, I wouldn't touch a fax line in this manner.
> >
> > If you could give a bit more info on what you want asterisk to do, we
> > could maybe give better advice on how to solve your problem.
> 
> Hi Kevin,
> 
> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at  from "

Simple dialplan. Depending on the type of caller ID system, you may need
to wait a few seconds (in case the caller ID is sent after the first
ring).

Thus, assuming you have a DAHDI device, your dialplan is:

exten => s,1,Wait(5)   ; check how much and if waiting is needed
 same => s,n,NoOp(Caller ID is ${CALLERID(num)} on DAHDI channel 
${CHANNEL(dahdi_channel)})

And move on to report from there.

If you also need to report the total time of the call: that might be
possible if the remote side reverses polarity of the channels on call
start and end. Information about it is currently only reported in debug
messages by chan_dahdi. So it is possible (given polarity reversal), but
tricky.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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Re: [asterisk-users] log incoming calls without answering

2017-04-21 Thread J Montoya or A J Stiles 
On Thursday 20 Apr 2017, Fabio Moretti wrote:
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program an
> asterisk for "checking" the inbound calls without answering them, doing
> something like this:
> 
> analog line 1 -+-- asterisk
> 
>\__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid, date
> and time, etc) somewhere, but nothing more, people will answer using the
> old analog phone. The goal is to have a log of the inbound calls without
> touching the old analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: "ok,
> call this AGI, and then don't answer and do nothing more".
> 
> Any idea?

You can get FXO cards, such as the TDM410P/E; which connect to an analogue 
line just like a telephone and then allow the line to be treated as a DAHDI 
channel.  (You would need an FXS card to connect to an analogue telephone; 
this has an additional power supply to generate the 48V DC line voltage and 
80V, 25 Hz ringing voltage.  Usually these cards have 4 or 8 slots for either 
FXO or FXS modules; make sure you buy the right ones.  Explaining to the 
vendor what you want it for may be the best way.)

So you could have Asterisk as an extension to your analogue line, and it could 
monitor incoming calls.  (You would not be able to monitor outgoing calls, 
though; unless you connected each analogue telephone to an FXS port, and used 
Asterisk as a simple PABX.)  

You do not need to Answer() the line to get the caller ID; it comes down the 
line  (either as 300 baud modem tones representing ASCII, or DTMF digit tones, 
depending on your telephone company)  between the polarity reversal and the 
first ring.  And there is no need to have Answer()ed before running an AGI  (in 
fact, you might have good reasons why you would want to have a pre-answer 
script, such as deciding which extension to ring, if you want to direct 
inbound callers straight back to the agent who called them earlier).

All your AGI need do is fork(), and now -- as far as you are concerned -- 
there are two parallel universes, each containing a full copy of the original 
process with the same variables and file handles.  In one, which is the parent, 
fork() returned a non-zero value  (which is the PID of the child process, and 
you might have to clear it out of the process table if it exits before you 
do);  so that process can just exit and return control to Asterisk.  In the 
child process, where the fork() call returned zero  (you don't need to know 
your parent process's PID, since any still-living grandparent will just take 
care of you),  you need just to close STDIN, STDOUT and STDERR  (which are 
still connected to Asterisk)  so it will no longer be waiting for your process 
to do anything -- now you are free!  :)  You then connect to some sort of 
database and run a query like

INSERT INTO call_history(date,source,dest) 
VALUES(NOW(),"${CALLERID(num)}","${EXTEN}"); 

except that ${CALLERID(num)} and ${EXTEN} will have to be passed into the AGI 
script as parameters.  

Then. back in extensions.conf, Dial() a local channel that does nothing but 
Wait() for long enough for the ringing to stop; either because the analogue 
phone has been picked up, or because the caller has stopped trying.  Asterisk 
should get a Hangup() event when the far end hangs up, at any rate.


If you buy a single-channel one of the cheap Chinese clone cards  (from The 
Usual Place),  it ought to last long enough and work well enough for you to 
experiment with; but DO NOT rely on these cheap cards for anything even 
remotely mission-critical.

