Re: [asterisk-users] Writing CDR's to two database servers

[Sebastian Gutierrez]

use replication

best regards

> On Jun 19, 2017, at 17:47, Tech Support  wrote:
> 
> All;
> I know that there are probably several solutions to this problem, but 
> what I am trying to do is provide some redundancy for my customers CDR data. 
> I know that doing simple backups of MySQL is probably the easiest way to go, 
> but I’m thinking that there may be some benefit to simultaneously writing the 
> CDR data to multiple servers at once. However, I’m drawing a blank on this 
> one. Has anyone else done this before? Any insight at all would be greatly 
> appreciated.
> Thanks Much;
> John V.
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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[Sebastian Gutierrez]

same here.

> On Jun 12, 2017, at 10:02, Kseniya Blashchuk  wrote:
> 
> Same about me - need to re-enable membership all the time. Annoying ((
> 
> пн, 12 июн. 2017 г. в 15:59, John Novack  >:
> Not just gmail
> Happening as well with Comcast.net
> 
> My Comcast address is set to forward to another domain, as Comcast seems to 
> now block sending mail with a non Comcast "from" address. they turned that on 
> a couple years ago with no  notice.
> 
> John Novack
> 
> 
> Jonathan H wrote:
>> Me too, also gmail. I emailed the list owner a couple of days ago, but no 
>> reply.
>> 
>> Is everyone else affected also forwarding to another email address
>> (gmail or not)?
>> 
>> Could be wrong, but I'm guessing there may be an incorrect DMARC
>> policy somewhere - although this is the only fail I could find in the
>> headers.
>> 
>> boun...@lists.digium.com ;
>>dmarc=fail (p=NONE sp=NONE dis=NONE) header.from=gmail.com 
>> 
>> 
>> 
>> 
>> On 12 June 2017 at 09:12, Steve Davies  
>>  wrote:
>>> I am also getting this, three or four times in the last month after years of
>>> no problems.
>>> 
>>> I agree that Gmail is the likely common factor, but I would love to have
>>> access to these bounce messages to know whether it is actually an
>>> overly-paranoid list server!
>>> 
>>> Steve
>>> 
>>> On Mon, 12 Jun 2017 at 09:09 Andrew Furey  
>>>  wrote:
 Ditto; a Gmail issue?
 
 Andrew
 
 On 12 June 2017 at 16:00, Marcelo Terres  
  wrote:
> It is happening the same with me.
> 
> 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 12 June 2017 at 08:07, Olivier  
>  wrote:
>> Hello,
>> 
>> I'm a faithful reader of this mailing list, for several years now.
>> 
>> Lately, I'm receiving emails asking me to re-enable my list
>> subscription due
>> to "excessive bouncing".
>> 
>> What does this exactly mean and why am I receiving this ?
>> Beside re-enabling my subscription, what can I do to improve things ?
>> 
>> Regards
>> 
>> --
>> _
>> -- 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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>  --
> 
> 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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Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

[Sebastian Gutierrez]

Everything is now on release folder on GitHub, documentation and executable.

Hope it helps

On Mar 18, 2017, 20:17 +0100, Sebastian Gutierrez <scg...@gmail.com>, wrote:
> This should work with at least .net framework 4, no dependency needed, just 
> .net framework, I think you should be able to compile it from a vs express 
> version. If you are not able to let me know and next week I will build it for 
> you and upload it as an artifact, in my Astricon 2015 talk (Workflows and 
> Maintainability ) you can also see how to extend this very easily with your 
> custom applications.
>
> Let me know if you need assistance.
>
> Best regards
>
>
> On Mar 18, 2017, 20:13 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> > Hi, thanks - that looks really good!
> >
> > I was about to embark on some non-visual stuff using Ragic, but this
> > looks great.
> >
> > Is there a binary anywhere, or any instructions to compile? I've never
> > compiled C# code before, and although a quick google suggests it
> > shouldn't be too hard, I might need to know a few things like what
> > version of .net it should be compiled with.
> >
> > The readme just points to the website.
> >
> > Thanks!
> >
> > On 18 March 2017 at 18:57, Sebastian Gutierrez <scg...@gmail.com> wrote:
> > > Check this one:
> > >
> > > https://github.com/IntegraCCS/integradesigner
> > >
> > > You can do many things, document each node, and save xml with each
> > > extension.
> > > We´ve made it open source on Astricon 2015 you can extend it the way you
> > > want.
> > >
> > > Hope it helps you.
> > >
> > > Best regards
> > >
> > >
> > >
> > >
> > > On Mar 18, 2017, 12:50 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> > >
> > > How are we all documenting complex dialplan?
> > >
> > > Is there something similar to Doxygen?
> > >
> > > I've got around 20 config files covering around 60 contexts and 40
> > > variables. Of course, I've maintained a basic list of the major stuff,
> > > and documented the code throughout, but it's grown to the stage where
> > > it needs to be better documented, have a proper flowchart etc.
> > >
> > > Talking of flowcharts, I see there are several flowchart makers for
> > > Asterisk and other IVRs - specifically, in the flowchart, I need "set
> > > this variable, uses that variable, calls this context, uses that
> > > gosub" and so on.
> > >
> > > So it's not just dragging extensions together.
> > >
> > > Any ideas?!
> > >
> > > Thanks
> > >
> > > --
> > > _
> > > -- 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
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Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

