[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3368 - 12 msgs

Thu Apr 8 08:11:57 MST 2004

This message is in response to Flash operator problem. My op_server.pl seems to be same. I also created the variable.txt to the /var/www/html/panel folder and when I run htt://192.168.0.0/panel it just says at the bottom transferring data. I don't see anything on the screen.
I also checked my manager.conf file. I was able to telnet into the manager interface and it's running fine.
So I am not sure why I don't see anything on the screen. What about he op_server.cfg file. Do I need to change that. Can the default one still work at least bring up the screen to tell me it is working fine.
Thanks
 -----Original Message-----
From: 	asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com]  On Behalf Of asterisk-users-request at lists.digium.com
Sent:	Thursday, April 08, 2004 11:13 AM
To:	asterisk-users at lists.digium.com
Subject:	Asterisk-Users digest, Vol 1 #3368 - 12 msgs

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Today's Topics:

   1. RE: Out of trunk data space on call number 16386, dropping (Justin Carlson)
   2. Caller ID on TDM400P Quad FXS (Daniel ANDRE)
   3. Re: Fritz ISDN PCI v2 and CAPI (Michael Welter)
   4. Web interface for Asterisk (Jain, Sonal)
   5. PC based Switchboard application (Keith D'Atrio)
   6. Restart Asterisk (Jain, Sonal)
   7. Re: Restart Asterisk (Thomas Gallaway)
   8. Re: Restart Asterisk (Steve Foy)
   9. Re: Restart Asterisk (WipeOut)
  10. Re: Web interface for Asterisk (Steve Foy)
  11. Re: Web interface for Asterisk (Altus Snyman)
  12. RE: dreaded Caller*ID failed checksum (Jeremy Hall)

--__--__--

Message: 1
From: "Justin Carlson" <justin at lach.net>
To: <asterisk-users at lists.digium.com>
Subject: RE: [Asterisk-Users] Out of trunk data space on call number 16386, dropping
Date: Thu, 8 Apr 2004 08:16:09 -0500
Reply-To: asterisk-users at lists.digium.com

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how did you guys go about diableing it.  Is it the threwaycalling directive
in zapata.conf ?
  -----Original Message-----
  From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Warren H. Prince
  Sent: Thursday, April 08, 2004 8:01 AM
  To: asterisk-users at lists.digium.com
  Subject: Re: [Asterisk-Users] Out of trunk data space on call number
16386, dropping

  I work with Tony, so I'm responding for him.  Yes, it appears only during
a conference call.  So, if we disable conferencing, we do not receive the
error.

  Justin Carlson wrote:
if you disable conferencing does the problem go away?

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Tony Buser
Sent: Wednesday, April 07, 2004 2:30 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Out of trunk data space on call number
16386, dropping

I'm having the same kind of issues.  We get the out of trunk data space
error consistently during conference calls between asterisk servers.
And occasionally on regular iax calls.  Also while we're on a conference
call it seems to cause other calls going out through iax to fail and
also give this error.  (weather its to another asterisk server or
through say oneunified)

If you figure this out, please let us know here.  I'm pretty much at a
loss as to what could be causing it.

Justin Carlson wrote:

	Hi all,

We keep getting these and all the calls between these two asterisk boxes

get

dropped.  what is going on here, I have been trying to solve this problem

on

my own but maybe I don't have the trunk setup right.  also I have posed

the

output of my full log of the machine with the zap interface, the other is
using ztdummy.

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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE></TITLE>

<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR></HEAD>
<BODY>
<DIV>how=20
did you guys go about diableing it.&nbsp; Is it the threwaycalling =
directive in=20
zapata.conf ?</DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
 <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft>-----Original Message----- From:=20
 asterisk-users-admin at lists.digium.com=20
 [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of =
Warren H.=20
 Prince Sent: Thursday, April 08, 2004 8:01 AM To:=20
 asterisk-users at lists.digium.com Subject: Re: =
[Asterisk-Users] Out of=20
 trunk data space on call number 16386, dropping </DIV>I =
work=20
 with Tony, so I'm responding for him.&nbsp; Yes, it appears only =
during a=20
 conference call.&nbsp; So, if we disable conferencing, we do not =
receive the=20
 error. Justin Carlson wrote:=20
 <BLOCKQUOTE cite=3Dmid00ff01c41cd9$3789b870$9b01010a at sphinx =
type=3D"cite"><PRE wrap=3D"">if you disable conferencing does the =
problem go away?

