Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

[michael koehler]

It is a good resource for neck tie non-geeks in small offices and will 
hopefully evangelize
many of the "uhh, it's open source and it is for free = so this could 
not be good" heathens.

Michael
On Jul 8, 2004, at 11:19 PM, usedcanon wrote:

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Harold
Workman
Sent: 08 July 2004 20:15
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
what does that have to do with an overpriced book?
and i agree with Joe.  With this book sourcing most of the 
documentation
directly from wiki, why pay for something thats free?  Id rather 
donate $49
to keeping wiki free to the enviroment.
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Re: [Asterisk-Users] asterisk to asterisk config

[Brian K. West]

Well do you yahoo?  har har har j/k

bkw

- Original Message - 
From: "Gonzalo Servat" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 09, 2004 12:13 AM
Subject: RE: [Asterisk-Users] asterisk to asterisk config


> On 9/07/2004 6:06 AM +0100, Kevin Walsh wrote:
> 
> > Eugen Cristea [EMAIL PROTECTED] wrote:
> >> Find local movie times and trailers on Yahoo! Movies.
> >> http://au.movies.yahoo.com
> >>
> > What does Yahoo have to do with it?
> >
> > Have you considered trimming your quotes?  Clearly not.
> 
> Have you considered maybe his webmail provider (Yahoo) is automatically 
> inserting the advertisement footer? Clearly not ;)
> 
> Gonzalo
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RE: [Asterisk-Users] Mandrake 10, Request for comments.

[[EMAIL PROTECTED]]

Thanks for the reply Reid, much appreciated.
At 22:24 7/8/2004, you wrote:
I've been running * (various HEAD versions) on Mandrake 10.0 for over a month
with no major issues. I've had trouble with my zapata card, but I believe
that was due to a flaky motherboard.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Wednesday, July 07, 2004 7:32 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Mandrake 10, Request for comments.
>
> My * is presently running fine on Mandrake 9.2, but Ive been
> entertaining
> moving to Mandrake 10.0 to enjoy the obvious improvement in
> kernal speed Im
> seeing on other 10.0 boxes Ive recently built for other
> applications. (10.0
> is the first implementation of the 2.6 kernal)
>
> Any comments from anyone who's running on 10.0?
> IS anyone running * on Mandrake 10.0?
> If so, any issues stand out?
>
> I'm hesitant because of the dot zero release of anything is
> always broken,
> and so far this has not been an exception, but not insurmountable.
>
> Thanks in advance.
> Marc
>
>
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RE: [Asterisk-Users] asterisk to asterisk config

[Gonzalo Servat]

On 9/07/2004 6:06 AM +0100, Kevin Walsh wrote:
Eugen Cristea [EMAIL PROTECTED] wrote:
Find local movie times and trailers on Yahoo! Movies.
http://au.movies.yahoo.com
What does Yahoo have to do with it?
Have you considered trimming your quotes?  Clearly not.
Have you considered maybe his webmail provider (Yahoo) is automatically 
inserting the advertisement footer? Clearly not ;)

Gonzalo
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RE: [Asterisk-Users] asterisk to asterisk config

[Kevin Walsh]

Eugen Cristea [EMAIL PROTECTED] wrote:
> > prepare to get flamed for not reading or at least
> > googling:
> > 
> > go here:
> > http://www.voip-info.org/wiki-IAX
> > 
> > ___
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> > 
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > 
> > 
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> a better one:
> http://www.voip-info.org/wiki-Asterisk+-+dual+servers
> 
> Find local movie times and trailers on Yahoo! Movies.
> http://au.movies.yahoo.com
>
What does Yahoo have to do with it?

Have you considered trimming your quotes?  Clearly not.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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RE: [Asterisk-Users] ok

[Kevin Walsh]

Shanmuganathan Kumaravel [EMAIL PROTECTED] wrote:
>
> Post me in the mailing list.
>
No.

-- 
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_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Kevin Walsh]

Dameon D. Welch-Abernathy [EMAIL PROTECTED] wrote:
> On Thu, 2004-07-08 at 16:44, Sathya wrote:
> > > $50 is a bit steep for a book
> > 
> > Usually author of a book makes 10 to 15 percent of the cover price. So
> > whoever who wrote this book will get 5 bucks a book.
> 
> In my experience, the royalty is 10 to 15% of the *wholesale* price,
> which means it's more like two to three bucks a book. Trust me, unless
> you're Stephen King, John Grisham, or someone like that, you don't make
> all that much writing books (at least directly).
> 
All the more reason for the author to consider OpenDoc publishing.