-- 
JM

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

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Re: [asterisk-users] tcpbind and source IP address

2017-04-21 Thread Kseniya Blashchuk 
JFYI - https://issues.asterisk.org/jira/browse/ASTERISK-26922

чт, 20 апр. 2017 г. в 11:38, Kseniya Blashchuk :

> Hi!
> The issue did not reproduce with pjsip. As for ppa - somebody recommended
> me ppa:sapian/asterisk. Does anybody use it maybe?
>
>
> вт, 18 апр. 2017 г. в 2:24, Ludovic Gasc :
>
>> Hi,
>>
>> I recommend you to install from sources, especially because the latest
>> Asterisk 13 has several bugfixes for pjsip.
>> To my knowledge, nobody proposes ppa or Debian backports for Asterisk.
>>
>> Wazo has Debian packages, but it's only for Debian Jessie and with extra
>> patches for Wazo.
>>
>> --
>> Ludovic Gasc (GMLudo)
>> Lead Developer Architect at ALLOcloud
>> https://be.linkedin.com/in/ludovicgasc
>>
>> 2017-04-16 21:36 GMT+02:00 Kseniya Blashchuk :
>>
>>> Hi!
>>>
>>> Unfortunately pjsip is broken in Ubuntu Asterisk installed from repo.
>>> Yes I also thought to try with pjsip, just to know if it's also affected.
>>> I'll try to make a test next days.
>>>
>>> On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc  wrote:
>>>
 Hi Kseniya,

 You might test with chan_pjsip: We have less production experience with
 chan_pjsip than chan_sip, however, for now, we are more and more confident
 in this new stack while we're digging in documentation and we're testing on
 production.

 However, I've no idea if you'll have the same issue with pjsip, but
 more chances of support on the issues tracker of Asterisk to have help.

 Regards.


 --
 Ludovic Gasc (GMLudo)
 Lead Developer Architect at ALLOcloud
 https://be.linkedin.com/in/ludovicgasc

 2017-03-13 14:41 GMT+01:00 Kseniya Blashchuk :

> Ok, thank you for the assistance!
>
> пн, 13 мар. 2017 г. в 16:38, Joshua Colp :
>
>> On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote:
>> > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel
>> 4.4.0-66-generic
>> > and
>> > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same
>> behavior.
>> > Joshua, maybe you can advice what can be done further?
>>
>> You can file an issue but chan_sip is a community supported module, so
>> there is no guarantee of when it would be looked at and resolved.
>> Ultimately though someone has to spend the time to replicate what is
>> going on, look into the code, and understand what is going on.
>>
>> --
>> 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
>>
>> --
>> _
>> -- 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
>

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

Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

2017-04-21 Thread Marcelo Terres 
Ah, ok.

Everytime you install a package you need to run configure again to
allow detection of new lib.

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 20 April 2017 at 13:16, Pierre Couderc  wrote:
> Thank you very much, Marcello. You got it. The point is to restart
> .configure AFTER installing  these pakcages.
>
> PC
>
>
> On 04/20/2017 01:13 PM, Marcelo Terres wrote:
>>
>> Good question.
>>
>> I am running Asterisk 14 on Ubuntu 16.04 and I had this packages
>> installed:
>>
>> ii  libodbc1:amd64   2.3.1-4.1
>>   amd64ODBC library for Unix
>> ii  odbc-postgresql:amd641:09.06.0200-1.pgdg14.04+1
>>   amd64ODBC driver for PostgreSQL
>> ii  odbcinst 2.3.1-4.1
>>   amd64Helper program for accessing odbc ini files
>> ii  odbcinst1debian2:amd64   2.3.1-4.1
>>   amd64Support library for accessing odbc ini files
>> ii  unixodbc 2.3.1-4.1
>>   amd64Basic ODBC tools
>> ii  unixodbc-dev 2.3.1-4.1
>>   amd64ODBC libraries for UNIX (development files)
>>
>> ii  libltdl-dev:amd642.4.6-0.1
>>   amd64System independent dlopen wrapper for GNU
>> libtool
>> ii  libltdl7:amd64   2.4.6-0.1
>>   amd64System independent dlopen wrapper for GNU
>> libtool
>>
>>
>> Also, I really don't remember of having any kind of problems with odbc
>> support.
>>
>> Did you have all this packages (or equivalents) installed too?
>>
>> Regards,
>> Marcelo H. Terres 
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 19 April 2017 at 17:50, Pierre Couderc  wrote:
>>>
>>> Than you very much.
>>> I use asterisk 14, and yes, menuselect shows me  the need for
>>> generic_odbc(E), res_odbc_transaction(M) and ltdl(E)
>>>
>>> but what does this imply under debian  ?
>>>
>>> I have unixodbc installed an tested and too  libltdl-dev !
>>> But what  am I missing ?
>>>
>>>
>>> On 04/19/2017 10:10 AM, Marcelo Terres wrote:

 What version of Asterisk are you using?