-----Original Message-----
From: <A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:asterisk-users-admin at lists.digium.com">asterisk-users-admi=
n at lists.digium.com</A>
[<A class=3Dmoz-txt-link-freetext =
href=3D"mailto:asterisk-users-admin at lists.digium.com">mailto:asterisk-use=
rs-admin at lists.digium.com</A>]On Behalf Of Tony Buser
Sent: Wednesday, April 07, 2004 2:30 PM
To: <A class=3Dmoz-txt-link-abbreviated =
href=3D"mailto:asterisk-users at lists.digium.com">asterisk-users at lists.digi=
um.com</A>
Subject: Re: [Asterisk-Users] Out of trunk data space on call number
16386, dropping

I'm having the same kind of issues.  We get the out of trunk data space
error consistently during conference calls between asterisk servers.
And occasionally on regular iax calls.  Also while we're on a conference
call it seems to cause other calls going out through iax to fail and
also give this error.  (weather its to another asterisk server or
through say oneunified)

If you figure this out, please let us know here.  I'm pretty much at a
loss as to what could be causing it.

Justin Carlson wrote:

  </PRE>
    <BLOCKQUOTE type=3D"cite"><PRE wrap=3D"">	Hi all,

We keep getting these and all the calls between these two asterisk boxes
    </PRE></BLOCKQUOTE><PRE wrap=3D""><!---->get
  </PRE>
    <BLOCKQUOTE type=3D"cite"><PRE wrap=3D"">dropped.  what is going on =
here, I have been trying to solve this problem
    </PRE></BLOCKQUOTE><PRE wrap=3D""><!---->on
  </PRE>
    <BLOCKQUOTE type=3D"cite"><PRE wrap=3D"">my own but maybe I don't =
have the trunk setup right.  also I have posed
    </PRE></BLOCKQUOTE><PRE wrap=3D""><!---->the
  </PRE>
    <BLOCKQUOTE type=3D"cite"><PRE wrap=3D"">output of my full log of =
the machine with the zap interface, the other is
using ztdummy.
    </PRE></BLOCKQUOTE><PRE wrap=3D""><!---->

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  </PRE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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--__--__--

Message: 2
Date: Thu, 08 Apr 2004 15:31:20 +0200
From: Daniel ANDRE <dandre at iris-tech.fr>
Organization: IRIS Technologies
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Caller ID on TDM400P Quad FXS
Reply-To: asterisk-users at lists.digium.com

Hello,

I have a quad FXS TDM400P and it works fine with my asterisk 
configuration. I wonder to know if there is any configuration option so 
that Caler ID information should be properly sent by the TDM400 to the 
phone connected to it.

Best Regards,

Daniel

-- 
Daniel ANDRE (mailto:daniel.andre at iris-tech.fr)
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com

--__--__--

Message: 3
Date: Thu, 08 Apr 2004 07:35:09 -0600
From: Michael Welter <mike at introspect.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI
Reply-To: asterisk-users at lists.digium.com

Can CAPI and the ASUSCOM ISDNLink card be used in the US?  What goes on 
the /etc/capi.conf file instead of "fcpci"?

Brian Cuthie wrote:
> Can this Frtiz card be used in the US?
> 
> -brian 
> 

-- 
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
mike at introspect.com
www.introspect.com

--__--__--

Message: 4
Date: Thu, 8 Apr 2004 09:45:47 -0400
From: "Jain, Sonal" <Sonal.Jain at Sterlingbancorp.com>
To: <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] Web interface for Asterisk
Reply-To: asterisk-users at lists.digium.com

I installed the flash operator panel and I also installed the =
flash-shockwave in my mozilla browser. I followed the read me =
instructions in the Flash operator and made the changes to the =
op_server.pl but when I run the browser I get transferring data and just =
sits there. I don't see anything being transferred. If any body has used =
this software please tell me what am I doing wrong.
 I copied the two files from the html directory to /var/www/html/panel =
directory which is the web root.
I also changed the manager.conf file and created a user ID and secret =
which I specified in the op_server.pl.

