Re: [asterisk-users] Reporting for Asterisk Call Center

[Nicolás Gudiño]

Hi Tzafrir,

On Sat, Sep 10, 2011 at 4:28 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Fri, Sep 09, 2011 at 01:28:28PM -0500, Gerardo Barajas wrote:
  There are a lot of reporting tools.
  I have used:
 
  Asternic: http://www.asternic.biz/

Non of those are Free (Open Source).


Clarification: Asternic Call Center Stats Lite is free (GPL3) and can be
downloaded from the above link. The PRO version is commercial.
Asternic CDR reports for FreePBX is also free and available for download on
the asternic.biz site.

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I'm speaking at 4K Conference: http://www.4kconf.com
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Re: [asterisk-users] Receptionist GUI?

[Nicolás Gudiño]

On Mon, Oct 5, 2009 at 6:31 PM, Danny Nicholas da...@debsinc.com wrote:

 $595 US.  Cheap, but depends on the price of local dirt.


LOL... dirt in Argentina is cheaper.

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Re: [asterisk-users] Queues recording CDR

[Nicolás Gudiño]

Hello,

Just a correction, Asternic Call Center Stats is not from
asteriskguru. Asteriskguru has its own statistic program that is not
open source, but free to use. Asternic was written by me (not
asteriskguru) and has an open source version and a commercial one.

Best regards,

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On Mon, Jul 6, 2009 at 12:08 AM, Kurian Thayilkurianmtha...@gmail.com wrote:
 Hi Sriram,

 1. Set the channel variable MonitorFilename before Queue() in dialplan
 and you can give some meaningful filename for record.
 2. I guess you can use an AGI to capture events and then integrate this
 with a DB in the Backend. This should help you to track the activity.
 3. asternic from asteriskguru is kind of OK. Gives you a live and
 detailed report. Parses the queue_log to the MySQL DB and works. This
 parse program could be used in your AGI which I mentioned in point 2.

 Hope this helps.

 Regards,

 Kurian Thayil.

 On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote:
 Hi

 1. I want to record all calls that land to an agent via a queue using
 a meaningful name - as of now i name the recorded file on the fly
 using {CALLERID} variable so that the file gets stored using the
 caller id iunder /var/spool/asterisk/monitor , now if i want to store
 it as CALLERIDEXTEN where call landed from queue how can i do
 this ?
 2. I have a CDR issue - when A calls he is put in Queue and say he is
 answered by Agent B ..Agent B transfers the Call to agent C as it is
 to Agent C whom A wants to talk..when the call gets d/c the CDR for
 that call shows the destination field as B whereas it shd be C...how
 do i take care of this ...in my call center agents are paid on the
 basis of talk time on inbound calls - this way an agent who just
 transfers calls is at merry !!
 3. Are their any GPL based queue reporting software - hows the
 asterisk queue statistics program from asteriskguru.com has anyone
 tried it ?

 Thanks
 Sriram
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Re: [asterisk-users] advice on OrderlyStats (or other cc software)

[Nicolás Gudiño]

As for queue_log analyzers, you can also look at
http://stats.asternic.org/ . I do not want to give you an opinion
because I wrote it myself. There is a fully functional free version.
Best regards,

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On Mon, May 4, 2009 at 9:15 AM, Louis-David Mitterrand
vindex+lists-asterisk-us...@apartia.org wrote:
 On Mon, May 04, 2009 at 10:04:53PM +1000, Rob Hillis wrote:
 Louis-David Mitterrand wrote:
  Hi,
 
  Is anyone here using OrderlyStats with asterisk in a call center
  setting? If so what what is your experience with it? Is that software
  really free for asterisk users?
 
  Or is there a better option out there?

 The short answer is OrderlyStats isn't really free for Asterisk.

 The long answer is that OrderlyStats is free for Asterisk systems with
 two or less agents.  That's really only applicable for the tiniest of
 call centres.

 I haven't used OrderlyStats, so I can't speak for the relative merits of
 it.  However, I have used QueueMetrics (which incidentally is /also/
 free for call centres of two or less simultaneous agents)  and am fairly
 happy with it.  It's not spectacularly pretty - only the latest version
 has begun to introduce graphs and charts, but it's functional.  The
 price is similar to that of OrderlyStats and the licence you purchase
 for both of them is time limited - 4 years in the case of QueueMetrics,
 5 for OrderlyStats.  QueueMetrics will offer a 50% discount for
 non-profit organisations - I don't know whether OrderlyStats offers the
 same thing or not.

 Thank you Rob for the detailed and informative answser. Much
 appreciated.

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Re: [asterisk-users] Channels crossing...

[Nicolás Gudiño]

Hey Steve, it's been a while...

On Thu, Oct 2, 2008 at 8:33 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
 I have seen and heard the recordings to prove crossed calls between people
 and agents in a busy call center, it was kind of funny with two agents
 trying to figure out what was going on and a very confused customer.

 I certainly would not rule out Asterisk, but you always start with the
 cables.



I have just seen this myself on a pure voip and busy callcenter...
with recordings to prove it too. I would not rule Asterisk either, as
I do not have many cables to check!

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Re: [asterisk-users] Digium announcement: new community manager - John Todd

[Nicolás Gudiño]

Hi John, congratulations!... and to Digium for hiriing such talented people.

 I'd like to take a few moments to introduce myself and the new role
 in which I'll be working for Digium to further the Asterisk project
 and environment.  As you may know, Digium plays a key part in
 assisting with the development of the Asterisk project, and so I am
 pleased to be working for them in a full-time capacity as Asterisk
 Open Source Community Director, replacing the talented Jared Smith
 who has moved over to do customer training and bootcamps in order to
 bring more people into the quickly-growing Asterisk environment.

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Re: [asterisk-users] Problem while running Flash Operator Panel

[Nicolás Gudiño]

On Thu, May 15, 2008 at 7:57 AM, Sukhbir Singh
[EMAIL PROTECTED] wrote:
 Hi All,

   Whenever i try to start FOP using script
  ./op_panel_redhat.sh start given in directory
 /usr/local/op_panel-snapshot/init

 I got the following error:

 Starting Flash Operator Panel: execvp: No such file or directory

[FAILED]

 Please let me know the reason for this.
 Thanks in Advance

 With Regards,
 newbie


You will have to edit op_panel_redhat.sh and change it so it reads:

DAEMON=/usr/local/op_panel-snapshot/op_server.pl
OPTIONS=-d

Then you will want to move the file to /etc/rc.d/init.d

And add it to startup with:

chkconfig --add op_panel_redhat.sh


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[asterisk-users] Queue Stats

[Nicolás Gudiño]

Hello,

I have finally released the queue stats package to the public.. please go to:

http://www.asternic.org/stats

To get it or see the online demo.

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Re: [asterisk-users] Drag and Drop transfer application

[Nicolás Gudiño]

Hello,

On Thu, Apr 24, 2008 at 9:24 PM, Al lists [EMAIL PROTECTED] wrote:
 any of you guys have used FOP for drag and drop transfer on 30 40 phones
 environment?
 how stable is that?
 I'm playing with it but so far drag and dropping phone icon to another phone
 disconnectes the call.


If your calls disconnects it is probably a misconfiguration (the
asterisk CLI with some debug and verbose levels will help you find
out). You have to match a context and extension as defined in
op_buttons.cfg with the [EMAIL PROTECTED] you use in your dialplan.
Also you have two ways or points of view to perform transfers,
dragging the other leg (default behaviour), or your own.. you have to
select which one you like with reverse_transfer=1 in op_server.cfg

Best regards,

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Re: [asterisk-users] Web page to show online extensions?

[Nicolás Gudiño]

Hi, I have released 0.28 the other day... I will probably make a new
branch for asterisk 1.6 soon..

On Thu, Apr 3, 2008 at 3:42 PM, Dean Collins [EMAIL PROTECTED] wrote:
 Cute :)

  I was thinking about getting something more complex developed but yes
  FOP is a great product though getting a little old.time for the next
  version?




  Regards,

  Dean Collins
  Cognation Pty Ltd
  [EMAIL PROTECTED]
  +1-212-203-4357
  +61-2-9016-5642 (Sydney in-dial).


   -Original Message-
   From: [EMAIL PROTECTED] [mailto:asterisk-users-

  [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
   Sent: Thursday, 3 April 2008 2:32 PM
   To: asterisk-users@lists.digium.com
   Subject: Re: [asterisk-users] Web page to show online extensions?
  


  On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote:
What about building an Adobe AIR application that can do this.
  
   Any application that connects to the manager interface can do that. It
   can be AIR, or FIRE or GROUND.
  
   The FOP exists and does that.
  
   --
  Tzafrir Cohen
   icq#16849755  jabber:[EMAIL PROTECTED]
   +972-50-7952406   mailto:[EMAIL PROTECTED]
   http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
  
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Re: [asterisk-users] Web page to show online extensions?

[Nicolás Gudiño]

As other people mentioned, there is also  a DHTML client, if you look
at the javascript code, it is not hard to pull your own interface for
it... like:

1) Extension list like a grid, you can manage it via keyword only (hit
F for filter, type some letters to narrow the list,, hit T for
transfer the call - altought it is not implemented yet, is just an
experiment)

http://www.asternic.org/dhtml3/

2) Special queue monitor with alarms (it will fire an alarm when a
timer or call count is reached, color of queue goes from green to red
depending on number of calls and threshold alarm). Hybrid with a
regular flash panel at the bottom.

http://www.asternic.org/dhtml4/

The sky is the limit (but you will have to code it in javascript!)