[Sebastian Gutierrez]

I´ve just added a .exe to the base folder of the project, check if works for 
you, I will try to attach the pdf where everything is explained, is 25mb I will 
see if allows me to upload it.

Best regards


On Mar 18, 2017, 20:13 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> Hi, thanks - that looks really good!
>
> I was about to embark on some non-visual stuff using Ragic, but this
> looks great.
>
> Is there a binary anywhere, or any instructions to compile? I've never
> compiled C# code before, and although a quick google suggests it
> shouldn't be too hard, I might need to know a few things like what
> version of .net it should be compiled with.
>
> The readme just points to the website.
>
> Thanks!
>
> On 18 March 2017 at 18:57, Sebastian Gutierrez <scg...@gmail.com> wrote:
> > Check this one:
> >
> > https://github.com/IntegraCCS/integradesigner
> >
> > You can do many things, document each node, and save xml with each
> > extension.
> > We´ve made it open source on Astricon 2015 you can extend it the way you
> > want.
> >
> > Hope it helps you.
> >
> > Best regards
> >
> >
> >
> >
> > On Mar 18, 2017, 12:50 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> >
> > How are we all documenting complex dialplan?
> >
> > Is there something similar to Doxygen?
> >
> > I've got around 20 config files covering around 60 contexts and 40
> > variables. Of course, I've maintained a basic list of the major stuff,
> > and documented the code throughout, but it's grown to the stage where
> > it needs to be better documented, have a proper flowchart etc.
> >
> > Talking of flowcharts, I see there are several flowchart makers for
> > Asterisk and other IVRs - specifically, in the flowchart, I need "set
> > this variable, uses that variable, calls this context, uses that
> > gosub" and so on.
> >
> > So it's not just dragging extensions together.
> >
> > Any ideas?!
> >
> > Thanks
> >
> > --
> > _
> > -- 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
-- 
_
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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

[Sebastian Gutierrez]

This should work with at least .net framework 4, no dependency needed, just 
.net framework, I think you should be able to compile it from a vs express 
version. If you are not able to let me know and next week I will build it for 
you and upload it as an artifact, in my Astricon 2015 talk (Workflows and 
Maintainability ) you can also see how to extend this very easily with your 
custom applications.

Let me know if you need assistance.

Best regards


On Mar 18, 2017, 20:13 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> Hi, thanks - that looks really good!
>
> I was about to embark on some non-visual stuff using Ragic, but this
> looks great.
>
> Is there a binary anywhere, or any instructions to compile? I've never
> compiled C# code before, and although a quick google suggests it
> shouldn't be too hard, I might need to know a few things like what
> version of .net it should be compiled with.
>
> The readme just points to the website.
>
> Thanks!
>
> On 18 March 2017 at 18:57, Sebastian Gutierrez <scg...@gmail.com> wrote:
> > Check this one:
> >
> > https://github.com/IntegraCCS/integradesigner
> >
> > You can do many things, document each node, and save xml with each
> > extension.
> > We´ve made it open source on Astricon 2015 you can extend it the way you
> > want.
> >
> > Hope it helps you.
> >
> > Best regards
> >
> >
> >
> >
> > On Mar 18, 2017, 12:50 +0100, Jonathan H <lardconce...@gmail.com>, wrote:
> >
> > How are we all documenting complex dialplan?
> >
> > Is there something similar to Doxygen?
> >
> > I've got around 20 config files covering around 60 contexts and 40
> > variables. Of course, I've maintained a basic list of the major stuff,
> > and documented the code throughout, but it's grown to the stage where
> > it needs to be better documented, have a proper flowchart etc.
> >
> > Talking of flowcharts, I see there are several flowchart makers for
> > Asterisk and other IVRs - specifically, in the flowchart, I need "set
> > this variable, uses that variable, calls this context, uses that
> > gosub" and so on.
> >
> > So it's not just dragging extensions together.
> >
> > Any ideas?!
> >
> > Thanks
> >
> > --
> > _
> > -- 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
-- 
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Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

