Jain, Sonal
2004-Apr-08 08:11 UTC
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3368 - 12 msgs
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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
asterisk-users-request@lists.digium.com
Sent: Thursday, April 08, 2004 11:13 AM
To: asterisk-users@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@lach.net>
To: <asterisk-users@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@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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Warren H. Prince
Sent: Thursday, April 08, 2004 8:01 AM
To: asterisk-users@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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Tony Buser
Sent: Wednesday, April 07, 2004 2:30 PM
To: asterisk-users@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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<DIV><SPAN class=3D334301513-08042004><FONT face=3DArial
color=3D#0000ff size=3D2>how=20
did you guys go about diableing it. Is it the threwaycalling directive
in=20
zapata.conf ?</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
<DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT
face=3DTahoma=20
size=3D2>-----Original Message-----<BR><B>From:</B>=20
asterisk-users-admin@lists.digium.com=20
[mailto:asterisk-users-admin@lists.digium.com]<B>On Behalf Of
</B>Warren H.=20
Prince<BR><B>Sent:</B> Thursday, April 08, 2004 8:01
AM<BR><B>To:</B>=20
asterisk-users@lists.digium.com<BR><B>Subject:</B> Re:
[Asterisk-Users] Out of=20
trunk data space on call number 16386,
dropping<BR><BR></FONT></DIV>I work=20
with Tony, so I'm responding for him. Yes, it appears only
during a=20
conference call. So, if we disable conferencing, we do not receive
the=20
error.<BR><BR>Justin Carlson wrote:=20
<BLOCKQUOTE cite=3Dmid00ff01c41cd9$3789b870$9b01010a@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@lists.digium.com">asterisk-users-admin@lists.digium.com</A>
[<A class=3Dmoz-txt-link-freetext
href=3D"mailto:asterisk-users-admin@lists.digium.com">mailto:asterisk-users-admin@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@lists.digium.com">asterisk-users@lists.digium.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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--__--__--
Message: 2
Date: Thu, 08 Apr 2004 15:31:20 +0200
From: Daniel ANDRE <dandre@iris-tech.fr>
Organization: IRIS Technologies
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID on TDM400P Quad FXS
Reply-To: asterisk-users@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@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@introspect.com>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI
Reply-To: asterisk-users@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@introspect.com
www.introspect.com
--__--__--
Message: 4
Date: Thu, 8 Apr 2004 09:45:47 -0400
From: "Jain, Sonal" <Sonal.Jain@Sterlingbancorp.com>
To: <asterisk-users@lists.digium.com>
Subject: [Asterisk-Users] Web interface for Asterisk
Reply-To: asterisk-users@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@manetheren.no-ip.com>
To: <asterisk-users@lists.digium.com>
Subject: [Asterisk-Users] PC based Switchboard application
Reply-To: asterisk-users@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> I am looking for a PC based
=0Aswitchboard application. Cisco CallManager has a web attendant console that
=0Aallows you to use the PC to transfer calls and the like and I was wondering
if =0Athere 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@Sterlingbancorp.com>
To: <asterisk-users@lists.digium.com>
Subject: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users@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@port11.net>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users@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@unite.net>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users@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@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@users.sourceforge.net>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users@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@unite.net>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Web interface for Asterisk
Reply-To: asterisk-users@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@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@stormcorp.co.za>
To: asterisk <asterisk-users@lists.digium.com>
Date: Thu, 08 Apr 2004 16:10:16 +0200
Reply-To: asterisk-users@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@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@mpccorp.com>
To: <asterisk-users@lists.digium.com>
Reply-To: asterisk-users@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@zimage.com]=20
Sent: Thursday, April 08, 2004 12:06 AM
To: asterisk-users@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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