Best regards,




On Thu, Apr 3, 2008 at 4:55 PM, Earl Terwilliger [EMAIL PROTECTED] wrote:
 On Thursday 03 April 2008 02:59:07 pm faraz wrote:
   FOP is quite clunky!

  one reason i wrote the event montor... which is in PHP (and Ajax or rather
  Ajap) and does not poll the asterisk manager (which in my opinion overloads
  asterisk)



  
   Also the flash is almost un-usable with a large number of extensions
   Would love to see something in PHP/Ajax which could be lightweight and
   fast.
  
   We are working on something along those lines which we should be able to
   release in a few months.
  
   On Thu, 2008-04-03 at 14:42 -0400, Dean Collins wrote:
Cute :)
   
I was thinking about getting something more complex developed but yes
FOP is a great product though getting a little old.time for the
next
version?
   
   
   
Regards,
   
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
   
 -Original Message-
 From: [EMAIL PROTECTED]
   
[mailto:asterisk-users-
   
 [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
 Sent: Thursday, 3 April 2008 2:32 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Web page to show online extensions?

 On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote:
  What about building an Adobe AIR application that can do this.

 Any application that connects to the manager interface can do that.
   
It
   
 can be AIR, or FIRE or GROUND.

 The FOP exists and does that.

 --
Tzafrir Cohen
 icq#16849755  jabber:[EMAIL PROTECTED]
 +972-50-7952406   mailto:[EMAIL PROTECTED]
 http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] AddQueueMember and Flash Operator Panel

[Nicolás Gudiño]

Hi,


On Wed, Jan 16, 2008 at 10:11 PM,  [EMAIL PROTECTED] wrote:
 Hello users!

  Recently I read that AgentCallbackLogin is going to be deprecated soon.
  Wanting to set up a few callback type queues, I set them up as suggested
  in queues-with-callback-members.txt.

  I was able to set the queues up completely this way, however, I'm trying
  to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login
  status.  FOP monitors their status if I call AddQueueMember with their
  actual interface (which, by the way, makes more sense to me than logging
  them in via chan_local), and it even seems to work with
  Local/[EMAIL PROTECTED]  But if I use any context other than
  default here, FOP doesn't recognize that the agent is logged in.

  (The users' default context isn't even set to default, and it behaves this
  way even if the users' voicemail context is something else, so I am
  guessing that is hard-coded in FOP somewhere.)

No, it is not hardcoded, it will try to match the context defined in
the op_buttons.cfg file for the channel you monitor. If you use
Local/[EMAIL PROTECTED], then in a button definition you have to
use something like:

[SIP/james]
Extension=1000
Context=my-agent-context


And it will *probably* work... I say probably because local channels
are a #!@ to track with manager events. Look at the source, there is a
function named local_channels_are_driving_me_mad

Best regards,

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Re: [asterisk-users] Attended transfers through a GUI

[Nicolás Gudiño]

Hi

On Wed, Feb 27, 2008 at 5:41 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
 Greetings list,

  I've been playing around this afternoon with Flash Operator Panel, trying to
  get it to do attended transfers. I am running the latest version.

  Has anyone managed to get this working reliably, and if so, would you mind
  sharing how you did it please?


Native attendant transfers are not available in FOP (nor in asterisk
unless you use 1.6 or patch 1.4).

FOP use a kind of hack to perform the task, putting a caller on hold,
redirecting channels to a meetme room, and reconnecting the held
channel to the meetme when one of the parties hangs up. So, it is kind
of  a kludge. It works. But it uses meetme for bridging and for that
very same reason the call logic is broken... it is not intuitive to
the user (because once the call is transferred you are inside a meetme
room).

I will implement native attendant transfers with asterisk 1.6.

Best regards,




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Re: [asterisk-users] calls not terminating

[Nicolás Gudiño]

Hello,



In short – Asterisk is not able to recognize that the 'other' person to whom
call was made has hung up – hence the channel stays busy.



http://kb.digium.com/entry/1/6/

I would try with busydetect and busycount.. Best regards,

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Re: [asterisk-users] FOP is not displaying all my SIP extensions neither all E1 channels , why?

[Nicolás Gudiño]

Hi,

I must say that i'm not very used with customization of FOP. I've a box
runing Flash Op.Panel, and i notice that the screen is full of buttons from
my sip users, as well as Zapata channels.

The problem is that i have more Zapata channels as well as SIP users, is
there any way to get a scroll on this to display everything? do i need to
resize the buttons?

For sure someone now how to solve this basic question:)


You can reduce the button size, update to the latest snapshot that
includes 'slow' horizontal scrolling, or both. Best regards,

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Re: [asterisk-users] pop a web page with DID in url

[Nicolás Gudiño]

On 10/5/06, Michael Sampson [EMAIL PROTECTED] wrote:

I'm looking to do this.
When a call comes in to an agent in a queue, pop a web page like this
http://www.mydomain.com/cgi-bin/script.cgi?did=952900
Where did is the number the caller dialed to reach the system in the
first place.

I know Hudlite can do this we caller ID, but the DID feature is not
there yet.

Does anyone have any other software they know of that can do this?


FOP can do that. http://www.asternic.org

Best regards,

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Re: [asterisk-users] Spying a channel in a meetme

[Nicolás Gudiño]

I'm using the ChanSpy command for monitor a conversation of a channel which is
in a meetme conference. All comunications go throught voip, with some voip
phones attached to the lan and an external voip providor in order to make
external calls.
The problem is that sometimes the spy call can hear the other persons of the
conference, but sometimes it works ok. Almost all conferences are only of two
channels.
exten=s,1,Chanspy(${SPYCHAN}|q)

I will try using monitor mode in meetme application, but I prefer Chanspy
because I can spy the call always, not only when it is in a conference.


You can join the Meetme room muted...

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Re: [asterisk-users] How to install HUDLite Server

[Nicolás Gudiño]

And also it would be nice to have more detailed instructions on how to
upgrade FOP.


This belongs to FOP mailing list... but anyways:

0) backup your previous install, just in case
1) replace op_server.pl
2) replace operator_panel.swf
3) read UPGRADE, you might need to add 3 or 4 new parameters to
op_style.cfg and maybe op_server.cfg and op_buttons.cfg. Those
parameters are usually optional, so this step is probably optional.
4) restart the panel

If you are upgrading for an old version and you are using an outdated
version of FreePBX/AMP, you might need to modify the script that
generate the config file (retrieve_op_conf_from_mysql.pl) in order to
display queue activity, because queue buttons where renamed from
[queuename] to [QUEUE/quename]. That script is part of AMP/FreePBX,
not FOP.

All of this is documented somehow in the tarball, online documentation.

You are always welcome to improve the documentation if you feel is
innacurate or incomplete, make translations, etc.

And if you do not have time to do custom mods, you can hire someone to
do it for you... ;)

Best regards,


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Re: [asterisk-users] FOP Installation help

[Nicolás Gudiño]

Hi,

You might need to recreate fop config files with amp tools. The script
that does that is named retrieve_op_conf_from_mysql.pl. You might also
have permissions problems on the directory or files inside them. They
must be owned by user asterisk if I recall correctly. Good luck,

On 9/18/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:


I had FOP installed previously on [EMAIL PROTECTED] which was working perfectly 
fine until
today when I decided to upgrade it to its ver. 2.6

I ended up losing my old FOP and new one no success.

First I backed up /var/www/html/panel to /var/www/html/panel_old. But after
no sucess with new FOP, moved the files back to original location.

But now the error I get is that it doesn't show all the extensions, and no
trunks and queues at all.

Please help me how to fix this problem. I didn't change anything in these
files at all, then why all of a sudden they've stopped working.

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Re: [asterisk-users] How to install HUDLite Server

[Nicolás Gudiño]

As for FOP, when clients come to meet you after seeing attractive interfaces
from other proprietary systems, its just embarrassing to show them such an
ugly interface like FOP.


FOP interface, altough it has some limitations (fixed button positions
and size for the flash client), it is pretty much configurable. It can
look ugly or impresive, depending on the time you put to create your
own look. You can use custom background for the whole screen, or for
each button, etc.

Using the dhtml panel there is no limitations on the layout, you can
make it look as anything, including HUD. It is limited to your html
skills. Have you taken a look at http://www.asternic.org live demo?

Anyways, attractive is not everything, sometimes the information
displayed is more important. The most advanced call center reports I
saw are just plain text with standard formatting. I guess the
important thing on this interfaces should be usability and flexibility
and not only looks.

Best regards,


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Re: [asterisk-users] Phone status

[Nicolás Gudiño]

Hello,



I'm working on a project, where I need the status of every telephone on the
system. (Idle,ringing,busy)


snip


I have made a deamon, which query Asterisk every second for active calls,
this works by issuing a Status to the manager-interface, and processing
the return data and then put the result into a MySQL table.


snip



Am I doing it in a stupid way, I'm aware that the Manager can give me
realtime events, but I'm under the impression, that it is not very stable in
a high traffic environment?


It is far less stable if you perform queries every second than just
pasively listening to the events.

Regards,

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Re: [asterisk-users] Joining calls via manager.api or AGI

[Nicolás Gudiño]

Is there a way to initiate 2 different calls and connect them together with
Asterisk, using the manager.api or the AGI system? I want to link the calls
without using DTMF, such as with an SMS or web triggered script.


The only way right now is using meetme. There is a patch with a
'bridge' function but is marked as post 1.4  (
http://bugs.digium.com/view.php?id=5841 ). This is a very much needed
feature.


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Re: [asterisk-users] Queue Management

[Nicolás Gudiño]

Hi Eric,


I'm working in a small call center, but with special requirements. We
currently have a couple of clients, all of them have
specific phone numbers configured in our system, so when we get a call
for a specific client we take down the information via a webpage
then it sent via email to them.

One of the major problem that I'm seeing is the queue management. Right
now with our current system, the agents are able to see what call
are coming in, which one haven't been answered, which one are on hold.
(That part is not so bad with Asterisk since it's already taking care of
this)
But the part I'm worring about is that the agent can see the Greeting
message for the customer line. So the agent knows what to say before
answering the
line then IE popups with the URL for that client.