 When I go to cdr_adaptative_odbc in Asterisk 14 it depends of res_odbc
 and res_odbc depends on: generic_odbc(E), res_odbc_transaction(M),
 ltdl(E)

 Regards,
 Marcelo H. Terres 
 IM: mhter...@jabber.mundoopensource.com.br
 https://www.mundoopensource.com.br
 https://twitter.com/mhterres
 https://linkedin.com/in/marceloterres


 On 17 April 2017 at 23:36, nous  wrote:
>
> Thank you, but unixodbc and odbcinst are installed... end even
> unixodbc-dev
>
> But I get the same need for "generic odbc(E)".
>
>
>
> On 17/04/2017 10:48, Marcelo Terres wrote:
>>
>> You need unixodbc and odbcinst packages too, to configure the odbc.
>>
>> []s
>> Marcelo H. Terres 
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 13 April 2017 at 19:41, Pierre Couderc  wrote:
>>>
>>> I use debian stretch and I have installed unixodbc-dev
>>>
>>> but I have a dependency on genreric_odbc in make menuselect
>>>
>>> What am I missing ? Is there an howto ?
>>>
>>> Thanks
>>> PX
>>>
>>> --
>>> _
>>> -- 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] Asterisk 1.8.32.3 : no video (h.264)

2017-04-21 Thread Marcelo Terres 
Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 20 April 2017 at 12:42, Jonas Kellens  wrote:
> Hello
>
> in sip.conf I have ;
>
> videosupport=yes
>
>
>
>
> Kind regards.
>
> J.
>
>
>
> On 20-04-17 13:09, Marcelo Terres wrote:
>>
>> I suppose that you enable the video support on sip.conf, right?
>>
>> Regards,
>> Marcelo H. Terres 
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 19 April 2017 at 13:18, Jonas Kellens  wrote:
>>>
>>> Hello
>>>
>>> using asterisk 1.8.32.3
>>>
>>> I am not able to make a call with video support. I do not know what I am
>>> missing to make this video call.
>>>
>>> Codec h264 should be supported.
>>>
>>>
>>> sip*CLI> core show codecs
>>> Disclaimer: this command is for informational purposes only.
>>>  It does not indicate anything about your configuration.
>>>  INTBINARY  HEX   TYPE   NAME
>>> DESCRIPTION
>>>
>>> ---
>>>1 (1 <<  0)(0x1)  audio   g723
>>> (G.723.1)
>>>2 (1 <<  1)(0x2)  audiogsm
>>> (GSM)
>>>4 (1 <<  2)(0x4)  audio   ulaw
>>> (G.711 u-law)
>>>8 (1 <<  3)(0x8)  audio   alaw
>>> (G.711 A-law)
>>>   16 (1 <<  4)   (0x10)  audio   g726aal2
>>> (G.726 AAL2)
>>>   32 (1 <<  5)   (0x20)  audio  adpcm
>>> (ADPCM)
>>>   64 (1 <<  6)   (0x40)  audio   slin
>>> (16
>>> bit Signed Linear PCM)
>>>  128 (1 <<  7)   (0x80)  audio  lpc10
>>> (LPC10)
>>>  256 (1 <<  8)  (0x100)  audio   g729
>>> (G.729A)
>>>  512 (1 <<  9)  (0x200)  audio  speex
>>> (SpeeX)
>>> 1024 (1 << 10)  (0x400)  audio   ilbc
>>> (iLBC)
>>> 2048 (1 << 11)  (0x800)  audio   g726
>>> (G.726 RFC3551)
>>> 4096 (1 << 12) (0x1000)  audio   g722
>>> (G722)
>>> 8192 (1 << 13) (0x2000)  audio siren7
>>> (ITU
>>> G.722.1 (Siren7, licensed from Polycom))
>>>16384 (1 << 14) (0x4000)  audiosiren14
>>> (ITU
>>> G.722.1 Annex C, (Siren14, licensed from Polycom))