Authors of technical books tend to make more money out of consulting
anyway, as they're seen as an expert in their field.

-- 
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 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
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Re: [Asterisk-Users] asterisk to asterisk config

[Eugen Cristea]

--- Nik Martin <[EMAIL PROTECTED]> wrote: > 
> 
> Eugen Cristea wrote:
> > Hi,
> > 
> > I would like to set two separate asterisks to talk
> to
> > each other.
> > Any suggestions?
> > I'm a "baby" asterisk fan, only started to play
> two
> > weeks ago, first managed to use kphone with
> asterisk
> > and a  X100P card that is up and running as well.
> > 
> > Thanks,
> > Eugen
> > 
> 
> prepare to get flamed for not reading or at least
> googling:
> 
> go here:
> http://www.voip-info.org/wiki-IAX
> 
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>  
a better one:
http://www.voip-info.org/wiki-Asterisk+-+dual+servers

Find local movie times and trailers on Yahoo! Movies.
http://au.movies.yahoo.com
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[Asterisk-Users] ok

[Shanmuganathan Kumaravel]

  
Post me in the mailing list.

Re: [Asterisk-Users] Dead Budgetone-101?

[Gary Mart]

I experienced the same symptoms.  One of my Budgetones hung in state
in which the tftp IP number could not be changed and the tftp download
site was not working.  I put the Budgetone behind a double natting
(maybe there is another name for this but that is what it does)
gateway and redirected the outgoing IP to a tftp server that worked.
The Budgetone resumed functioning after a successful image download.

Gary


On Sun, Jun 27, 2004 at 05:46:05PM +0100, Max Lock wrote:
> 
>  Hi Folks,
> 
>  Since there isn't a grandstream forum AFAIK I guess someone here may be able to 
> shed some light on this. Apologies if this is viewed as offtopic..
> 
>  I think I may have killed the firmware on my Grandstream Budgetone 101. I found a 
> source for the 1.0.5.30 firmware and made the files available over tftp. The phone 
> downloaded the files but now doesn't boot and hangs with a blank screen, although 
> the keypad lights blink occasionally.
> 
>  After that I tried to go back to the 1.0.4.55 firmware I was running previously, 
> but the tftp transfer no longer seems to work, the phone sends a tftp request, but 
> doesn't seem to get the files?!
> 
>  Has anyone else seen this? is there a way to access the phone via the internal PCB 
> edge connector?
> 
>  -Cheers Max
> 
> --
> Max Lock
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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Dameon D. Welch-Abernathy]

On Thu, 2004-07-08 at 16:44, Sathya wrote:
> > $50 is a bit steep for a book
> 
> Usually author of a book makes 10 to 15 percent of the cover price. So
> whoever who wrote this book will get 5 bucks a book.

In my experience, the royalty is 10 to 15% of the *wholesale* price,
which means it's more like two to three bucks a book. Trust me, unless
you're Stephen King, John Grisham, or someone like that, you don't make
all that much writing books (at least directly).

-- PhoneBoy

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Re: [Asterisk-Users] wake-up call script in wiki

[Gonzalo Servat]

On 9/07/2004 10:21 AM +0700, Isianto Istiadi wrote:
Dear guys,
I'm searching the wake-up call script in wiki, found one, but I have no
idea how to use it. Can you give some direction how to install it?
Thanks
I presume you're talking about this wake up call script: 


Stick the following in cron:
* * * * * root /path/to/run_wakeups.sh
/path/to/run_wakeups.sh contains:
= cut ==
#!/bin/bash
PENDING=/tmp/wakeups
OUTGOING=/var/spool/asterisk/outgoing
SLEEP=5
TIME=$(/bin/date +%H%M)
for fn in $PENDING/$TIME.*.call
do
if test -r $fn
then
 /bin/mv -f $fn $OUTGOING/
 sleep $SLEEP
fi
done
= cut ==
The following is my wakeup.agi. Changes to the original version are: some 
debugging functionality (as I was troubleshooting an issue where it would 
read out the wrong time when the script tells you what time the wake up 
call was set to), and it also creates the /tmp/wakeups directory if it 
doesn't already exist. I suggest using the one on the voip-info.org page 
first, and if you decide to use my version then use at your own risk :)