Thanks,

--__--__--

Message: 5
Date: Thu, 8 Apr 2004 09:43:27 -0400
From: "Keith D'Atrio" <keith at manetheren.no-ip.com>
To: <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] PC based Switchboard application
Reply-To: asterisk-users at lists.digium.com

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Hello All
    I am looking for a PC based switchboard application. Cisco =
CallManager has a web attendant console that allows you to use the PC to =
transfer calls and the like and I was wondering if there was a similar =
program compatible with *.
Thank you in advance
Keith D'Atrio

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML =
DIR=3Dltr><HEAD><META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1"></HEAD><BODY><DIV><FONT face=3D'Arial' =
color=3D#000000 size=3D2>Hello All</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; I am looking for a =
PC based =0A=
switchboard application. Cisco CallManager has a web attendant console =
that =0A=
allows you to use the PC to transfer calls and the like and I was =
wondering if =0A=
there was a similar program compatible with *.</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Thank you in advance</FONT></DIV>=0A=
<DIV><FONT face=3DArial size=3D2>Keith D'Atrio</FONT></DIV></BODY></HTML>
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--__--__--

Message: 6
Date: Thu, 8 Apr 2004 09:48:57 -0400
From: "Jain, Sonal" <Sonal.Jain at Sterlingbancorp.com>
To: <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users at lists.digium.com

Is it true that every time we make a change in the configuration file we =
need to restart the asterisk server. This will not be practical in the =
production environment.=20
Thanks,

--__--__--

Message: 7
Date: Thu, 08 Apr 2004 09:58:51 -0400
From: Thomas Gallaway <rescue at port11.net>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users at lists.digium.com

Jain, Sonal wrote:

>Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. 
>Thanks,
>  
>
Entering reload in the console should do if you edit the extensions.conf 
and some other files. There are some files if you edit them you need to 
shut down and restart asterisk.

--__--__--

Message: 8
Date: Thu, 8 Apr 2004 14:59:44 +0100
From: Steve Foy <steve at unite.net>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users at lists.digium.com

You can reload the config files with the 'reload' command in the CLI.

On Thu, Apr 08, 2004 at 09:48:57AM -0400, Jain, Sonal wrote:
> Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. 
> Thanks,
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Steve Foy        |  http://www.unite.net
UNITE Solutions  |  Tel: 028 9077 7338 

--__--__--

Message: 9
Date: Thu, 08 Apr 2004 15:00:23 +0100
From: WipeOut <wipe_out at users.sourceforge.net>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users at lists.digium.com

Jain, Sonal wrote:

>Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. 
>Thanks,
>  
>
No, you don't have to "restart", you have to "reload"..

 From the CLI just type "reload" and hit enter..
or for a command line run "asterisk -rx reload"..

Later..

--__--__--

Message: 10
Date: Thu, 8 Apr 2004 15:01:02 +0100
From: Steve Foy <steve at unite.net>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Web interface for Asterisk
Reply-To: asterisk-users at lists.digium.com

Hi again :)

Can you give me a URL for the software you mentioned?

Cheers,
Steve

On Thu, Apr 08, 2004 at 09:45:47AM -0400, Jain, Sonal wrote:
> I installed the flash operator panel and I also installed the flash-shockwave in my mozilla browser. I followed the read me instructions in the Flash operator and made the changes to the op_server.pl but when I run the browser I get transferring data and just sits there. I don't see anything being transferred. If any body has used this software please tell me what am I doing wrong.
>  I copied the two files from the html directory to /var/www/html/panel directory which is the web root.
> I also changed the manager.conf file and created a user ID and secret which I specified in the op_server.pl.
> 
> Thanks,
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Steve Foy        |  http://www.unite.net
UNITE Solutions  |  Tel: 028 9077 7338 