>>>32768 (1 << 15) (0x8000)  audio slin16
>>> (16
>>> bit Signed Linear PCM (16kHz))
>>>65536 (1 << 16)(0x1)  image   jpeg
>>> (JPEG
>>> image)
>>>   131072 (1 << 17)(0x2)  imagepng
>>> (PNG
>>> image)
>>>   262144 (1 << 18)(0x4)  video   h261
>>> (H.261 Video)
>>>   524288 (1 << 19)(0x8)  video   h263
>>> (H.263 Video)
>>>  1048576 (1 << 20)   (0x10)  video  h263p
>>> (H.263+ Video)
>>>  2097152 (1 << 21)   (0x20)  video   h264
>>> (H.264 Video)
>>>  4194304 (1 << 22)   (0x40)  video  mpeg4
>>> (MPEG4 Video)
>>>  8388608 (1 << 23)   (0x80)  videounknown
>>> (unknown)
>>> 16777216 (1 << 24)  (0x100)  videounknown
>>> (unknown)
>>> 33554432 (1 << 25)  (0x200)   textunknown
>>> (unknown)
>>> 67108864 (1 << 26)  (0x400)   textred
>>> (T.140 Realtime Text with redundancy)
>>>134217728 (1 << 27)  (0x800)   text   t140
>>> (Passthrough T.140 Realtime Text)
>>>268435456 (1 << 28) (0x1000)   textunknown
>>> (unknown)
>>>536870912 (1 << 29) (0x2000)   textunknown
>>> (unknown)
>>>   1073741824 (1 << 30) (0x4000)  (unk)unknown
>>> (unknown)
>>>   2147483648 (1 << 31) (0x8000)  (unk)unknown
>>> (unknown)
>>>   4294967296 (1 << 32)(0x1)  audio   g719
>>> (ITU
>>> G.719)
>>>   8589934592 (1 << 33)(0x2)  audiospeex16
>>> (SpeeX 16khz)
>>>  17179869184 (1 << 34)(0x4)  audiounknown
>>> (unknown)
>>>  34359738368 (1 << 35)(0x8)  audiounknown
>>> (unknown)
>>>  68719476736 (1 << 36)   (0x10)  audiounknown
>>> (unknown)
>>> 137438953472 (1 << 37)   (0x20)  audio

Re: [asterisk-users] log incoming calls without answering

2017-04-21 Thread Fabio Moretti 
Thank for all the replies, a lot of input and information!

Sorry for this useless mail, but I really wanted to say thank you.


Il 20/04/2017 17:26, Fabio Moretti ha scritto:
> Hi,
>
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
>
> analog line 1 -+-- asterisk
>|
>\__ analog phone
>
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
>
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
>
> Any idea?
>
> Thanks
>
>
>
>   
>
>

-- 
Fabio Moretti
Gerente de Sistemas
www.tecytal.com 
0800 8780
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Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

2017-04-21 Thread Jonas Kellens 

Hello


you mean while placing a video call ? What info am I looking for in the 
debug output ?





Kind regards.

J.



On 21-04-17 12:28, Marcelo Terres wrote:

Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 20 April 2017 at 12:42, Jonas Kellens  wrote:

Hello

in sip.conf I have ;

videosupport=yes




Kind regards.

J.



On 20-04-17 13:09, Marcelo Terres wrote:

I suppose that you enable the video support on sip.conf, right?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 19 April 2017 at 13:18, Jonas Kellens  wrote:

Hello

using asterisk 1.8.32.3

I am not able to make a call with video support. I do not know what I am
missing to make this video call.