Not sure if that can be replicated with Asterisk. We could probably
adapt our selfs by doing a query about the DNIS and then store the DNIS
associated with his
greeting. Almost like what we do now actually... The only thing is to
put all that together... hehehe



There are serveral packages that can be used in your situation. You
can certainly use FOP ( http://www.asternic.org ) to handle popups and
displays status of your asterisk queues.  There are also soft phones
that can handle url passing, some feature complete callcenter packages
(astguiclient). You can also do it yourself using the manager API or
custom AGI scripts. Regards,

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Re: [asterisk-users] Manager interface

[Nicolás Gudiño]

On 7/30/06, Dovid Bender [EMAIL PROTECTED] wrote:

any programs out there that do this ?

Dovid


You can use FOP: http://www.asternic.org

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Re: [asterisk-users] Flash operator panel

[Nicolás Gudiño]

Hi Jordan,

You might want to subscribe to FOP mailing list. You can do that from
http://www.asternic.org



Can anybody steer me in the right direction? I have installed the fop and
have it working okay, first problem is agent logins not changing the state
color when an agent logs in. I configured it on two boxes one works the
other doesn't, same configs alll the way. The other is more of me not
understanding how it works. I only see the buttons that i have programmed
and am unable to get the password entry box and can't figure out how to do
transfers.


Agent logins work depending on the type of login that you use. You can
use agentlogin, agentcallbacklogin or addqueuemember in asterisk. Each
one has a special treatment/config setting in FOP. You can read it in
the example config files or the online documentation.

About your problems with the security code box, I do not understand
what your problem is. You have to enter the security code at least one
(and it has to match the one defined in op_server.cfg) in order to
perform any action, including transfers. Once that the security code
is verifies, the lock icon shows closed and you can perform the
action. To transfer a call you have to drag the phone icon to the
destination. Anyways, please check FOP archives or subscribe to the
mailing list as this is related to FOP and not Asterisk itself.

Regards,


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Re: [Asterisk-Users] ASTCC: How to reset periodically all card in use flag back?

[Nicolás Gudiño]

Hi Ronald,

If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.


You should install php-pcntl (or compile php to add support for
process control functions). The inuse problem will be fixed then.

Regards,

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Re: [Asterisk-Users] ASTCC: How to reset periodically all card in use flag back?

[Nicolás Gudiño]

Replying to myself... I was thinking on a2billing, not astcc, so
php-pcntl will make no difference.

The problem might be the same anyways. Asterisk now sends a HUP signal
to every agi script when it detects a hangup. If the script exits when
receiving the signal, it will not handle the clean up routines. The
most recent astcc found in svn includes the code to ignore the HUP at
the top of the script. Be sure to use that version. Regards,

On 6/25/06, Nicolás Gudiño [EMAIL PROTECTED] wrote:

Hi Ronald,
 If a user calls and hangs up before the destination party rings, than
 the in-use flag remains set! This is one case, but maybe there are many
 other cases.

You should install php-pcntl (or compile php to add support for
process control functions). The inuse problem will be fixed then.


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Re: [Asterisk-Users] agi, STREAM FILE and SIGHUP

[Nicolás Gudiño]

  I have developed a custom AGI in C++. Whenever I stream a file or say out
digits with STREAM FILE and SAY NUMBER and hangup the call in between the
AGI ends abruptly.
 I did a bit of surfing through previous posts and found out that asterisk
sends a SIGHUP signal as soon as a caller ends a call. The suggesion was to
catch the SIGHUP signal in the process and ignore it. I wrote the following
piece of code at the star of the agi.


You might want to read this:

http://bugs.digium.com/view.php?id=6491


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Re: [Asterisk-Users] Page cmd FOP

[Nicolás Gudiño]

Hi,

On 6/1/06, Mike Clark [EMAIL PROTECTED] wrote:

We have a location with around 50 Polycom phones. Asterisk version is
1.2.1 We have implemented paging through the Polycoms, which works
great. We are now trying to get FOP  .26 going for the receptionist. It
seems to work fine, except that when someone does and overhead page,
about 3/4 of the phones will continue to show that they are on the phone
after the page is complete and hung up. It clears up for any extension
when they use that phone. Any ideas?


I will need to look at op_server.pl level 1 debug output while doing
the page until the problem shows up to see if it is a bug in FOP or
not. You can send the capture off list to me together with a
description of your problem and a copy of your op_buttons.cfg file.

You can continue asking FOP related questions in its mailing list, you
can subscribe from the webpage: http://www.asternic.org

Regards,

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Re: [Asterisk-Users] Generate two calls from Asterisk and bridge them

[Nicolás Gudiño]

You can also try this patch:

http://bugs.digium.com/view.php?id=5841

On 5/24/06, Arjan Kroon [EMAIL PROTECTED] wrote:

I recommended simple Meetme conference bridge

http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe


Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Álvaro Palma
Sent: woensdag 24 mei 2006 16:36
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Generate two calls from Asterisk and bridge them

Is there a way in Asterisk (I guess there's, it's only I can't figure
out how :-)) to:

1.- Generate a call to channel 1 (example, to PSTN vía an E1 card, using
Zap/g1)

2.- Generate a call to channel 2 (example, an internal SIP extension).

3.- Once both channel have answered, connect the call between them.

This way, I can, for example, play audios in both channels before they
are connected between each other.

So my question is: Does anybody figures out a way to do this? If I use
Manager/Originate, the call necesarily needs a channel to be picked up
(the originating channel) before the call can be placed. What I'd like
to do is:

Asterisk - Channel 1 and do something in channel 1
Asterisk - Channel 2 and do something in channel 2

Bridge both channels: Channel 1  Channel 2

Is maybe Local the solution?

Thanks a lot for your help.

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Re: [Asterisk-Users] Announcement: FOP 0.26 released

[Nicolás Gudiño]

I know that this is a more strictly FOP related question than Asterisk but
I'd like to know if regexp buttons support a '-' char, i.e.:

[_Zap/1-.*]
...

In fact, I have:
 Zap/1 to Zap/10 as incoming channels
 Zap/11 to Zap/15, Zap/17 to Zap/21 as outgoing channels
(it is an E1 PRI)
and I'd like to differentiate them in an easier way than fully listing.


The only reserved characters for regexp buttons are underscore and
ampersand. Any other character should be treated as per PERL regular
expressions. Anyways, your milleage might vary, you need to try it
out.

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Re: [Asterisk-Users] Announcement: FOP 0.26 released

[Nicolás Gudiño]

I am using amp since lasto August, and I am happy with it and its new
version FreePBX

Unfortunately, in all the asterisk servers I installed so far (about 10) I
was never able to make FOP correctly running.

I see the extensions, I see the queue, sometimes I also see the trunks, if
they are zap or iax or sip.

But I ever saw a custom misdn trunk, or  an extensions speaking, nor I
obviously succeded in passing calls.

So, of course, I am sistematically doing something wrong, but all the other
things work fine.

I checked all the README and so on, and everithing seems to be OK.


Well, FOP is a standalone product. FreePBX (former AMP) has a script
that automatically generates FOP's config files (as well as Asterisk
conf files).

If AMP config generator is not good enough for your requirements, you
should edit the config files manually. You have the *_custom.cfg files
in FreePBX to do just that.

Some channel drivers have their particular way of working, like
OH323 and MiSDN.

For monitoring MiSDN channels you will have to use CLID buttons, you
can use the regular channel names but it will monitor in one direction
only due to the way isdn works and how the channel names are crafted.

Read the documentation, not the readme, at http://www.asternic.org to
get an idea of the button types and how to configure them. There is
also a low traffic FOP mailing list you can subscribe from the same
page.

Regards,


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[Asterisk-Users] Announcement: FOP 0.26 released

[Nicolás Gudiño]

I'm pleased to announce that Flash Operator Panel 0.26 has been released!

FOP is a GPL'd switchboard type application for the Asterisk PBX. It
runs on a web
browser with the flash plugin. It is able to display information about
your Asterisk box in real time. It is included in FreePBX,
[EMAIL PROTECTED], DeStar, startShop, and several other projects both free
and commercial. You can grab the latest version from
http://www.asternic.org

The (incomplete) list of new features follows:

   * DTHML client: There is now a DHTML client included. It is
actually an hybrid client where real time communication is handled by
an invisible flash movie and the presentation is done entirely using
DHTML/CSS and Javascript. It allows you to develop completely
customized panels using well known web technologies. No need to learn
ming/flash! The current version allows you to monitor status, not
perform actions.

   * Eye candy: it is possible to specify individual button
backgrounds with external .jpg files. Together with
enable_label_background and no_rectangle you can have nice results
like in the live demo. You can also scroll when you have too many
buttons, just move your mouse to the right edge and see it in action.

   * Transfer directly to voicemail: you can now drag a phone over
the MWI icon and it will transfer the call directly to the voicemail
extension for that button (if defined). Use VoiceMailExt in each
op_buttons.cfg entry.

   * Callerid Privacy per button: first pass at enabling individual
button clid privacy. Just set Privacy=true in op_buttons.cfg for each
button you want to protect.

   * Improved agent status: you can watch agent status more
accurately, including Paused agents. Set agent_status=1 in
op_server.cfg.

   * Several bug fixes, internal refactoring, profiling and optimizations.

The upgrade instructions are on the tarball UPGRADE file. Remember to
upgrade the .swf file and to flush your browser cache!

Many thanks to everyone who provided feedback, patches, ideas and suggestions.


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Re: [Asterisk-Users] Switchboard solutions, interactions with handset

[Nicolás Gudiño]

On 5/4/06, Arnar Birgisson [EMAIL PROTECTED] wrote:

Hi there,

I'm looking into developing an in-house switchboard application. Does
anyone here know of a way to control a hard-phone from such an
application.