[Sebastian Gutierrez]

Check this one:

https://github.com/IntegraCCS/integradesigner

You can do many things, document each node, and save xml with each extension.
We´ve made it open source on Astricon 2015 you can extend it the way you want.

Hope it helps you.

Best regards




On Mar 18, 2017, 12:50 +0100, Jonathan H , wrote:

> How are we all documenting complex dialplan?
>
> Is there something similar to Doxygen?
>
> I've got around 20 config files covering around 60 contexts and 40
> variables. Of course, I've maintained a basic list of the major stuff,
> and documented the code throughout, but it's grown to the stage where
> it needs to be better documented, have a proper flowchart etc.
>
> Talking of flowcharts, I see there are several flowchart makers for
> Asterisk and other IVRs - specifically, in the flowchart, I need "set
> this variable, uses that variable, calls this context, uses that
> gosub" and so on.
>
> So it's not just dragging extensions together.
>
> Any ideas?!
>
> Thanks
>
> --
> _
> -- 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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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

[Sebastian Gutierrez]

is back online now thanks!

On Feb 14, 2017, 11:18 -0300, Joshua Colp <jc...@digium.com>, wrote:
> On Tue, Feb 14, 2017, at 10:13 AM, Sebastian Gutierrez wrote:
> > The 13.14 tar gz doesn’t even exists on the current or in the old
> > releases folder.
> >
> > there seems to be an issue with the latest build not generating the
> > artifacts?
>
> It was temporarily removed during a synchronization but is now back up
> and the issue should be resolved.
>
> --
> 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
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Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

[Sebastian Gutierrez]

The 13.14 tar gz doesn’t even exists on the current or in the old releases 
folder.

there seems to be an issue with the latest build not generating the artifacts?

best regards



On Feb 14, 2017, 11:04 -0300, Marcelo Terres , wrote:
> Thanks Joshua.
> Marcelo H. Terres  IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 14 February 2017 at 14:01, Joshua Colp  wrote:
> > On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote:
> > > Same problem with me.
> > >
> > > I downloaded the file in 2 different places and had the same error...
> >
> > An issue was filed for tracking this[1] and it will be resolved later
> > today.
> >
> > [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791
> >
> > --
> > 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
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[asterisk-users] Question about async channel or macro for monitoring a call

[Sebastian Gutierrez]

Hi,

 Im trying to do this:  


1) Originate a call between an external number and a ivr that do some things in 
background

2) after the originate I bridge the person that dial that extent with the 
external number

I would like to have the ivr in background while the bridge is up for 
monitoring porpoises, but seems to stop processing when the local bridge is done



other possibility could be having a Macro async??   when I make a dial and 
execute a macro I would like to put a while there and control some stuff, but 
until the macro is over I can´t have audio pass between the 2 channels that 
where dialled.


any hint on this??


thanks


best regards--
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[asterisk-users] audiohook errors

[Sebastian Gutierrez]

Hi,


I´m facing some issues on asterisk 1.8.10. I can see this on the console:

[May 28 15:46:19] ERROR[28099]: lock.c:438 __ast_pthread_mutex_unlock: 
audiohook.c line 705 (audio_audiohook_write_list): Error releasing mutex: 
Operation not permitted
[May 28 15:46:19] ERROR[28099]: lock.c:280 __ast_pthread_mutex_lock: 
audiohook.c line 688 (audio_audiohook_write_list): Error obtaining mutex: State 
not recoverable
[May 28 15:46:19] ERROR[28099]: lock.c:407 __ast_pthread_mutex_unlock: 
audiohook.c line 705 (audio_audiohook_write_list): mutex '(audiohook)-lock' 
freed more times than we've locked!


any ideas???


best regards

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Re: [asterisk-users] Fax Problem on direct FXO port

[Sebastian Gutierrez]

Hi Steve,

Thanks for the reply, I didn't change anything else, just changed spandsp 
version to de FFA, I use the spandsp version with more success in other places, 
but in this particular case, sending faxes works ok with both versions but with 
spandsp I couldn't receive any fax, with FFA I may get 70% of faxes ok.