Codec h264 should be supported.


sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
  It does not indicate anything about your configuration.
  INTBINARY  HEX   TYPE   NAME
DESCRIPTION

---
1 (1 <<  0)(0x1)  audio   g723
(G.723.1)
2 (1 <<  1)(0x2)  audiogsm
(GSM)
4 (1 <<  2)(0x4)  audio   ulaw
(G.711 u-law)
8 (1 <<  3)(0x8)  audio   alaw
(G.711 A-law)
   16 (1 <<  4)   (0x10)  audio   g726aal2
(G.726 AAL2)
   32 (1 <<  5)   (0x20)  audio  adpcm
(ADPCM)
   64 (1 <<  6)   (0x40)  audio   slin
(16
bit Signed Linear PCM)
  128 (1 <<  7)   (0x80)  audio  lpc10
(LPC10)
  256 (1 <<  8)  (0x100)  audio   g729
(G.729A)
  512 (1 <<  9)  (0x200)  audio  speex
(SpeeX)
 1024 (1 << 10)  (0x400)  audio   ilbc
(iLBC)
 2048 (1 << 11)  (0x800)  audio   g726
(G.726 RFC3551)
 4096 (1 << 12) (0x1000)  audio   g722
(G722)
 8192 (1 << 13) (0x2000)  audio siren7
(ITU
G.722.1 (Siren7, licensed from Polycom))
16384 (1 << 14) (0x4000)  audiosiren14
(ITU
G.722.1 Annex C, (Siren14, licensed from Polycom))
32768 (1 << 15) (0x8000)  audio slin16
(16
bit Signed Linear PCM (16kHz))
65536 (1 << 16)(0x1)  image   jpeg
(JPEG
image)
   131072 (1 << 17)(0x2)  imagepng
(PNG
image)
   262144 (1 << 18)(0x4)  video   h261
(H.261 Video)
   524288 (1 << 19)(0x8)  video   h263
(H.263 Video)
  1048576 (1 << 20)   (0x10)  video  h263p
(H.263+ Video)
  2097152 (1 << 21)   (0x20)  video   h264
(H.264 Video)
  4194304 (1 << 22)   (0x40)  video  mpeg4
(MPEG4 Video)
  8388608 (1 << 23)   (0x80)  videounknown
(unknown)
 16777216 (1 << 24)  (0x100)  videounknown
(unknown)
 33554432 (1 << 25)  (0x200)   textunknown
(unknown)
 67108864 (1 << 26)  (0x400)   textred
(T.140 Realtime Text with redundancy)
134217728 (1 << 27)  (0x800)   text   t140
(Passthrough T.140 Realtime Text)
268435456 (1 << 28) (0x1000)   textunknown
(unknown)
536870912 (1 << 29) (0x2000)   textunknown
(unknown)
   1073741824 (1 << 30) (0x4000)  (unk)unknown
(unknown)
   2147483648 (1 << 31) (0x8000)  (unk)unknown
(unknown)
   4294967296 (1 << 32)(0x1)  audio   g719
(ITU
G.719)
   8589934592 (1 << 33)(0x2)  audiospeex16
(SpeeX 16khz)
  17179869184 (1 << 34)(0x4)  audiounknown
(unknown)
  34359738368 (1 << 35)(0x8)  audiounknown
(unknown)
  68719476736 (1 << 36)   (0x10)  audiounknown
(unknown)
 137438953472 (1 << 37)   (0x20)  audiounknown
(unknown)
 274877906944 (1 << 38)   (0x40)  audiounknown
(unknown)
 549755813888 (1 << 39)   (0x80)  audiounknown
(unknown)
1099511627776 (1 << 40)  (0x100)

Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

2017-04-21 Thread Derek Bolichowski 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, April 21, 2017 10:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

Hello


you mean while placing a video call ? What info am I looking for in the debug 
output ?




Kind regards.

J.




Why not try removing all codecs from the SIP Peer (deny all, 
allow only H264), unregister the peer, and try a video call again?  If it 
works, try adding G711 back but keep H264 at the top of the priority.
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Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

2017-04-21 Thread Victor Villarreal 
Hi, Jerry,

I don't know what S.O. you have in the Server, but you can check the man
page (https://linux.die.net/man/8/in.tftpd) for tftpd and use the options
--address, so you can tell tftp from what interface/port this service
listen request.

>From the IP in your logs (69.64.57.18) the request came from a web hosting
provider (http://www.heg.com/). So, the request came from Internet, so your
server listen TFTP request from outside, what is bad.