For example, the attendant forwards a call with another one in queue.
Once the first call has been forwarded (by keyboard shortcuts or
dragging-n-dropping) - she presses a button (on the computer) to
answer the waiting call.

Now, if the switchboard application embeds a soft-phone, I can figure
out how to do this. But suppose the attendant is using a hard-phone
(since it's more reliable) with a headset - can she do the above
things without having to press any of the phones buttons?

Wouldn't this require the application to somehow control if the phone
is off-hook or on-hook? Is there some other way I'm not seeing and/or
has someone here implemented similar stuff?

Could I possibly keep an open channel in Asterisk to the attendants
phone, and bridge that with whatever channel requested by the
switchboard application? I have found some mention of this, bridging
channels, in the mailing list archives, but not in the AMI
documentation. Is this maybe something that's still only on the svn
trunk?


I have done something similar using a modified Flash Operator Panel
and a phone with autoanswer capabilities (polycom 501), while the
operator is using a headset. Then you can use standard manager actions
to redirect calls to the operator. Regards,

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Re: [Asterisk-Users] Flash Panel / Queue Slots

[Nicolás Gudiño]

 is there any way to make the Flash Operator Panel show which agents are
 logged in in a specific queue? (both static and dynamic agents)

 I've played around with the queue / queue agents settings from the Flash
 Panel documentation (http://www.asternic.org). The way it is described
 there, I could only make the Flash panel show that a queue 8in general)
 received a call from a specific extension.

If you double click on the arrow for a queue summary button you should
see some stats, including agents logged into that queue.

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Re: [Asterisk-Users] change/toggle flash operator panel components

[Nicolás Gudiño]

 Hi,
 is it possible to remove the no timeout combo box in flash operator panel?
 How can I reduce the flash area? I set small buttons and half of the
 area is white and I want to resize it.

Comment transfer_timeout in op_server.cfg. To reduce the flash area
you will have to play with the .html file on the flash embedded
object. Regards,

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Re: [Asterisk-Users] Dial from php

[Nicolás Gudiño]

 Flash Operator Panel already has similar functionality, just create a
 CID entry drag and drop.  There may of course be other (better) ways to
 do this but this is one option/alternative.

You can also use FOP and javascript to initiate a call to the number
entered on a text input box, an href link, etc.

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Re: [Asterisk-Users] Reporting?

[Nicolás Gudiño]

 NICE!

 On 3/30/06, Joe Dennick [EMAIL PROTECTED] wrote:
  I see (and like) the demo, but where can we get it?
 
  Doug Lytle wrote:
 
   Nicolás Gudiño wrote:
  
   shameless plug Something like this perhaps?
  
   http://www.asternic.org/stats/demo

It is not released yet... I'm not having much time to write the web
page, documentation, tarball, etc.

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Re: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk

[Nicolás Gudiño]

Well, the Flash Operator Panel supports barge in too, with the option
to barge muted so the people involved in the conversation won't notice
the interruption. And then the supervisor can drop one of the channels
or mute/unumute them. But it uses meetme, as well as all the other
manager applications that supports this.


 VICIDIAL supports this when using VICIDIAL for inbound and/or outbound 
 calling.
 Blind monitoring, barging in on the call and hijacking the customer
 from the agent.

  I have been doing some work with the Asterisk Management API and there
  is a commadn where you can transfer a call. This is what you may be
  looking for
 
  Not sure, trying to be as helpful as I can
 
   -Original Message-
   From: Cory Andrews [mailto:[EMAIL PROTECTED]
   Sent: Tue 3/28/2006 9:59 PM
   To: asterisk-users@lists.digium.com
   Cc:
   Subject: [Asterisk-Users] Call Monitoring / Call Takeover with 
   Asterisk
  
  
   Does Asterisk support, in a call center type environment, the 
   ability for a supervisor to monitor a call between a system user and a 
   3rd party, and allow them to physically take over the call.  For instance 
   if a call center supervisor is randomlay monitoring agent calls, and for 
   some reason need to intervene on a call without first having been 
   conferenced into the call?

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Re: [Asterisk-Users] Reporting?

[Nicolás Gudiño]

shameless plug Something like this perhaps?

http://www.asternic.org/stats/demo


On 3/29/06, Matt [EMAIL PROTECTED] wrote:
 Is there anyway in asterisk to figure out how much time an agent has
 spent on the phone?  I know I can see total time for a call (inbound
 or outbound) but where/how do I view queue stats?

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Re: [Asterisk-Users] OT: FOP and reverse_transfer

[Nicolás Gudiño]

 When I drag and drop a call from the PSTN to a SIP phone (the SIP phone's
 icon) using FOP .25 with * 1.0.9 with the intent of transferring it, the
 called party gets transferred rather than the calling party. This is
 controlled by the reverse_transfer parameter in op_server.cfg but the
 behavior is exactly the same whether the parameter is set to 0 or 1. This is
 after the call is picked up by the transferring party. If I drag and drop
 the call while it is in the ringing state, the transfer works correctly.

 I have tried hardcoding reverse_transfer = 1 in op_server.pl:


 if ( !defined $reverse_transfer ) {
 $reverse_transfer = 1;
 }

 and commenting out reverse_transfer in op_server.cfg, no effect.

 Anyone else seen this?

Nope, I have not seen it. I would need to look at a full debug (255) 
from op_server.pl. Send that file to me off list together with all
your .cfg files. Please address any future question about FOP itself
to FOP mailing list (you can subscribe from http://www.asternic.org )

Best regards,

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Re: [Asterisk-Users] pickup a call in queue

[Nicolás Gudiño]

  When we have several calls in different queues, is there some sort of
  way to open a channel between a (sip-)phone and a SPECIFIC call in a
  queue using the Asterisk manager api?
 
  We would like to do this even when we are not a member of that specific
  queue.
 
  Thanks in advance for any suggestions!
 If you have FOP, and if the call come in thru a ZAP channel, you can
 drag the ZAP channel to your extension. This should work.

In the latest version (0.25), you can have buttons for each queue
position showing clid number and clid name, and you can drag them to
any other phone you want... no need to look into the Zap buttons.. and
it also shows individual timers for each person waiting on that queue.

It uses the manager redirect command

Regards,

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Re: [Asterisk-Users] mpg123 alternative?

[Nicolás Gudiño]

Hi Rich,

 Been using mpg123 for moh for the last two years or so. However, when
 I have * config errors, often times get a endless stream of console
 messages and need to kill the two mpg123 processes.

 Is there an alternative to mpg123 that eliminates that issue?

 I see references in musiconhold.conf relative to madplay, native file
 format, asterisk-addons, etc. Not sure why the asterisk-addon approach
 hasn't been moved into trunk, or if madplay is a better choice on this
 fc3 trunk box.

 Any suggestions?

I've switched to native moh and never had to worry again about dead or
unresponsive mpg123 processes.

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Re: [Asterisk-Users] Install instructions for FOP Flash Operator Panel do not make sense...

[Nicolás Gudiño]

Hi,

 Anyone got AFOP working. The install instructions tell you to copy all
 of the files extracted under the 'html' directory to a subdirectory
 under your main web directory (in my case this is /var/www/html/panel/)
 and then the instructions talk about modifying the 'op_server.cfg' file
 but they do not tell you were to put this file. There is something wrong
 with the instructions???

English is not my first language, so the instructions might be hard to
understand. If you are reading the README, it is because you already
extracted the tarball. You can place all files wherever you want: a
nice place is under /usr/local. I usually use /usr/local/fop. , the
only thing that is mandatory is to move the html and swf files
somewhere into your webroot for an obvious reason.

All other files can live anywhere.. and the .cfg file is just in the
same directory as the README, op_server.pl, etc. Anyways, FOP's
mailing list is a better place to seek for help about it, as is not
directly asterisk related.

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Re: [Asterisk-Users] Queues and Agents

[Nicolás Gudiño]

 Is it possible from within the dialplan to determine if an Agent channel is 
 already a member of
 a queue?  Would like to use this as part of a check that will play a message 
 if the agent is the
 last person to log off the queue.

 I can sorta do it by using AddQueueMember and checking ${AQMSTATUS}, however 
 that generates a log
 warning if they are already a member.  I would also have to sometimes run 
 RemoveQueueMember real
 quick to remove people who shouldn't be a member after doing that.

 Any ideas?  Another idea I have it so make a fast AGI script that parses data 
 from a mamager
 connection to see, but this seems kinda overkill for something that should be 
 simple.

Maybe by using persistentmembers = yes in queues.conf and then query astdb:

Ffrom the cli: database get /Queue/PersistentMembers yourqueuename

Or from the dialplan using DB function or DBGet application depending
your asterisk version.

Regards,

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Re: [Asterisk-Users] Queues and Agents

[Nicolás Gudiño]

 I see what you mean and already have the option turned on.  However the 
 entries in astdb
 are a bit odd:

 //Agents/40042: [EMAIL PROTECTED];4004
 //Agents/4005 : [EMAIL PROTECTED];4005
 //Agents/4011 : [EMAIL PROTECTED];4011
 //Agents/4014 : [EMAIL PROTECTED];4014
 //Agents/4025 : [EMAIL PROTECTED];4025

Those entries are for agentcallbacklogin (you are seeing the
[EMAIL PROTECTED] where they can be reached, but not queue membership
status), my suggestion was to use persistenmember and the key
/Queue/PersistentMembers for dynamic added members via AddQueueMember,
together with some string manipulation functions for checking if an
agent is a member of a particular queue or not. It is maybe too
hackish, but doable.

Regards,

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Re: [Asterisk-Users] FOP button limit?

[Nicolás Gudiño]

 50 extensions, 27 trunks, 1 queue, any tips would be great appreciated,
 -Kerry

Inside op_style.cfg:

btn_width=191
btn_height=30
btn_padding=5

Then tweak all the scales and margin parameters for the icons. It
would give you all the buttons you need an a couple more.