--__--__--

Message: 11
Subject: Re: [Asterisk-Users] Web interface for Asterisk
From: Altus Snyman <altus at stormcorp.co.za>
To: asterisk <asterisk-users at lists.digium.com>
Date: Thu, 08 Apr 2004 16:10:16 +0200
Reply-To: asterisk-users at lists.digium.com

ok this is what I did
I moved all to my /var/www/html/control. did the changes is my files and
used the copy of manager.conf. I started asterisk and did
/var/www/html/control/op_server.pl and pointed my browser to
192.168.0.1/control/html ... had the  same problem. Then I went and set
debug to 1 in op_server(did not help cant read the lang?). So I made the
dir /var/www/html/wtf and moved the 2 files in the html dir to
here,restarted asterisk and /var/www/html/control/op_server.p and
pointed my browser to 192.168.0.1/wtf and wtf it worked,now Im not
talking about the transfer and hangup??
here is my conf
##########################################
# CONFIGURATION
#
# parameters to connect to Asterisk Manager
my $manager_host   = "192.168.0.1";
my $manager_user   = "altus";
my $manager_secret = "altus";

#
# parameters for the op_server
my $web_hostname  = "192.168.0.1";   # must be the same address you use
to contact the web server
my $listen_port   = 4445;
my $security_code = 'd39i393kd';           # secret code for performing
hangups and transfers

#
# location of variables.txt needed by the flash applet
# (must be the same directory as the web page and swf file)
my $flash_dir = "/var/www/html/wtf/";

#
# Debug level to stdot
my $debug = 1;

On Thu, 2004-04-08 at 15:45, Jain, Sonal wrote:
> I installed the flash operator panel and I also installed the flash-shockwave in my mozilla browser. I followed the read me instructions in the Flash operator and made the changes to the op_server.pl but when I run the browser I get transferring data and just sits there. I don't see anything being transferred. If any body has used this software please tell me what am I doing wrong.
>  I copied the two files from the html directory to /var/www/html/panel directory which is the web root.
> I also changed the manager.conf file and created a user ID and secret which I specified in the .
> 
> Thanks,
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 

--__--__--

Message: 12
Subject: RE: [Asterisk-Users] dreaded Caller*ID failed checksum
Date: Thu, 8 Apr 2004 08:03:22 -0600
From: "Jeremy Hall" <jeremyhall at mpccorp.com>
To: <asterisk-users at lists.digium.com>
Reply-To: asterisk-users at lists.digium.com

Jeff,

I see the same thing on my FXO card, but it is an Intel modem, not a
true Digium X100P.  I suspected it was my card, but if you are seeing it
on a true card, maybe there is hope for mine yet.  I haven't had time to
troubleshoot yet as I have been having too much fun playing with other
features.

Let us know if you find the solution, and I will do the same if I get
mine working.  I am hoping to be able to do some work on it this weekend
to try and see what is going on.  In my case I have several other phones
plugged into the line as I don't have any FXS ports yet, so eliminating
them was going to be one of my first steps.  The jack that my * server
is attached to is CAT5 run directly from the telco access box.

Aside from being a software decoding error or a telco sending error, my
first suspects are line noise on the cabling from other devices or
devices near the phone cabling.  Electrical noise introduced into the
signal inside the asterisk system is another failure point I want to try
to eliminate.

As a last resort, I was thinking of throwing that modem into my Windows
PC and loading the drivers and software for it and see if CallerID works
in that mode.  I don't know if Windows would be able to load modem
drivers for the Digium card or not, but that is another idea for you to
try.  These cards are basically glorified sound cards that attach to a
telephone line, so if the Windows software can correctly read the
signal, that would maybe point it in the software or driver area.  If
that turns out to be the case, I may be forced to go ahead and get an
actual Digium card sooner than I anticipated in order to prove the
theory.

Regards,

Jeremy

-----Original Message-----
From: Jeff Gustafson [mailto:ncjeffgus at zimage.com]=20
Sent: Thursday, April 08, 2004 12:06 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] dreaded Caller*ID failed checksum

	Caller*ID used to work as some point, but I can't seem to get it
going
these days.  The card is a x101p.  I've tried going up and down the
rxgain scale.  Can the txgain effect it at all?  When I plug in a phone
into the line with a splitter it can decode caller id with no problems.
        Reading through the mailing list archives hasn't given me any
move clues.  Any ideas?

                                ...Jeff

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--__--__--

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