= cut ==
#!/usr/bin/perl
use Asterisk::AGI;
use Date::Manip;
use strict;
#
# Settings:
my $pending_dir = '/tmp/wakeups';
unless (-d '/tmp/wakeups') {
   mkdir('/tmp/wakeups');
}
my $local_context = 'default';
# values for the call file:
my $maxretries = 60;
my $retrytime = 30;
my $waittime =  35;
my $debug = 1;
#my $application = 'MusicOnHold';
my $application = 'Playback';
my $data = 'wake-up';
my $callerid = 'Wakeup Call Service <297>';
#
my ($sec,$min,$hour,$mday,$mon,$year,$wday,$yday,$isdst) = localtime(time);
if ($debug) {
   my $log = '/tmp/wakeup.log';
   unlink($log);
   open (DBG,">>$log") or die "Cannot open debug file: $!";
   print DBG "\n" . "-" x 50 . "\n";
   print DBG "Logging started: " . join('/', $mday, $mon, $year) . " " 
. join(':', $hour, $min, $sec) . "\n";
   print DBG "-" x 50 . "\n";
}

my $agi = new Asterisk::AGI;
my %stuff = $agi->ReadParse;# MUST DO THIS! -- (add this to 
constructor!)

# this says "1 to create, 2 to confirm, 3 to cancel"
my $func = $agi->get_data('wakeup-menu', 2, 1);
exit if $func == -1;
my ($caller) = $stuff{callerid} =~ /<(\d+)>/;
if ($func == 1)
{
my $time = $agi->get_data('time', 15000, 4);
exit if $func == -1;
if ($time =~ /^(\d{2})(\d{2})$/)
{
 my $hour = $1 * 1;
 my $min = $2;
 print DBG 'HOUR entered: ' . $hour . "\n" if $debug;
 print DBG 'MINUTE entered: ' . $min . "\n" if $debug;
 if ($hour > 0 && $hour <= 12 && $min < 60)
 {
  my $time;
#   $agi->stream_file('pls-enter');
#   $agi->stream_file('digits/1');
#   $agi->stream_file('for');
#   $agi->stream_file('digits/a-m');
#   $agi->stream_file('or');
#   $agi->stream_file('digits/2');
#   $agi->stream_file('for');
#   my $ampm = $agi->get_data('digits/p-m', 15000, 1);
  my $ampm = $agi->get_data('am-or-pm', 15000, 1);
  exit if $ampm == -1;
  if ($ampm == 1)
  {
   $time = ParseDate(sprintf("%s:%02s AM", $hour, $min));
   print DBG 'TYPE entered: AM' . "\n" if $debug;
   print DBG '$time is set to: ' . $time . "\n" if $debug;
  }
  elsif ($ampm == 2)
  {
   $time = ParseDate(sprintf("%s:%02s PM", $hour, $min));
   print DBG 'TYPE entered: PM' . "\n" if $debug;
   print DBG '$time is set to: ' . $time . "\n" if $debug;
  }
  else
  {
   $agi->stream_file('vm-sorry');
  }
  if ($time)
  {
   my $h = UnixDate($time, "%I") * 1;
   my $m = UnixDate($time, "%M");
   my $a = UnixDate($time, "%p");
   foreach my $fn (<$pending_dir/*.$caller.call>)
   {
unlink $fn;
   }
   my $filename = sprintf("%s/%04s.%s.call", $pending_dir, UnixDate($time, 
"%H%M"), $caller);

   open(FILE, ">$filename");
   printf FILE q{#
Channel: Local/[EMAIL PROTECTED]
MaxRetries: %s
RetryTime: %s
WaitTime: %s
Application: %s
Data: %s
Callerid: %s
},
$caller, $local_context,
$maxretries,
$retrytime,
$waittime,
$application,
$data,
$callerid,
;
   close(FILE);
   # say "Your wakeup call"
   $agi->stream_file('has-been-set-to');
   print DBG 'UnixDate $time translates to ' . UnixDate($time, "%o") . 
"\n" if $debug;
   print DBG 'localtime (UnixDate $time) translates to ' . 
localtime(UnixDate($time, "%o")) . "\n" if $debug;
   $agi->exec('SayUnixTime', sprintf("%s||IMp", UnixDate($time, "%o")));