You can direct all this questions to FOP's mailing list, you can
subscribe from http://www.asternic.org or browse the archive for some
style examples. Regards,

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Re: [Asterisk-Users] Server Side AgentCallbackLogin

[Nicolás Gudiño]

On 11/22/05, Jason Lixfeld [EMAIL PROTECTED] wrote:
 Here's what I'm trying to do..  We have a small system, there are
 only two of us.  We both do sales and we both do support.  We like
 Queues better than music on hold with a bunch of dials happening in
 the background to try our phones, then cells, etc.  Problem is, we
 don't like the idea of having to login to a queue and are wondering
 if there is a way to force/automatically log agents into a queue
 without having to do anything on the phone; have it be server side
 that is.  I'm thinking some sort of cron job that runs every minute
 or five to make sure all expected agents (my partner and I) are in
 the queue and if not, log us in.  The extentions we use to enter the
 queue are find-me extensions so if we aren't at our desks, calls will
 hit our cells.

Add static members into the queue in your queues.conf entry. You can
use Local channels to find your follow-me [EMAIL PROTECTED] Like:

[myqueue]
music = default
strategy = ringall
timeout = 20
member = Local/[EMAIL PROTECTED]


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Re: [Asterisk-Users] One Touch Record in 1.2

[Nicolás Gudiño]

 I have been trying to find more information on the One Touch Record
 feature in 1.2 (features.conf) but have not been very successful.

 Basically, I've been trying to get more information as to:
 1) Do I need to specify any particular option in the Dial command

yes

w  W (for enablig caller  calle)

 2) How can I customize the location of the recorded file(s)

I don't know if you can change the location, I think not. You can
somewhat customize the file name setting the variable TOUCH_MONITOR.
You can set the format setting TOUCH_MONITOR_FORMAT, by default is
.wav

The name of the file will be
auto-{TIMESTAMP}-{CALLER-CLID}-{CALLEE-CLID} by default and
auto-{TIMESTAMP}-{TOUCH_MONITOR} if TOUCH_MONITOR is set.

 3) Will the files be soxmix'ed together or not

yes

 4) How to use it in general

Just dial the sequence specified in features.conf to start/stop the
recording. By default is *1.

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Re: [Asterisk-Users] Webui to show registered phones

[Nicolás Gudiño]

 Hi all, does anyone know if there is any app/webui that can show phones
 that are currently registered to *.  I guess this sort of funcionality
 counld be grabbed from the CLI with iax2 show peers and sip show peers,
 but having little programming knowledge wouldn't know where to start.

 I'm asking because we currently have several sip phones onsite and lots
 of remote iax2 users who would like to see availability without dialing.

plugYou can try with the Flash Operator Panel/plug
http://www.asternic.org , it does all sort of things including sip and
iax availability (you have to enable qualify for them). Regards,

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Re: [Asterisk-Users] ASTCC and Asterisk 1.2?

[Nicolás Gudiño]

 Does everything with AstCC work properly under Asterisk 1.2?

Yes. But checkout astcc again because it got patched to fix a bug with 1.2

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Re: [Asterisk-Users] Fwd: ASTCC - INUSE Flag

[Nicolás Gudiño]

 Hi all. Just to update list and increase the souls-save database.

 The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with
 asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working
 fine with ASTCC and inuse flag.

 The link of the patch is:  http://bugs.digium.com/view.php?id=5400

Glad to save your soul... Best regards,

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Re: [Asterisk-Users] ASTCC - INUSE Flag

[Nicolás Gudiño]

 It works.
 I terminated the call during the playback.

 AGI debug
 AGI Tx  200 result=-1 endpos=480

 HUP received!

 Allowing

 setinuse() to get called

Please add your comments on Mantis so it gets commited quickly.

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Re: [Asterisk-Users] ASTCC - INUSE Flag

[Nicolás Gudiño]

On 10/5/05, Darren Wiebe [EMAIL PROTECTED] wrote:
 Any developers out there that would like to look at this one?  It works
 fine on Asterisk CVS-v1-0-09/22/05-22:23:34 on a i686 running Linux but
 it does not work on the 1.2 betas.  I agree that the number should be
 set aside then.  I wonder what the problem is.


http://bugs.digium.com/view.php?id=5400

Seems to fix the problem... please test and give feedback.

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Re: [Asterisk-Users] C Manager Interface Client

[Nicolás Gudiño]

 Below is the code that we have. We are getting ready to run a sniffer
 and see if/why asterisk is doing the writes separately instead of in one
 chunk.

There were some changes in CVS that appear to address this issue.
However, if you are trusting the manager to write full events for your
application to work, then expect it to break when the manager does
(and I warranty you that it will eventually happen, as the manager
protocol does not mandate that the events should be written by whole
chunks). You should make your application resilient to those changes,
or to work even with partial writes. Regards,

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Re: [Asterisk-Users] Flash Operator Panel Help

[Nicolás Gudiño]

Hello,

On 9/17/05, Insider KT [EMAIL PROTECTED] wrote:
  
 Hi. I am using the Flash operator panel 0.24 and it works, but I don't see
 the voicemail icon when I have incoming voicemail. 
   
 In the op_buttons.cfg I have the following setup: 
   
 [SIP/100] 
 Position=2 
 Label=Office tel. 1 
 Extension=100 
 Icon=1 
 Mailbox=100 
   
 I've tried to google on the subject, but have not found any answers. 
 I've tried [EMAIL PROTECTED] also. (full = the context for the 100 extension)

Is full the voicemail context or the extensions.conf context? If you
are using a standard asterisk setup, the mailbox should probably be
[EMAIL PROTECTED] (being default the voicemail context for that extension,
as specified in voicemail.conf)

Regards,   

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[Asterisk-Users] Announcement: FOP 0.23 released

[Nicolás Gudiño]

Dear all,
 
I'm happy to announce the Flash Operator Panel 0.23 release. FOP is a
switchboard type application for the Asterisk PBX. It runs on a web
browser with the flash plugin. It is able to display information about
your Asterisk box in real time. It is included in AMP, [EMAIL PROTECTED],
etc. You can grab the latest version from http://www.asternic.org
 
The incomplete list of new features is:
 
* Internationalization support (thanks to everyone who contributed a
language file! If your language is missing, please contribute with the
translation. It is a small file, a couple of minutes worth of your time)
 
* Command line options. You can specify the logdir, pidfile, debug level
and much more from the command line.
 
* The web_hostname parameter is now optional. It eases the installation
a lot! All systems that include FOP installation and configuration
scripts can now leave the field commented, and the client-server
connection will just work(tm). No need to fiddle with ip addresses,
hostnames, etc.
 
* Popups via UserEvent can be restricted to one button/viewer only.
 
* Added font and shadow color parameters for button labels, text
legends, clid and timer.
 
* Added event_mask parameter to filter unwanted events from the manager

* Improved debian init script. Thanks to Tzafir Cohen.
 
* It uses a lot less CPU than previous versions on heavy asterisk boxes
 
* Improved support for parking when using native sip transfers
 
* Minor bugfixes
 
* A stupid buglet that I don't know yet about, that will force me to
release 0.24 in a short while.
 
The upgrade instructions are on the tarball UPGRADE file. Remember to
upgrade the .swf file and to flush your browser cache!
 
Many thanks to everyone who provided feedback, patches, ideas and suggestions.
 
I'm also saving money to attend to Astricon Fall. Please consider a
small donation to help me cover the travel expenses. (my deepest thanks to
everyone who already donated to the project!)
  
PS: Just to let you know, I'm playing now with new tools to develop flash
clients: swfmill and mtasc. They are great tools, it will ease and make
the development faster. But the port is not easy, I will work slowly on
that. If you are a flash OOP actionscript fan and want to help, please
let me know.  

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Re: [Asterisk-Users] XML Revisited

[Nicolás Gudiño]

Hi Anton,
 
 I recently contacted polycoms tech support asking if their phones supported
 XML pushed information to which they replied that only model 600 had a
 microbrwoser capable of reading dhtml files and such.
 
 My question to the community is: is somebody doing any XML info push to any
 brand of phones except Cisco? How are you doing it?
 
 One of the wonders of VoIP should be the means to send information back to
 the phone which ould be displayed on those wonderful screens that they have
 :) besides showing callerid and time which normal phones do..
 
 Any ideas/comments?

I wish I had a Polycom 600 to try. ;)  Regards,

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Re: [Asterisk-Users] FW: Asterisk-panel

[Nicolás Gudiño]

Hello,On 8/18/05, Soner Tari [EMAIL PROTECTED] wrote:
It sounds like web_hostname in your /var/www/html/panel/op_server.cfg is setto your external ip. If you change it to your internal ip, I think you'llhave the opposite of what you describe. I couldn't find a decent solution to
this dilemma. Any one?
In the latest FOP snapshot the web_hostname parameter is optional: you
can comment it out and it will just work from the internal an external
net.

Regards,
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Re: [Asterisk-Users] Queue/Agents

[Nicolás Gudiño]

 Looking for a good web app that will show agents that are login to
 queue. I tried the operator panel but I'm unable to get the LED to
 change color per the doco I have.. It works well for everything else but
 no luck on the agent part..

How are your agents loging into queues? Depending on that you should
use slightly different configurations. Contact me off list if you need
assistance.

Regards,

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Re: [Asterisk-Users] Set syntax equivalent of DBDel?

[Nicolás Gudiño]

 I found an unanswered mail in the archives that implied that perhaps
 there is no direct way to delete a DB entry with the new Set syntax.
 
 Set(DB(family/key)=) sets the value for the key to null, but that
 doesn't appear to be equivalent to removing the key entirely.
 
 Or maybe DBDel isn't deprecated, like the other two are.
 