You can use iptables in any Linux distro to block incoming TFTP traffic.
TFTP is a UDP protocol at port 69.

Example:

/sbin/iptables -A INPUT -i eth0 -p udp --destination-port 69 -j DROP

Change eth0 to the correct name of your public internet server interface.



2017-04-21 13:27 GMT-03:00 Jerry Geis :

> I "justed" happened to look at /var/log/messages...
>
> I saw:
> Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
> 0004f2034f6b.cfg
> Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
> 0004f2034f6b.cfg
> Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
> 0004f2034f6c.cfg
> Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
> 0004f2034f6c.cfg
> Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
> 0004f2034f6d.cfg
> Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
> 0004f2034f6d.cfg
> Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
> 0004f2034f6e.cfg
>
> so basically an sequential read of polycom MAC address config files.
> Some is trying to read to determine if I have any polycom files just
> sequential read after read.
> And if so - it would get any extension and password at that time.
> Luckily I have none.
>
> However - how does one block attempts like this ?
>
> Thanks!
>
> Jerry
>
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Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

2017-04-21 Thread Tim S 
Is that IP in your network or outside (I can ping it so I'm guessing it's
outside your network)?  Do you have a firewall between your asterisk box
and the internet?  Is there a WHITELIST of IP addresses that only allow
your provider's limited IP pool to connect to your asterisk box from
outside?

If you are getting TFTP requests hitting your Asterisk box, they are not
properly being filtered at your firewall - ftp and tftp are considered
insecure communication methods, that port (69 I think) should be closed on
your firewall unless you have a really good reason to have it opened (and
unless you run a public FTP site, THERE IS NO GOOD REASON).

Fail2Ban is a BLACKLIST method, blacklists are most effective after good
network hygiene is implemented, as you drastically limit the pool of
potential bad actors with a whitelist.

Best,

-Tim

On Fri, Apr 21, 2017 at 9:38 AM, Dovid Bender  wrote:

> This is old news. They use Shodan and then try to connect. Set up Fail2Ban
> that say after 10 404's to ban the IP.
>
>
> On Fri, Apr 21, 2017 at 12:27 PM, Jerry Geis  wrote:
>
>> I "justed" happened to look at /var/log/messages...
>>
>> I saw:
>> Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
>> 0004f2034f6b.cfg
>> Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
>> 0004f2034f6b.cfg
>> Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
>> 0004f2034f6c.cfg
>> Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
>> 0004f2034f6c.cfg
>> Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
>> 0004f2034f6d.cfg
>> Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
>> 0004f2034f6d.cfg
>> Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
>> 0004f2034f6e.cfg
>>
>> so basically an sequential read of polycom MAC address config files.
>> Some is trying to read to determine if I have any polycom files just
>> sequential read after read.
>> And if so - it would get any extension and password at that time.
>> Luckily I have none.
>>
>> However - how does one block attempts like this ?
>>
>> Thanks!
>>
>> Jerry
>>
>> --
>> _
>> -- 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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>
> 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
>
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Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

2017-04-21 Thread Victor Villarreal 
Hi David, Tim,

Try to use Bail2Ban at last resort. Fail2Ban is a ractive approach, that
permit the traffinc AND ONLY BLOCK them after certain level triggered.


Use iptables to block the unused services faced to public networks like
Internet. And configure these services properly, so they listen only
selected interfaces and IPs, and not from 0.0.0.0

2017-04-21 13:47 GMT-03:00 Tim S :