   $agi->stream_file('for');
   $agi->stream_file('extension');
   $agi->say_digits($caller);
   $agi->stream_file('auth-thankyou');
  }
 }
 else
 {
  $agi->stream_file('vm-sorry');
 }
}
else
{
 $agi->stream_file('vm-sorry');
}
}
elsif ($func == 2)
{
my ($fn) = <$pending_dir/*.$caller.call>;
if ($fn)
{
 my ($time) = $fn =~ /\/(\d{4})\.\d+\.call/;
 $time =~ s/(\d\d)(\d\d)/$1:$2/;
 $agi->stream_file('is-set-to');
 $agi->exec('SayUnixTime', sprintf("%s||IMp", UnixDate(ParseDate($time),
 "%s")));
}
else
{
 $agi->stream_file('is-not-set');
}
$agi->stream_file('auth-thankyou');
}
elsif ($func == 3)
{
foreach my $fn (<$pending_dir/*.

[Asterisk-Users] Two outbound calls at once

[David Goldfein]

Hello,

I am having an issue with making two simultaneous outbound calls.

When I dial, both phones try to take the same channel and it causes an
error.  Anyone have any suggestions.  My set up is as follows:

CO – PRI – ASTERISK – VODAVI(pbx).

 

Thanks,

Dave

 

*CLI>

    -- Starting simple switch on 'Zap/69-1'

    -- Executing Wait("Zap/69-1",
".1") in new stack

    -- Executing DISA("Zap/69-1", "no-password|local")
in new stack

    -- Starting simple switch on 'Zap/68-1'

    -- Executing Wait("Zap/68-1",
".1") in new stack

    -- Executing DISA("Zap/68-1", "no-password|local")
in new stack

Jul  8 20:44:20 WARNING[-1394406480]: cdr.c:286 ast_cdr_init:
CDR already initialized on 'Zap/69-1'

    -- Executing Dial("Zap/69-1", "Zap/g2/6022831234")
in new stack

    -- Called g2/6022831234

Jul  8 20:44:20 WARNING[-1416709200]: cdr.c:286 ast_cdr_init:
CDR already initialized on 'Zap/68-1'

    -- Executing Dial("Zap/68-1",
"Zap/g2/6022831234") in new stack

    -- Called g2/6022831234

    -- Channel 0/2, span 2 got hangup

    -- Forcing restart
of channel 0/2 on span 2 since channel reported in use

    -- Hungup 'Zap/26-1'

  == No one is available to answer at this time

    -- Executing Congestion("Zap/68-1",
"") in new stack

    -- Channel 0/1, span 2 got hangup

    -- B-channel 0/2 successfully restarted on span 2

    -- Hungup 'Zap/25-1'

  == No one is available to answer at this time

    -- Executing Congestion("Zap/69-1",
"") in new stack

  == Spawn extension (local, 2831234, 2) exited non-zero on
'Zap/69-1'

    -- Hungup 'Zap/69-1'

  == Spawn extension (local, 2831234, 2) exited non-zero on
'Zap/68-1'

    -- Hungup 'Zap/68-1'

[Asterisk-Users] wake-up call script in wiki

[Isianto Istiadi]

Dear guys,
I'm searching the wake-up call script in wiki, found one, but I have no idea how to 
use it.
Can you give some direction how to install it?
Thanks

Isianto Istiadi
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Re: [Asterisk-Users] Intermittent SIP 404 Not Found response?

[Andrew Yager]

On 09/07/2004, at 8:00 AM, Bruce Komito wrote:
I have several SIP devices (Sipuras) that are working fine with *, 
except
for one annoying little problem.  Occassionally, after being registered
for some period of time, the Sipura returns a 404 Not Found to (I 
assume)
an INVITE request.  Of course, this makes the extension appear busy.
When this happens, I check the Sipura and it is thinks it is still
registered and I check * and it shows registered.  If I reboot the 
Sipura
or restart *, the problem clears.  It also clears by itself eventually.

Has anyone seen this behaviour and/or know how to cure it?
I believe I'm experiencing the same problem with Grandstream phones, 
although I haven't had time to track it down yet.