 Anyone know the score?

I was wondering the same thing myself... I guess dbdel is not
deprecated, and that is confusing because dbput is deprecated. Maybe
this should be posted to the -dev list because I don't think that
-users is being heavily monitored by Digium.

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Re: [Asterisk-Users] Getting FOP working with ICD?

[Nicolás Gudiño]

Hi,

 Hello my name is Axel Pache and  i and some kolleges are working on a
 callcenter solution. We use ICD to manage skill based routing. But now
 we got some problems integrating FOP, for example FOP  doesnt
 acknowledge the ICD-queues right. I have to  use a normal  asterisk
 queue to get  FOP working with it.
 
 So my question is:
 
 Is there any way to get FOP working together with ICD in that case?

If ICD does not use regular queues, it will require custom development
or a clever use of UserEvent in your dialplan. BTW, my rates are
cheap.. :) Regards,

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Re: [Asterisk-Users] Monitoring Sirrix quad BRI channels

[Nicolás Gudiño]

 Is there a way for me to individually identify each BRI channel on the
 Sirrix quad BRI board. 
   
 The reason I ask is because our client uses the Asterisk Flash Operator
 Panel to monitor its external lines and transfer calls from the lines to
 the various SIP phones. 
   
 The Flash Operator Panel requires that we set a static value for each line
 or channel. With analogue cards its easy as the lines are Zap/1, Zap/2,
 Zap/3 etc. With the Sirrix board the value seems to change: 0814f1f8,
 08129f38, 0837ad40. 
   
 Is there anyway I can get this right so that each channel (8 of them) can be
 monitored ? 

You can try by using regexp to match the channel name, prefix it with
an underscore and then a perl regexp pattern. Regards,

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Re: [Asterisk-Users] CRM integration (was RE: CallerID)

[Nicolás Gudiño]

  If it was for pure HTML only, yes, you are
  correct. But with javascript you can start a timer and
  execute a javascript function every once in a while. If this
  javascript loads an XML document off the server, you're there ;)
 
 So you have now instructed the browser, via javascript, to periodically
 poll the server every once in a while.
 This is exacly what the previous poster (the one I replied to) was
 trying to AVOID, and for good reason. It doesn't scale. In order to be
 effective as a way to present the user with caller-ID driven data, it
 would have to poll quite frequently.  With a handful of clients
 constantly doing this, the impact is inconsequential, but as the number
 of clients hammering the server in this manner climbs, things are going
 to break.

Well, I have to chime in here. The Flash Operator Panel can do this
without polling. The downside is that you need the browser window open
for it to work. If you don't want to display any buttons, you can
embed the flash client on a hidden frame or iframe, no problem. You
can also use it for adding 'click-to-dial' capabilities to your web
application. Newbie alert: you have to configure several files to
achieve this, and you must know a little javascript and/or php.

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Re: [Asterisk-Users] Manager Port

[Nicolás Gudiño]

 I am using flash operator panel, when i stop iptables everthing is fine, but
 once iptables is started, the operator panel doesn't work anymore. Anyone
 know how to set up the iptable in order for to op panel to work? I am using
 tcp port 5038 for asterisk manager, and I have try open both tcp and udp
 port 5038 in iptables but without success.

Open up port 4445 tcp. Regards, 

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Re: [Asterisk-Users] Fail over solutions

[Nicolás Gudiño]

 The disk array would be the only expensive add on, more than a normal
 asterisk system.  It all depends on how important voicemail is in your
 application, although there are cheaper alternatives (NFS for example,
 but then your NFS server becomes a single point of failure, depending on
 the disk array that same issue could be true there as well).

If you are on a budget, I would suggest to look at a drbd+heartbeat
combination. DRBD is a block device which is designed to build high
availability clusters. This is done by mirroring a whole block device
via (a dedicated) network. You could see it as a network raid-1.

Regards,

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Re: [Asterisk-Users] Redirect two channels to each other?

[Nicolás Gudiño]

 It almost sounds like there needs to me a new manager action:
 
 Action: Bridge
 ChannelA: SIP/199testfone-1f3c
 ChannelB: Zap/6-1
 
 It sounds like the intrinsic functionality for 'bridging' is already there in
 Asterisk (duh!), it just needs to be encapsulated in a manager action.

Yes, we need that action on the manager! (but replace ChannelA and
ChannelB to Channel1 and Channel2 as on the link event).

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Re: [Asterisk-Users] using * for Internet call waiting

[Nicolás Gudiño]

On 4/21/05, Gary Carr [EMAIL PROTECTED] wrote:
 Wondering if it is possible or if something already exist to setup * to
 offer Internet Call Waiting. For those that do not know what it is, it's a
 small application that runs on a users computer that will pop up a window
 letting them know they have a incoming call and who it is from then they can
 choose to take the call which will disconnect their dialup modem and ring
 their phone or send the call to voice mail.

You need a V92 capable modem for your client and a V92 capable access
server for you.  The feature is called modem on hold, it lets you
pick up a call without loosing your internet connection, and resume
the dialup session after hangup. The only feature you need for your
telco is call waiting. It does not need forward on busy. Regards,

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Re: [Asterisk-Users] inquire about connected channel (show channels)

[Nicolás Gudiño]

 I know this information can be parsed out of show channels
 but I was just wondering if the is an easier way?

Its easier to use 'Status' on the manager directly. Regards,
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Re: [Asterisk-Users] how can i connect a cost display on asterisk

[Nicolás Gudiño]

 Is it possible to connect a display that shows the costs of a call in
 progress? 

We are doing a 100% asterisk based software/hardware solution for
callshops. The hardware part consists of little boxes with an LCD that
displays information on the call, including number/name of dialed
number, duration, cost, etc, all in real time.

The boxes are connected to a 'master box' that at the same time is
connected through the serial port to the asterisk server. It does not
use polarity reversals or tones from the phone company, but just
asterisk status, so it can work in a pure voip environment, without
regular pots lines.

The operator has a console where he can view the status of each box
(using my Flash Operator Panel), including the cost for the call when
it ends, and the latest CDRs for every phone booth. It can
enable/disable phone booths, manage rates, etc.

The product is not yet finished, but its progressing nicely. If you
are interested in see how does it looks like, drop me a note off list.

Regards,

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Re: [Asterisk-Users] Flash pannel: time display

[Nicolás Gudiño]

On Tue, 22 Mar 2005 14:16:54 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote:
 I have three different time displays:
 
 Flash panelcaller 615 48:00
  called 62058:18
 Snom phone shows for the same call   47:55
 
 Why is there a difference at all?

If you reload the panel, it will take the timer directly from
Asterisk. You can do a '
show channel SIP/xxx'  from asterisk's cli during the call and see
what does it say in Elapsed  Time.

It might be: 1) FOP bug, 2) SNOM bug, 3) Asterisk bug, 4) clock drift
on your server, 5) clock drift on your phone.

I had reports of bad timers within FOP but I was not able to fully
reproduce the problem. You better check from the cli and compare that
with FOP display or your SNOM phone, and tell us your results.

Regards,

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Re: [Asterisk-Users] Dial from a URL - Possible?

[Nicolás Gudiño]

 Is it possible to initiate/receive calls from a url (that is without
 having to install and configure a PC soft phone) using asterisk?
 If yes, may I please get some sites, pointers, HOWTOs on how its done?

You can also try the Flash Operator Panel, http://www.asternic.org. It
supports click-to-dial, screen pops, etc.

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Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons?

[Nicolás Gudiño]

Hi Ronald,

 I have setup flash pannel, ... looks nice, but so far I could not
 configure it to get more than 4x7 buttons.
 I tried to make the buttons smaller, but than just the entire picture is
 smaller.

What did you change in op_style.cfg? You can have literally hundred of
buttons per screen, or multiple 'context' to split your buttons into
several screens. I wll send you an alternate op_style.cfg with smaller
buttons offlist. Regards,

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Re: [Asterisk-Users] About shadydial

[Nicolás Gudiño]

Hello,

 I need to know the following: when a call is answered, this call es sending
 a to agent, but I need that the agent when receive the call in the desktop
 appear all information of this number.

You can do this with the Flash Operator Panel. http://www.asternic.org

Regards,

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Re: [Asterisk-Users] Transferring calls into MeetMe

[Nicolás Gudiño]

Hello,
 
 I posted earlier with regards to three way calls and X-Lite, this kind of
 yielded everything I already suspected.  However I suspect someone has a
 good working config for connecting a third party to an existing call
 (a-la-skype), or a detailed solution of using MeetMe to achieve this,
 without having to make two calls, transfer them in, then connect my self. 

Maybe is not what you want, but anyways.. . you can try with the Flash
Operator Panel (http://www.asternic.org). It can do barge-ins using
meetme for that purpose. Regards,

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Re: [Asterisk-Users] How to see ExtensionStatus in manager

[Nicolás Gudiño]

 I try to see ExtensionStatus (event) when I'm logged on manager. But
 nothing :/
 This is implemented in manager.c. May be I compile my astersik with out
 a parameter ?

You have to use the hint priority in your dialplan. Then the
ExtensionStatus will work.
http://www.voip-info.org/wiki-Asterisk+standard+extensions

Regards,

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Re: [Asterisk-Users] Print-to-Fax client

[Nicolás Gudiño]

 What about a driver that will send the print out to Asterisk, on the same
 network to be sent as Fax ? 
 Is there anything that already exists for this?

Hello,

Several months ago I worked on such a solution using salsafax. The
problem was on how to determine the fax number to send the fax to. I
tried with OCR but had a 60% success rate extracting the number. It
was cool for me but not good for a bussiness.

FYI, salsafax is a script for use with Samba and CUPS/Lpr. Basically
you export a printer to the network, and then you can setup that
network printer in your windows/samba clients and print to it. Then
you have to convert the postrcript file to .tiff to be used by txfax.