> Is that IP in your network or outside (I can ping it so I'm guessing it's
> outside your network)?  Do you have a firewall between your asterisk box
> and the internet?  Is there a WHITELIST of IP addresses that only allow
> your provider's limited IP pool to connect to your asterisk box from
> outside?
>
> If you are getting TFTP requests hitting your Asterisk box, they are not
> properly being filtered at your firewall - ftp and tftp are considered
> insecure communication methods, that port (69 I think) should be closed on
> your firewall unless you have a really good reason to have it opened (and
> unless you run a public FTP site, THERE IS NO GOOD REASON).
>
> Fail2Ban is a BLACKLIST method, blacklists are most effective after good
> network hygiene is implemented, as you drastically limit the pool of
> potential bad actors with a whitelist.
>
> Best,
>
> -Tim
>
> On Fri, Apr 21, 2017 at 9:38 AM, Dovid Bender  wrote:
>
>> This is old news. They use Shodan and then try to connect. Set up
>> Fail2Ban that say after 10 404's to ban the IP.
>>
>>
>> On Fri, Apr 21, 2017 at 12:27 PM, Jerry Geis 
>> wrote:
>>
>>> I "justed" happened to look at /var/log/messages...
>>>
>>> I saw:
>>> Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6b.cfg
>>> Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
>>> 0004f2034f6b.cfg
>>> Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6c.cfg
>>> Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
>>> 0004f2034f6c.cfg
>>> Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6d.cfg
>>> Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
>>> 0004f2034f6d.cfg
>>> Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
>>> 0004f2034f6e.cfg
>>>
>>> so basically an sequential read of polycom MAC address config files.
>>> Some is trying to read to determine if I have any polycom files just
>>> sequential read after read.
>>> And if so - it would get any extension and password at that time.
>>> Luckily I have none.
>>>
>>> However - how does one block attempts like this ?
>>>
>>> Thanks!
>>>
>>> Jerry
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
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>> _
>> -- 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
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>
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Re: [asterisk-users] Any way of limiting incoming caller connection time without making 2 active calls for each incoming call?

2017-04-21 Thread D Tucny 
Jonathan,

Have you tried:
same => n,Set(TIMEOUT(absolute)=3540)

You could override later if required.

Thanks,

Dan

On 17 Apr 2017 04:49, "Jonathan H"  wrote:

The following setup prevents callers from going over 59 minutes:

--

[setup]
exten => setup,1,Answer()
same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
same => n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/
time_limit_reached)
same => n,Dial(Local/s@root/n,3,L(354:6))
same => n,Hangup()

[root]

exten => s,1,Verbose(Call to: ${CALLERID(name)} from: ${CALLERID(num)})
same => n,etc etc

--

Thing is, each call shows as 2 calls in the console. Not a big
problem, but also now I want to offer a way of overriding that time
limit.

Is there a different or better way of doing this while still hanging
up after 59 minutes, even when the original incoming call has been
transferred elsewhere?

Thanks.

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Re: [asterisk-users] asterisk as non root

2017-04-21 Thread Atux Atux 
the output of ls -l is
root@pbx: ~ $ ls -l /var/run/asterisk/asterisk.ctl
srwxr-xr-x 1 asterisk asterisk 0 Apr 20 19:47 /var/run/asterisk/asterisk.ctl
root@pbx: ~ $


On Thu, Apr 20, 2017 at 7:46 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Thursday 20 April 2017 at 18:31:03, Atux Atux wrote:
>
> > root@PBX: /var/www/html $ /etc/init.d/asterisk start
> > [ ok ] Starting asterisk (via systemctl): asterisk.service.
>
> I'm somewhat puzzled that your root-user prompt is "$"
> instead of the more normal "#", but never mind...
>
> > root@PBX: /var/www/html $ ps aux | grep asterisk
> > asterisk  1007  0.7  2.3  67128 23748 ?Ssl  Apr19   8:49
> /usr/sbin/asterisk -U asterisk -G asterisk
>
> So, the first column of that output shows you that asterisk is
> running as the user "asterisk".
>
> On my Debian system I only have "-U asterisk" without the "-G asterisk".
>
> > root  4186  0.0  0.1   4192  1992 pts/0S+   17:30   0:00 grep
> asterisk
>
> ...and the grep command was run by "root"
>
> > root@PBX: /var/www/html $ /usr/sbin/asterisk –rx "sip show peers"
> > Privilege escalation protection disabled!
> > See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
> > Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use
> 'asterisk
> > -r' to connect.
>
> Who does "ls -l" show you that file /var/run/asterisk/asterisk.ctl
> is owned by?
>
> On my machine it's:
>
> srwxrwx--- 1 asterisk asterisk 0 Apr 11 10:32
> /var/run/asterisk/asterisk.ctl
>
>
> Antony.
>
> --
> There's a good theatrical performance about puns on in the West End.  It's
> a play on words.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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[asterisk-users] Hack attempt sequential config file read looking for valid files.