On May 18, 2012, at 1:35 AM, Steve Underwood wrote:

 Hi Sebastian,
 
 has still some issues that not all faxes pass ok, but does the work == 
 still badly broken
 
 Your log doesn't seem to show a spandsp error. It looks more like a bad 
 signal. Did you change anything else when you installed FFA? Usually people 
 move the other way to improve their results.
 
 Steve
 
 
 On 05/18/2012 09:38 AM, Sebastian Gutierrez wrote:
 Rusty,
 
 thanks for the reply, the issue seems a spandsp issue, I changed to digium 
 free asterisk fax and works much better, has still some issues that not all 
 faxes pass ok, but does the work.
 
 thanks!
 
 
 
 On May 17, 2012, at 1:06 PM, Rusty Newton wrote:
 
 Sebastian,
 
 Seeing as this an issue related to faxing using the SpanDSP library; if you 
 do not get an answer leading to a solution here, then you may try asking on 
 the SpanDSP mailing list http://lists.soft-switch.org/mailman/listinfo
 
 It's likely that the Asterisk users, specifically using SpanDSP, may be on 
 that list.
 
 Thanks,
 
 Rusty Newton
 Open Source Community Support Manager
 Digium, Inc |www.digium.com  |www.asterisk.org
 
 On 5/16/2012 12:44 PM, Sebastian Gutierrez wrote:
 Hi,
 
 
 I´m with asterisk 1.6.2.20
 DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2
 SpanDSP: spandsp-0.0.6pre20.tgz 
 http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre20.tgz
 
 FXO lines.
 
 Sending faxes works ok.
 
 but receiving gives me error:
 
 here is the debug:
 
 http://pastebin.com/qfFeXWQW
 
 
 any idea??
 
 
 Thanks!
 
 
 
 
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[asterisk-users] Fax Problem on direct FXO port

[Sebastian Gutierrez]

Hi,


I´m with asterisk 1.6.2.20
DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2
SpanDSP: spandsp-0.0.6pre20.tgz

FXO lines.

Sending faxes works ok.

but receiving gives me error: 

here is the debug:

http://pastebin.com/qfFeXWQW


any idea??


Thanks!

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[asterisk-users] 1.8 busypatterns

[Sebastian Gutierrez]

Hi,


is it possible to detect 4 length pattern busy cadence detection on FXO lines 
in 1.8??

Here the tones are:

425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off)


in asterisk 1.4 busy detect worked
in asterisk 1.6 didn´t work and i was told that 1.6 can´t handle 4 length 
patterns, but what about 1.8??

for now I can only hangup by asking the provider polarity switch.

Thanks

best regards.


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[asterisk-users] QueueLog from AMI

[Sebastian Gutierrez]

Hi,

 

How can I pass the following data to te queuelog via ami??

 

Agent,data.

 

??

 

I'm doing this:

Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n

 

And thath works fine getting the log with the event but I cant find how to
pass the agent and data parameters

 

Any idea?

 

Thnks

 

 



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Re: [asterisk-users] QueueLog from AMI

[Sebastian Gutierrez]

Not if I have realtime, I'm inserting and deleting from queue_members table,
so I don't have that info.




-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins
Enviado el: Wednesday, November 12, 2008 3:16 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] QueueLog from AMI

On Wed, Nov 12, 2008 at 6:44 PM, Sebastian Gutierrez
[EMAIL PROTECTED] wrote:
 Hi,



 How can I pass the following data to te queuelog via ami??



 Agent,data.



 ??



 I'm doing this:

 Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n



 And thath works fine getting the log with the event but I cant find how to
 pass the agent and data parameters



 Any idea?


From app_queue.c (1.6.0):

 queuename = astman_get_header(m, Queue);
  uniqueid = astman_get_header(m, UniqueId);
  interface = astman_get_header(m, Interface);
  event = astman_get_header(m, Event);
  message = astman_get_header(m, Message);

ast_queue_log(queuename, S_OR(uniqueid, NONE), interface, event,
%s, message);

So, agent would be Interface and data would be Message.

However, i wonder why do you need to pass Login event, as any kind
of Queue Login (dialplan or AMI) would do that automatically.