Another problem is that I do not know if spandsp can return the status
of the fax after it is sent, so you know if it was received ok or not.

Regards,


 Quoting Florian Overkamp [EMAIL PROTECTED]:
 
  Hi,
 
  -Original Message-
  You should be able to download one (for WIndows and possibly Mac) from
  efax or j2.com I think.
 
  http://www.efax.com/en/efax/twa/page/download?rqcp=2
 
  http://www.j2.com/jconnect/twa/page/download
 
  You might be able to do that, but take a good look at the license agreement
  on the driver - you might not be allowed to use the software fully without
  having a subscription to their services.
 
  Florian
 
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Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable

[Nicolás Gudiño]

 Paul, 1.0.5 stable suffers from caller id issues as well, at least for
 SIP channels. What fixed things for me was swapping in app_dial.c from
 1.0.2 stable (didn't try others). You could also just diff app_dial.c
 between versions to find the problem but I took the lazy way out the
 first time around.

Drumkilla reverted the callerid changes on the latest stable (thanks
Russell!). You will be fine if you checkout stable from CVS now.
Regards,

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Re: [Asterisk-Users] really easy FOP asterisk@home question

[Nicolás Gudiño]

Hello,

 
 I deleted the config examples in the op_buttons.conf folder for how to set
 up the meetme representation 
[skip]   
 
   
 
 [Meetme/801]; Meetme must be defined by its room number 

change the above line to:

[801]


Regards,

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Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVS Stable

[Nicolás Gudiño]

Hello,

 2 nights ago I upgraded one of my remote servers to the latest CVS Stable,
 Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID
 stopped working.

My suggestion would be to downgrade to 1.0.3. It might solve your
problem. There were a number of changes in callerid handling in the
last couple of weeks.  Many manager based applications stopped working
because of them. Maybe your setup is affected too. Regards,

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Re: [Asterisk-Users] live monitoring (SIP only)

[Nicolás Gudiño]

Hello,

 is it and how is it possible to live monitor (barge - in) a SIP to SIP call
 without 
 any Zap Interface? I am using asterisk 1.0.5 with chan_capi from Junghanns 
 and SIP clients. I was looking for chan_spy application but it seems to be 
 no longer available. 

You can do something like this with the Flash Operator Panel (
http://www.asternic.org ). chan_spy would be a better option because
you can use it from the dialplan. As a workaraound, FOP lets you drag
your phone to a bridged call and put the three in a meetme room, with
the option to start the 3rd led muted so the other won't notice  the
interruption. Regards,

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Re: [Asterisk-Users] Callerid problems with 1.0.5

[Nicolás Gudiño]

Hi,

On Fri, 4 Feb 2005 21:35:19 -0500, mattf [EMAIL PROTECTED] wrote:
 Hello,
 
 patching v1.0.5 on my system removed the problem for me. But yes it seems
 strange that this feature was inserted into a final release with very little
 documentation of the wide implications that are caused by the change.

You can use a previous revision of app_dial.c too.

 
 This was corrected in CVS with the addition of a diabling flag for the dial
 command, but maybe this is a message that we should start an official beta
 release period before a release so that people can test pre-releases even
 for just a week to report problems before it is unleashed upon the world as
 an official release
 

I agree... I believe drumkilla is doing a great and hard work
maintaining stable (a big thank you for that!), but  I think it was a
mistake (or overlook) to backport the callerid 'bugfix' to stable. It
is not really a bugfix, but a design change that in fact disrupts many
working installations.  If I were responsible for that, I would
release 1.0.6 (just like 1.0.5 had to be released also because of
callerid problems), without that change applied.

And I'm also on the opinion that the 'o' flag in CVS-HEAD is not
really the solution to the problem, because if Mark thinks that the
previous handling of callerid has a logical error, if you use the 'o'
flag in dial, your callerid will probably work, but it will have that
'logical error' as Mark's opinion.

The real solution would be to fix the 'logical error' and not brake
the callerid in many situations.  I thought it only affected the
manager interface, but it seems that its not only limited to the
manager as these thread and many bug reports point out.

I'm probably not seeing the whole picture, but the callerid is really
not that hard. If you are receiving a call, the callerid should be the
remote callerid. If you originate a call, the callerid should be your
callerid. And if I want to preserve the original remote callerid when
doing a transfer, or being parked, or whatever (I consider making a
transfer to be the originator of a call), you can use a dial flag (I
believe its 'p' in cvs-head?), so the recipient of the transfer will
see the original remote callerid instead of your own.

Regards,

-- 
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Re: [Asterisk-Users] Re: astGUIclient users should not upgrade to Asterisk 1.0.5

[Nicolás Gudiño]

Hello,

 I'm not a astGUIclient user, but I'm puzzled by the following statement:
 
 mattf [EMAIL PROTECTED] wrote:
 
  In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP, IAX2),
  once the call picks up, Asterisk will change the callerid to the number that
  you just dialed, no matter if you set a custom callerID for that call.
 
 What you've said there suggests that the CallerID is being set to the
 DESTINATION number, which sounds to me not what CallerID should be at all.
 CallerID normally indicates the source of a call.

Just wanted to say that Flash Operator Panel users will have the same
problem. I'm puzzled too. IMHO there's something missing or wrong in
the new callerid handling.

If you trace the manager events and try to match the callerid via
Uniqueid, you will notice that the only way to have a match is *after*
the call is bridged. That means that you cannot  find the callerid of
a call before you pick up the phone. At least thats what I'm seing on
Asterisk 1.0.5. (did not try with HEAD) So, the callerid is plain
useless (Users expect to see the callerid before picking it up, dont't
they?)

It would be nice to have the callerid available on the manager when a
phone is RINGING and before picking it up.

I did not look at the Local channels, and it seems that it makes
things harder.. but I still think that we do not have to code
workarounds on manager based applications. We need to have an event in
the manager informing the callerid of the caller in the RINGING event
or associated directly with the Uniqueid of the callee.

Personally I had to downgrade app_dial.c to a previous releaes to get
the callerid as before.

Just my 2 cents...


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Re: [Asterisk-Users] Re: API Call Bridge?

[Nicolás Gudiño]

Hello,

 I've tried all the Wiki pages and still can't seem to
 get this thing working and that's why I've posted this
 mail.
 
 I would like to dial two external numbers and
 conference them together using the asterisk api
 manager.

Hint: search the wiki for local channels

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Re: [Asterisk-Users] Operator Panels?

[Nicolás Gudiño]

Hello,

 The problem we're having is transfers don't seem to work? ie: when
 someone calls inbound, you drag and drop the call on the extension you'd
 like and it just bridges the 2 phones together instead of transfering
 the call? Maybe this was intentional or maybe I'm just doing something
 wrong? Other than that the panel seems to work great.

You can set reverse_transfer to 1 in op_server.cfg and it will
transfer the other leg of the call (Ex: if you drag phone A to phone
B, it will transfer the other leg of phone A (maybe an iax trunk or
whatever) to B, instead of dropping the trunk and bridging A with B.

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Re: [Asterisk-Users] Total newbie here looking to do a VoIP conference call?

[Nicolás Gudiño]

Hello,

 I don't have a great grasp as to what Asterick is capable of, but my
 thoughts were that perhaps with VoIP telephone lines (either hooked up to
 the company's network or just using a 3rd party VoIP provider such as
 Packet8, which is whatI have for personal use) and an Asterick server, that
 we could setup a VoIP conference bridge.
 
 Can someone enlighten an unknowledged as to whether or not this is possible,
 and if so, how might it be done?  Would the Asterick server need X number of
 VoIP lines?  I.e. If there's 10 participants, it'd need 10 VoIP lines?

You do not need VoIP lines as you call them... You need an asterisk
server and ip phones or softphones to dial your server conference room
(the application is called meetme)

If you would like to accept regular pstn calls into your conference,
then you will also need some hardware to connect pstn lines to your
asterisk box. There are several kinds of cards you can purchase
them from digium at http://www.digium.com

If you do not have time to set this up, you can hire a consultant. You
can find a lot of usefull documentation, and a list of asterisk
consultants at http://www.voip-info.org

Good luck,

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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 5, Issue 192

[Nicolás Gudiño]

Hello,

 Thank you for your response. I had tried that before and it didn't work. I
 am trying to look up the route for a dialed number, so its a full E.164
 number. Please see my query below when I try to look up the route for a USA
 number;
 
 mysql SELECT * FROM routes WHERE ^13237309880 RLIKE pattern ORDER BY
 LENGTH(pattern) DESC;
 +-++-+--+-+-+--+
 | pattern | country| comment | trunks   | connectcost | includedseconds
 | cost |
 +-++-+--+-+-+--+
 | 880 | Bangladesh | Proper  | Carrier |   0 |  30 |
 0.18 |
 | 237 | Cameroon   | Proper  | Carrier |   0 |  30 |
 0.24 |
 | 32  | Belgium| Proper  | Carrier |   0 |  30 |
 0.06 |
 | 1   | USA| USA | Carrier |   0 |  30 |
 0.04 |
 +-++-+--+-+-+--+

The pattern column should be:

^880.*
^237.*
^32.*
^1.* 

and the query:

SELECT * FROM routes WHERE 13237309880 RLIKE pattern ORDER BY
LENGTH(pattern) DESC

It works.

Best Regards,

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Re: [Asterisk-Users] ASTCC

[Nicolás Gudiño]

Hi,

 I have a problem with ASTCC. When I create all my routes, I not able to get
 the destination pattern I desire. I see it come up, but ASTCC seems to
 select the first available pattern, and not necessarily the exact one I
 want. I  found the MYSQL statement in astcc.agi: 
   
 SELECT * FROM routes WHERE $number RLIKE pattern ORDER BY LENGTH(pattern)
 DESC; 
   
 This returns the desired route, but also other routes that may be first in
 the select, and ASTCC uses that instead fo the exact matching route. 
   