2017-04-21 Thread Jerry Geis 
I "justed" happened to look at /var/log/messages...

I saw:
Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
0004f2034f6b.cfg
Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
0004f2034f6b.cfg
Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
0004f2034f6c.cfg
Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
0004f2034f6c.cfg
Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
0004f2034f6d.cfg
Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
0004f2034f6d.cfg
Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
0004f2034f6e.cfg

so basically an sequential read of polycom MAC address config files.
Some is trying to read to determine if I have any polycom files just
sequential read after read.
And if so - it would get any extension and password at that time.
Luckily I have none.

However - how does one block attempts like this ?

Thanks!

Jerry
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Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

2017-04-21 Thread Derek Bolichowski 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Friday, April 21, 2017 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Hack attempt sequential config file read looking for 
valid files.

I "justed" happened to look at /var/log/messages...

I saw:
Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename 0004f2034f6b.cfg
Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found 
0004f2034f6b.cfg
Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename 0004f2034f6c.cfg
Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found 
0004f2034f6c.cfg
Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename 0004f2034f6d.cfg
Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found 
0004f2034f6d.cfg
Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename 0004f2034f6e.cfg

so basically an sequential read of polycom MAC address config files.
Some is trying to read to determine if I have any polycom files just sequential 
read after read.
And if so - it would get any extension and password at that time.
Luckily I have none.

However - how does one block attempts like this ?

Thanks!

Jerry


Jerry,
Can you change to FTP Provisioning, or HTTPS etc? Atleast with FTP you can set 
a user/pass to your directory with mac.cfg to prevent open access.
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Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

2017-04-21 Thread Dovid Bender 
This is old news. They use Shodan and then try to connect. Set up Fail2Ban
that say after 10 404's to ban the IP.


On Fri, Apr 21, 2017 at 12:27 PM, Jerry Geis  wrote:

> I "justed" happened to look at /var/log/messages...
>
> I saw:
> Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
> 0004f2034f6b.cfg
> Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
> 0004f2034f6b.cfg
> Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
> 0004f2034f6c.cfg
> Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found
> 0004f2034f6c.cfg
> Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename
> 0004f2034f6d.cfg
> Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found
> 0004f2034f6d.cfg
> Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename
> 0004f2034f6e.cfg
>
> so basically an sequential read of polycom MAC address config files.
> Some is trying to read to determine if I have any polycom files just
> sequential read after read.
> And if so - it would get any extension and password at that time.
> Luckily I have none.
>
> However - how does one block attempts like this ?
>
> Thanks!
>
> Jerry
>
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>
> 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
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] asterisk name in mysql

2017-04-21 Thread John Kiniston 
You can use func_odbc to do this.

https://wiki.asterisk.org/wiki/display/AST/Getting+Asterisk+Connected+to+MySQL+via+ODBC2

There is a good chapter in the Asterisk book about using ODBC for
hotdesking that may help you understand ODBC as well.

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/getting_funky.html

On Fri, Apr 21, 2017 at 12:12 PM, Atux Atux  wrote:

> hi. currently i am running the phonebook in astdb with
>
>
> *database put cidname 0123456789 "name_surname"*
> and i retrive it with
>
>
> *exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})*
> Now, my system has mysql and i got all my contacts in there in a database
> is called *asterisk *and a table called *addressbook**. *password of the
> mysql is
>
> *whateverpasswd*
> how do i alter the above section of the extensions.conf code to query
> mysql everytime there is a call, please?
>
>
>
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> _
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>
> 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
>



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[asterisk-users] upgrading asterisk 13.13.1 to latest version best practices

2017-04-21 Thread Motty Cruz 
Hello, 

Best practices examples to upgrade Asterisk 13.13.1 to latest version?  

 

Any suggestions? 

 

Thanks,
Motty

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[asterisk-users] asterisk name in mysql

2017-04-21 Thread Atux Atux 
hi. currently i am running the phonebook in astdb with


*database put cidname 0123456789 "name_surname"*
and i retrive it with


*exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})*
Now, my system has mysql and i got all my contacts in there in a database
is called *asterisk *and a table called *addressbook**. *password of the
mysql is

*whateverpasswd*
how do i alter the above section of the extensions.conf code to query mysql
everytime there is a call, please?
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