Regards,
Atisw

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] QueueLog from AMI

[Sebastian Gutierrez]

Would this be part of 1.6.1 release???

Regards


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins
Enviado el: Wednesday, November 12, 2008 5:12 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] QueueLog from AMI

On Wed, Nov 12, 2008 at 7:31 PM, Sebastian Gutierrez
[EMAIL PROTECTED] wrote:
 Not if I have realtime, I'm inserting and deleting from queue_members
table,
 so I don't have that info.


As am I.

I posted a patch that fixes this, so you could be interested in
keeping it in mind (if not even backporting 3 added lines) when
upgrading to 1.6.1.

http://svn.digium.com/view/asterisk?view=revrevision=120166

Regards,
Atis




 -Mensaje original-
 De: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins
 Enviado el: Wednesday, November 12, 2008 3:16 PM
 Para: Asterisk Users Mailing List - Non-Commercial Discussion
 Asunto: Re: [asterisk-users] QueueLog from AMI

 On Wed, Nov 12, 2008 at 6:44 PM, Sebastian Gutierrez
 [EMAIL PROTECTED] wrote:
 Hi,



 How can I pass the following data to te queuelog via ami??



 Agent,data.



 ??



 I'm doing this:

 Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n



 And thath works fine getting the log with the event but I cant find how
to
 pass the agent and data parameters



 Any idea?


 From app_queue.c (1.6.0):

  queuename = astman_get_header(m, Queue);
  uniqueid = astman_get_header(m, UniqueId);
  interface = astman_get_header(m, Interface);
  event = astman_get_header(m, Event);
  message = astman_get_header(m, Message);

 ast_queue_log(queuename, S_OR(uniqueid, NONE), interface, event,
 %s, message);

 So, agent would be Interface and data would be Message.

 However, i wonder why do you need to pass Login event, as any kind
 of Queue Login (dialplan or AMI) would do that automatically.

 Regards,
 Atisw

 --
 Atis Lezdins,
 VoIP Project Manager / Developer,
 [EMAIL PROTECTED]
 Skype: atis.lezdins
 Cell Phone: +371 28806004
 Cell Phone: +1 800 7300689
 Work phone: +1 800 7502835

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-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] ODBCExec and Asterisk 1.6 New Thread

[Sebastian Gutierrez]

Thanks, I also ported my app to 1.6.




-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tilghman
Lesher
Enviado el: Friday, November 07, 2008 2:51 AM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] ODBCExec and Asterisk 1.6 New Thread

On Thursday 06 November 2008 18:59:29 Sebastian Gutierrez wrote:
 Dou you have any example? Can I call directly to querys without the
 templates???

func_odbc.conf:
[EXEC]
read=${ARG1}
write=${ARG1}
dsn=something

extensions.conf:
exten = 123,1,Set(result=${ODBC_EXEC(SELECT foo FROM bar)})

-- 
Tilghman

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[asterisk-users] 1.6 Production ready??

[Sebastian Gutierrez]

Anyone is using 1.6 in production??

Is it ready?

 



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Re: [asterisk-users] 1.6 Production ready??

[Sebastian Gutierrez]

What Hardware? For that performance?




-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Matthew
Fredrickson
Enviado el: Friday, November 07, 2008 3:18 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] 1.6 Production ready??

Steve Totaro wrote:
 
 
 On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:
 
 Sebastian Gutierrez wrote:
   Anyone is using 1.6 in production??
  
   Is it ready?
 
 I have a number of people using 1.6 in production doing SS7-SIP,
 SS7-IAX, and SS7-ISDN gatewaying.
 
 One example (doing SS7-IAX):
 
 System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds
 
 8617029 calls processed
 
 ---
 Matthew Fredrickson
 Digium, Inc.
 
 
 EEEK IAX!!  Do you use IAX for a reason?  Is it because Asterisk does 
 not setup SIP calls very well?  Just curious.

The customer chose to use IAX.  It has been working very well for him.

 Impressive, but very purpose specific.  Do you only load a couple of 
 modules?

Full suite of modules, although it is not using most of them.  I did 
specifically mention in the original message that it was primarily being 
used as a gateway machine.

 I think the question was more along the lines of what Asterisk was meant 
 to be, a feature rich PBX.

Maybe.. or maybe not.  In any case, this is some specific data that 
someone can use about 1.6's performance.