 How do I get ASTCC to select the routes starting at the begining of $number
 and not just anything that matches an expression in $number? 
  

Try with:

^01154.*

being 54 the country code for Argentina and 011 the internacional prefix.

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Re: [Asterisk-Users] ASTCC and Pattern question

[Nicolás Gudiño]

Hello,

  I just installed ASTCC and it was VERY easy to get running. I have a
 question about Pattern Via the web page I click the Routes link and
 everything makes sense to me but the pattern part. I tried _NXXNXX
 with the idea that everything would match this. Well it doesn't work...
 
 Does anyone have a good how-to?

I think you can use something like:

^ + code + .*

For example, if you want to add Argentina (54) as a route, and you
dial 011 for international:

^01154.*

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Re: [Asterisk-Users] Compiling zaptel 1.0.2 on Fedora Core

[Nicolás Gudiño]

Hello,

 I'm trying to get zaptel 1.0.2 compiled on FC2 or FC3 and I'm getting
 compile time errors.  Systems include:
[snip] 
 In file included from /usr/home/bwright/zaptel-1.0.2/zaptel.c:40:
 /usr/home/bwright/zaptel-1.0.2/zconfig.h:10:27: linux/version.h: No such
 file or directory
[snip]

Copied straight from the wiki (search for fedora):

cd /usr/src (or make sure you are in your source directory)
cp configs/config-for-my-kernel .config
make oldconfig
make include/asm
make include/linux/version.h
make SUBDIRS=scripts 

It worked for me last night. Regards,

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Re: [Asterisk-Users] unable to compile testcpuid.c in spandsp in x86_64

[Nicolás Gudiño]

Hello,

 I'm unable to compile testcpuid.c with the __x86_64__ architecture
 (Athlon 64 processor).  The messages are:
 
 /tmp/ccONleRV.s: Assembly messages:
 /tmp/ccONleRV.s: Error: suffix or operands invalid for 'pushf'
  'pop'
  'push'
  'popf'
 
 Is it safe to ignore this module?

I have similar problems compiling under PPC. I just removed that
module together with other troubling assembler parts (MMX detection
routines). After that it compiled fine and worked well (but the
platform is 32 bits, I dunno if it will work under a 64bit platform).
Regards,

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Re: [Asterisk-Users] SIP Phones-Receptionist Setup

[Nicolás Gudiño]

Hello,

 
 I'm certainly not an expert on this, but isn't one of the limiting
 factors the functionality implemented by manufacturers in their sip
 phones?  Or, are we assuming the lamp field is an external device
 unrelated to the current production phones?
 
 (I do understand that at least some sip phones have implemented the
 function.)

I agree, the limiting factor is moslty on the phones, not in asterisk.
The SNOM has the extension status working. I suppose that its possible
to code something for XML enabled screen phones, like the Polycom 600
or a Cisco, but I've never seen one myself. So I'm not sure if its
doable.

I wrote a FOP client to display ongoing call status on a LCD attached
to a linux computer via serial port (call duration, number dialed,
timer and rate of the call). If there is enough interest we can make a
relatively cheap external device for BLF, and possibly transfers,
night mode, et all.

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Re: [Asterisk-Users] Monitoring app. - see whats really going on in asterisk

[Nicolás Gudiño]

Hello,

 Is there a way to debug, more debug than already than the option - does
 ? 
   
 Like... 
 when Answer is executed, can I get more info from where this app is run,
 what data is processed.. ? 
 (is there a monitoring app. which I can use ?) 

You can connect to the manager port. Enable it in
/etc/asterisk/manager.conf then telnet to port 5038, log in, and see
the events fly by. It is not debug per se, but you will see *lots* of
useful information.

You can run op_server.pl from my Flash Operator Panel (
http://www.asternic.org ) with debug level set to 1 if you do not know
how to login.

Regards,
 
-- 
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] Execute a script upon registration

[Nicolás Gudiño]

Hello,

 Is it possible to execute a script upon successful registration and
 authentication of a SIP device in Asterisk? For instance, have a script log
 all successful registrations in a database or authenticate users instead of
 using the secret=password in the sip.conf file? Thanks -

You can write an application that listens to the manager port and
looks for registration events and launch scripts accordingly.

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Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] Busy Lamp Field

[Nicolás Gudiño]

Hola Jorge,

On Wed, 24 Nov 2004 09:51:38 -0500, Jorge Mendoza [EMAIL PROTECTED] wrote:
 Some days ago there was a subject regarding BLF (SIP Phone-receptionist
 Setup).
 We are the developers of a Price Verify Terminal for a French company.
 We have developed the hardware (small board based on a PPC 823e),
 working with Linux embedded (based on Wolfgang Denk's work).
 I think that it can be a good BLF.
 Probably it is possible to integrate the Nicolas's FOP or a new application.

If you want to build a prototype I would gladly help with the
software. I'm also thinking on building a cheap BLF just with leds
(not LCD) Regards,

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Re: [Asterisk-Users] SIP Phones-Receptionist Setup

[Nicolás Gudiño]

Hi,

 Me and another guy are working on LCD drivers etc for Linux.  The thing
 is, the display would be run from your Asterisk Server. I.E. It will
 need to be run from either Parallel, Serial or USB port.  We will open
 source it once finished, and are not too far off, probably just a spare
 day would do it...the problem is finding a spare day.  I guess the other
 option would be to use one of the small PC's to run Asterisk and a panel
 on the receptionists desk.

I'm the developer of FOP. I was busy the last weeks with a related
project: to build little LCD 2x16 lines boxes to display call status
for phone booths. We have a prototype working, it is connected to the
serial port and displays the name of the location you are dialing, and
the rate of the call. When the call starts, a timer and the current
cost of the call.

The software side is just a client for op_server.pl wirtten in PERL,
similar to the flash client, but for feeding the LCD display.

As I see there is a need for BLF indicators of some sort, I might
consider on making such a piece of hardware. A hardware panel with LCD
or just leds for lines, connected to the serial port of a linux pc (it
can be the asterisk machine or another one).

If there is enough interest, we can make it a reality... The question
is: what features do you need and how much are you willing to pay for
it.

Regards,

-- 
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] Problem with fax tone (CNG) and fax detection

[Nicolás Gudiño]

Hello,

 
 The problem with the call files is that the busy tone is not being
 detected, and the reason the busy tone is not detected is because the
 fax tone (CNG) is being injected onto the line by the TxFax application.
 
 When I remove |caller from the call files (no CNG tones), all fax
 calls are properly sent (busy is detected by * and the call retried).
 However, by removing |caller| and not sending the CNG tones, some fax
 receivers will not be able to properly handle the call.
 
 Is there some way for * to detect busy tones while ignoring (filtering)
 the CNG tones?

Maybe by setting the following in zapata.conf

;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

-- 
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Re: [Asterisk-Users] Log extension in CDR when forwarding calls to another number

[Nicolás Gudiño]

Hello,

 I've read something about CSV_LOGUSERFIELD in the CDR but I'm not sure how to 
 invoke the field and put in the terminating, or lookup, extension when the 
 call first comes in.  I believe this field has to be uncommented in the 
 cdr_csv.c file then asterisk recompiled to have this value field set in the 
 CDR?  What would the extension look like to record the value into the CDR?  
 exten = s,1,SetVar(loguserfield=${EXTEN})?

show application SetCDRUserField at the CLI

-- 
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Buenos Aires - Argentina
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Re: [Asterisk-Users] manager api: how to handle failed calls

[Nicolás Gudiño]

Hello,

Comments inline..

  The question is how to correctly handle failed calls. 
  In my application I want to make  hundreds of outgoing calls automatically.
  When the callee  pick up the phone he gets a playback message and give an
 acknowledge by means of dtmf code. 
  I make use of manager command originate, something like 
  Action:originate 
  channel: ZAP/g1/ 
  Variable:X|Y|Z 
  extension: test 
  the extension test is something like 
  [test] 
  exten  s,1 , wait ()
  exten  s, 2 , answer ()
  exten s, 3 playback(XX) 
  The problem is since I don't use the application dial  inside the extension
 I cannot get any value from 
  DIALSTATUS or HANGUPCAUSE variable 
  I tried several strategies: 
  1) 
  change the logic and use local pseudo channel 
  In the originate command if I use channel: local/[EMAIL PROTECTED]/n 
  where test1 is:
  [test1] 
  exten = _.,1,Dial(ZAP/g1/g${EXTEN}) 
  exten = _.,2,NoOp( 2 HANGUPCAUSE is ${HANGUPCAUSE}) 
  exten = _.,3,NoOp( 2 DIALSTATUS is ${DIALSTATUS}) 
  exten = _.,4,NoOp(  number is ${number}) 
  exten = _.,5,Hangup 
  
  I got the correct HANGUP value ( ie BUSY) but unfortunately  I cannot see
 the variables set on the originate command.
  I wonder  why not? 

Maybe, (just maybe, I did not try it myself)  the originate variables
are passed using asterisk CVS-HEAD and variable names prefixed with
underscore... Eg: Use variable _X instead of X in the originate
command. Let me know if it works.

Regards,

-- 
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] timeout

[Nicolás Gudiño]

Hello,

On Fri, 12 Nov 2004 14:40:10 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
 Good day all
 I have my extensions.conf configured so that it waits 8s the answers
 with a message saying press 1 for... and 2 for..
 How do I tell it then that if the did not press anything to should go to
 the operator.
 And/Or if they did not press something it will play the message again
 And/Or if they typed a wrong extension ti will read the menu again
 please let me know
 Thaks
 Altus
 

http://www.voip-info.org/wiki-Asterisk+standard+extensions

Look for the t and i extensions...

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Nicolás Gudiño
Buenos Aires - Argentina
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