Andrew
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RE: [Asterisk-Users] VoicePulse Connect DID Problems

[Reid A. Forrest]

Yes! I have this same problem. This along with the fact that they go down
REGULARLY make me not want to use their service for anything other than
testing. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Andrew Joakimsen
> Sent: Wednesday, July 07, 2004 3:57 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] VoicePulse Connect DID Problems
> 
> 
> 
> I have a DID with VoicePulse Connect, but the sound quality 
> is horrible, it is
> often choppy and the caller's voice cuts out for 2-3 seconds 
> at least once a
> minute, I have contacted VoicePulse many times, and they do 
> not do anything
> about it! Does anyone have any similar problems? It isnt my 
> Asterisk config
> because I have 0 problems using NuFone.
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> 
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RE: [Asterisk-Users] Mandrake 10, Request for comments.

[Reid A. Forrest]

I've been running * (various HEAD versions) on Mandrake 10.0 for over a month
with no major issues. I've had trouble with my zapata card, but I believe
that was due to a flaky motherboard. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Wednesday, July 07, 2004 7:32 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Mandrake 10, Request for comments.
> 
> My * is presently running fine on Mandrake 9.2, but Ive been 
> entertaining 
> moving to Mandrake 10.0 to enjoy the obvious improvement in 
> kernal speed Im 
> seeing on other 10.0 boxes Ive recently built for other 
> applications. (10.0 
> is the first implementation of the 2.6 kernal)
> 
> Any comments from anyone who's running on 10.0?
> IS anyone running * on Mandrake 10.0?
> If so, any issues stand out?
> 
> I'm hesitant because of the dot zero release of anything is 
> always broken, 
> and so far this has not been an exception, but not insurmountable.
> 
> Thanks in advance.
> Marc
> 
> 
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Re: [Asterisk-Users] Asterisk receives TMC Labs Internet Telephony Innovation Award

[Kevin P. Fleming]

James H. Thompson wrote:
Asterisk receives TMC Labs Internet Telephony Innovation Award
http://www.tmcnet.com/it/0704/tmclabs.htm
Well, according to this article, Asterisk _requires_ the use of Digium's 
"proprietary" hardware (including that which is available in clone form 
from other vendors), so all of you out there (including me) who are 
using Asterisk with no PSTN hardware, you must be dreaming or on the 
crack pipe, because it can't possibly be working :-)
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Re: [Asterisk-Users] asterisk to asterisk config

[Nik Martin]

Eugen Cristea wrote:
Hi,
I would like to set two separate asterisks to talk to
each other.
Any suggestions?
I'm a "baby" asterisk fan, only started to play two
weeks ago, first managed to use kphone with asterisk
and a  X100P card that is up and running as well.
Thanks,
Eugen
prepare to get flamed for not reading or at least googling:
go here:
http://www.voip-info.org/wiki-IAX
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Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Alberto Fernandez]

REDHAT linux is charging for redhat certifications :P HAHAHA.


NOW we need digium to start training and certify in asterisk. (* itselft
is more complicated than linux as an OS, so it makes more sence than
redhat training)

Just though of it.


POWER TO THE *



On Thu, 2004-07-08 at 20:56, James H. Thompson wrote:
> >From: "William Boehlke" <[EMAIL PROTECTED]>
> >We encourage anyone who is not an Asterisk beginner to send $50 to the Wiki
> >instead of buying the book,
> 
> Nice suggestion.
> The Wiki is not really in need of donations at the moment.  It is being fully 
> sponsored by
> www.commpartners.us
> So I would encourage you instead of sending money,  to spend some time adding or 
> updating content on
> the Wiki (www.voip-info.org)
> 
> I'd also like to acknowledge all of the many contributors that have made the Wiki 
> into a useful
> resource.
> 
> Thanks.
> 
> Jim
> 
> James H. Thompson
> [EMAIL PROTECTED]
> 
> 
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Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Alberto Fernandez]

Wow,
I admire you James, that was a a gesture. Thanks to the good people at
commpartners.com.

Thanks for wiki, probably saved my job and a big headache a couple of
times. Nevertheless i plan to buy the book.

PEOPLE TOMORROW NINTH WE SHALL ALL DRINK A BEER AT 10:00 ON BEHALF OF
THE WIKI.

nah, i don't want all your wife's to come after me after you tell them
that i said you must do this. So just go on with your lifes. also I'm
nobody to suggest something like this :-P