Matthew Fredrickson
Digium, Inc.

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[asterisk-users] ODBCExec from Dialplan

[Sebastian Gutierrez]

Hi,


I'm trying to make odbcexec work with Asterisk 1.6.

I had the attached code (app_odbcexec, not the standard one) working great
with asterisk 1.2 an MSSQL Server on heavy load PBXs with no problem, I'm
trying to port this to asterisk 1.6 but I'm failing to do so.
I attach de working code in 1.2 (app_odbcexec) and my try to port it to 1.6
(app_odbcexec1.6).

Anyone can help??

Thanks


/*
 * Asterisk -- A telephony toolkit for Linux.
 *
 * ODBC exec function
 *
 * Robert Hanzlik [EMAIL PROTECTED]
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License
 *
 * Copyright (c) Digium
 * 
 * Based on work by Mark Spencer and Jefferson Noxon - app_db.c
 * and Brian K. West - app_dbodbc.c
 *
 */

#include asterisk.h



#include asterisk/pbx.h
#include asterisk/module.h
#include asterisk/app.h
#include asterisk/channel.h
#include asterisk/config.h





#include sql.h
#include sqlext.h
#include sqltypes.h

#define AST_MODULE app_odbcexec

static char *tdesc = Database query functions for Asterisk extension logic;

static char *q_descrip =
ODBCquery(varname=query): Retrieves a value from the database query\n
  and stores it in the given variable.  Always returns 0.  If the\n
  query failes, jumps to priority n+101 if available.\n;

static char *e_descrip =
ODBCexec(query): Executes a database query. Always returns 0.\n
  If the query failes, jumps to priority n+101 if available.\n;

static char *q_app = ODBCquery;
static char *e_app = ODBCexec;

static char *q_synopsis = Retrieve a value from a ODBC query;
static char *e_synopsis = Execute a ODBC query;

AST_MUTEX_DEFINE_STATIC(odbc_lock);

static SQLHENV  HOdbcEnv;
static int  ODBC_res;   /* global ODBC Result of 
Functions */
static SQLHDBC  ODBC_con;   /* global ODBC Connection 
Handle */
static SQLHSTMT ODBC_stmt;  /* global ODBC Statement Handle 
*/

static char *config = odbcexec.conf;
static char *dsn = NULL, *username = NULL, *password = NULL;
static int dsn_alloc = 0, username_alloc = 0, password_alloc = 0;
static int connected = 0;

static int ast_odbcexec(const char *query, char *out, int outlen);
static int odbc_load_module(int);
static int odbc_init(void);
static int odbc_unload_module(void);
static int odbc_do_query(char *sqlcmd);
static void reconect(void);


void LogErrMsg(char * source, long rc,SQLSMALLINT HandleType,SQLHANDLE Handle);

void LogErrMsg(char * source, long rc,SQLSMALLINT HandleType,SQLHANDLE Handle)
{
SQLSMALLINT len;
SQLCHAR msg[200],buffer[200];
SQLCHAR sqlstat[10];

ast_log(LOG_ERROR, Error %s %d\n,source,rc);
SQLGetDiagRec(HandleType,Handle,1, 
sqlstat, rc,msg,100,len);
ast_log(LOG_ERROR, %s (%d)\n,msg,rc);
}


static int odbcexec_exec(struct ast_channel *chan, void *data)
{
int arglen, res;
char *argv;

arglen = strlen (data);
argv = alloca (arglen + 1);
if (!argv)  /* Why would this fail? */
{
ast_log (LOG_DEBUG, Memory allocation failed\n);
return 0;
}

memcpy (argv, data, arglen + 1);

  ast_verb (3, odbcexec: query=%s\n, argv);

ast_mutex_lock(odbc_lock);
res = odbc_do_query(argv);
ast_mutex_unlock(odbc_lock);
if(res==-1) {

ast_verb (3, odbcexec: Query failed.\n);
  /* Send the call to n+101 priority, where n is the current 
priority */

if (ast_exists_extension (chan, chan-context, chan-exten, 
chan-priority + 101, chan-cid.cid_num))
chan-priority += 100;
}
return 0;
}

static int odbcexec_query(struct ast_channel *chan, void *data)
{
int arglen;
char *argv, *varname, *query;
char dbresult[256];

arglen = strlen (data);
argv = alloca (arglen + 1);
if (!argv)  /* Why would this fail? */
{
ast_log (LOG_DEBUG, Memory allocation failed\n);
return 0;
}

memcpy (argv, data, arglen + 1);

if (strchr (argv, '='))
{
varname = strsep (argv, =);
query = strsep (argv, \0);
if (!varname || !query)
{
ast_log (LOG_DEBUG, Ignoring; Syntax error in 
argument\n);
return 0;
}


ast_verb (3, odbcquery: varname=%s, query=%s\n, 
varname, query);

if (!ast_odbcexec (query, dbresult, sizeof (dbresult) - 1))
{
pbx_builtin_setvar_helper (chan, varname, dbresult);

ast_verb (3, odbcquery: set variable %s to 
%s\n, varname, dbresult);
 

Re: [asterisk-users] ODBCExec from Dialplan

[Sebastian Gutierrez]

Ok, sorry for the response on the same thread.
The main thing is that with this I set the Store Procedure or Query directly
on the dialplan line, is easier to configure, change, manage, etc.
I also know that works great with heavy load, and it reconnects when the
network goes down and up.

Can you help me porting this app? I think woun`t be difficult for someone
that has port other app.


Thanks


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jared Smith
Enviado el: Thursday, November 06, 2008 12:51 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] ODBCExec from Dialplan

On Thu, 2008-11-06 at 12:16 -0200, Sebastian Gutierrez wrote:
 I'm trying to make odbcexec work with Asterisk 1.6.

Why not just use the functionality of func_odbc already built into
Asterisk 1.6?  Is there something you gain by going with odbcexec that
func_odbc doesn't provide?

Also, just as a word of caution, please don't send a message to the list
by replying to another thread in the list.  It's not good etiquette.
This causes your message to appear inside of another thread, and can be
confusing to people reading the mailing list.


-- 
Jared Smith
Training Manager
Digium, Inc.


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[asterisk-users] ODBCExec and Asterisk 1.6 New Thread

[Sebastian Gutierrez]

Ok, sorry for the response on the same thread.

This is a new one.

 

The main thing is that with this I set the Store Procedure or Query directly
on the dialplan line, is easier to configure, change, manage, etc.

I also know that works great with heavy load, and it reconnects when the
network goes down and up.

 

Can you help me porting this app? I think woun`t be difficult for someone
that has port other app.

 

 

Attachments: (app_odbcexec) working great on 1.2

 (app_odbcexec1.6) my try to port to 1.6

 

Any idea?? Anybody?

 

Should this be on development list?

 

Thanks



/*
 * Asterisk -- A telephony toolkit for Linux.
 *
 * ODBC exec function
 *
 * Robert Hanzlik [EMAIL PROTECTED]
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License
 *
 * Copyright (c) Digium
 * 
 * Based on work by Mark Spencer and Jefferson Noxon - app_db.c
 * and Brian K. West - app_dbodbc.c
 *
 */

#include sys/types.h
#include stdio.h
#include asterisk/options.h
#include asterisk/config.h
#include asterisk/file.h
#include asterisk/logger.h
#include asterisk/channel.h
#include asterisk/pbx.h
#include asterisk/module.h
#include asterisk/pbx.h
#include stdlib.h
#include unistd.h
#include string.h
#include stdlib.h
#include pthread.h

#include sql.h
#include sqlext.h
#include sqltypes.h

static char *tdesc = Database query functions for Asterisk extension logic;

static char *q_descrip =
ODBCquery(varname=query): Retrieves a value from the database query\n
  and stores it in the given variable.  Always returns 0.  If the\n
  query failes, jumps to priority n+101 if available.\n;

static char *e_descrip =
ODBCexec(query): Executes a database query. Always returns 0.\n
  If the query failes, jumps to priority n+101 if available.\n;

static char *q_app = ODBCquery;
static char *e_app = ODBCexec;

static char *q_synopsis = Retrieve a value from a ODBC query;
static char *e_synopsis = Execute a ODBC query;

AST_MUTEX_DEFINE_STATIC(odbc_lock);

static SQLHENV  HOdbcEnv;
static int  ODBC_res;   /* global ODBC Result of 
Functions */
static SQLHDBC  ODBC_con;   /* global ODBC Connection 
Handle */
static SQLHSTMT ODBC_stmt;  /* global ODBC Statement Handle 
*/

static char *config = odbcexec.conf;
static char *dsn = NULL, *username = NULL, *password = NULL;
static int dsn_alloc = 0, username_alloc = 0, password_alloc = 0;
static int connected = 0;

static int ast_odbcexec(const char *query, char *out, int outlen);
static int odbc_load_module(void);
static int odbc_init(void);
static int odbc_unload_module(void);
static int odbc_do_query(char *sqlcmd);
static void reconect(void);

STANDARD_LOCAL_USER;

LOCAL_USER_DECL;

void LogErrMsg(char * source, long rc,SQLSMALLINT HandleType,SQLHANDLE Handle);

void LogErrMsg(char * source, long rc,SQLSMALLINT HandleType,SQLHANDLE Handle)
{
SQLSMALLINT len;
SQLCHAR msg[200],buffer[200];
SQLCHAR sqlstat[10];

ast_log(LOG_ERROR, Error %s %d\n,source,rc);
SQLGetDiagRec(HandleType,Handle,1, 
sqlstat, rc,msg,100,len);
ast_log(LOG_ERROR, %s (%d)\n,msg,rc);
}


static int odbcexec_exec(struct ast_channel *chan, void *data)
{
int arglen, res;
char *argv;

arglen = strlen (data);
argv = alloca (arglen + 1);
if (!argv)  /* Why would this fail? */
{
ast_log (LOG_DEBUG, Memory allocation failed\n);
return 0;
}

memcpy (argv, data, arglen + 1);

if (option_verbose  2)
ast_verbose (VERBOSE_PREFIX_3 odbcexec: query=%s\n, argv);

ast_mutex_lock(odbc_lock);
res = odbc_do_query(argv);
ast_mutex_unlock(odbc_lock);
if(res==-1) {
if (option_verbose  2)
ast_verbose (VERBOSE_PREFIX_3 odbcexec: Query 
failed.\n);
  /* Send the call to n+101 priority, where n is the current 
priority */

if (ast_exists_extension (chan, chan-context, chan-exten, 
chan-priority + 101, chan-cid.cid_num))
chan-priority += 100;
}
return 0;
}

static int odbcexec_query(struct ast_channel *chan, void *data)
{
int arglen;
char *argv, *varname, *query;
char dbresult[256];

arglen = strlen (data);
argv = alloca (arglen + 1);
if (!argv)  /* Why would this fail? */
{
ast_log (LOG_DEBUG, Memory allocation failed\n);
return 0;
}

memcpy (argv, data, arglen + 1);

if (strchr (argv, '='))
{
varname = strsep (argv, =);
query = strsep (argv, \0);
if (!varname || !query)
{

Re: [asterisk-users] ODBCExec and Asterisk 1.6 New Thread

[Sebastian Gutierrez]

Dou you have any example? Can I call directly to querys without the
templates???

Thanks!



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tilghman
Lesher
Enviado el: Thursday, November 06, 2008 4:53 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] ODBCExec and Asterisk 1.6 New Thread

On Thursday 06 November 2008 12:27:05 Sebastian Gutierrez wrote:
 The main thing is that with this I set the Store Procedure or Query
 directly on the dialplan line, is easier to configure, change, manage,
etc.

 I also know that works great with heavy load, and it reconnects when the
 network goes down and up.

You can do the same with func_odbc.  While templates may make your job
easier, you certainly can use whatever syntax you like.  And func_odbc
manages
connections properly, as well.

Please also note that a good majority of the folks who would be qualified to
look at your app are forbidden to do so, as you have not signed a license
for
contributions, and even looking at unlicensed code may affect how we code.

-- 
Tilghman

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RE: [asterisk-users] SOS building fastagi C

[Sebastian Gutierrez]

Hi,

I'm having a problem with chanspy.
I have configured this way

Chanspy(SIP/1)

So it scans all my 1XXX extensions.

That's working just fine, but when I try to switch to an extension ej. 1234#
(it has a call in progress), but the chanspy jumps to another extension, no
te one I selected.

The * feature is working ok.

Thanks in advance for you help,

Bye,


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[asterisk-users] ChanSpy issue

[Sebastian Gutierrez]

Hi,

I'm having a problem with chanspy.
I have configured this way

Chanspy(SIP/1)

So it scans all my 1XXX extensions.

That's working just fine, but when I try to switch to an extension ej. 1234#
(it has a call in progress), but the chanspy jumps to another extension, no
te one I selected.

The * feature is working ok.

Thanks in advance for you help,

Bye,



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   http://lists.digium.com/mailman/listinfo/asterisk-users