But thanks people @ voip-info.com and people that write to voip-info.com


On Thu, 2004-07-08 at 20:56, James H. Thompson wrote:
> >From: "William Boehlke" <[EMAIL PROTECTED]>
> >We encourage anyone who is not an Asterisk beginner to send $50 to the Wiki
> >instead of buying the book,
> 
> Nice suggestion.
> The Wiki is not really in need of donations at the moment.  It is being fully 
> sponsored by
> www.commpartners.us
> So I would encourage you instead of sending money,  to spend some time adding or 
> updating content on
> the Wiki (www.voip-info.org)
> 
> I'd also like to acknowledge all of the many contributors that have made the Wiki 
> into a useful
> resource.
> 
> Thanks.
> 
> Jim
> 
> James H. Thompson
> [EMAIL PROTECTED]
> 
> 
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Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

[James H. Thompson]

>From: "William Boehlke" <[EMAIL PROTECTED]>
>We encourage anyone who is not an Asterisk beginner to send $50 to the Wiki
>instead of buying the book,

Nice suggestion.
The Wiki is not really in need of donations at the moment.  It is being fully 
sponsored by
www.commpartners.us
So I would encourage you instead of sending money,  to spend some time adding or 
updating content on
the Wiki (www.voip-info.org)

I'd also like to acknowledge all of the many contributors that have made the Wiki into 
a useful
resource.

Thanks.

Jim

James H. Thompson
[EMAIL PROTECTED]


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RE: [Asterisk-Users] Asterisk Book

[Paul Mahler]

I ordered a copy, but they said it's six weeks or so 'till delivery. 

Paul


Paul Mahler 
[EMAIL PROTECTED]   
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
 94107-1901

 Asterisk Services and Training

 

 

 

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of pat munis
> Sent: Thursday, July 08, 2004 11:15 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk Book
> 
> does anyone know anything about this Book?
> - Original Message -
> From: [EMAIL PROTECTED]
> Date: Mon, 5 Jul 2004 00:15:44 +0600 (MDT)
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk Book
> 
> > If anyone is interested in getting a book on asterisk I would 
> > recommend checking out  http://www.saww.net/asterisk/
> > 
> > 
> > 
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[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #4460 - 14 msgs

[Jason Kawakami]

> Message: 13
> Date: Fri, 9 Jul 2004 11:42:01 +1200 (NZST)
> From: =?iso-8859-1?q?Eugen=20Cristea?= <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] asterisk to asterisk config
> Reply-To: [EMAIL PROTECTED]
>
> Hi,
>
> I would like to set two separate asterisks to talk to
> each other.
> Any suggestions?
> I'm a "baby" asterisk fan, only started to play two
> weeks ago, first managed to use kphone with asterisk
> and a  X100P card that is up and running as well.
>
> Thanks,
> Eugen
>
you need to build a trunk between them.  iax trunks work great for this.
Add something like this to your iax.conf

[box1];this goes into the config of box2
type=friend
host=192.168.x.x;address of box1
secret=blah
context= (default)
trunk=yes

[box2];this goes into the config of box1
type=friend
host=192.168.x.x;address of box2
secret=blah
context= (default)
trunk=yes

then you need to register with each other

in box1 iax.conf

register=box1:[EMAIL PROTECTED]:5036

and in box2 iax.conf

register=box2:[EMAIL PROTECTED]:5036

that will get you trunked together.  then you need to build something to
dial out that trunk in the extensions.conf like

exten=_2xx,1,Dial,(IAX2/box1:[EMAIL PROTECTED]/${EXTEN}
;assuming that box2 has 2xx extensions, this is for box1's extensions file

exten=_1xx,1,Dial(IAX2/box2:[EMAIL PROTECTED]/${EXTEN}
;assuming that box1 has 1xx extensions, this is for box1's extensions file

that should give you an idea.  the syntax might not be perfect but it should
get you going down the path.

when in doubt, check the wiki  www.voip-info.org or google for something
like IAX trunking.   you are not the first person to do this.

good luck


Jason Kawakami

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[Asterisk-Users] Asterisk receives TMC Labs Internet Telephony Innovation Award

[James H. Thompson]

Asterisk receives TMC Labs 
Internet Telephony Innovation Award
http://www.tmcnet.com/it/0704/tmclabs.htm
 
Jim
 
James H. Thompson[EMAIL PROTECTED]

Re: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6

[Anton Tinchev]

Joe Baptista wrote:
I'm installing the new Slackware 10.0 distribution - but not sure if i
should go with the 2.4 kernal - which i think is the default install - or
the new 2.6 kernal?  anyone running * and slackware 10.0 with 2.6 kernal?
thanks
joe
Asterisk stable CVS with slack 10/2.4.25 custom kernel - it works rock 
stable
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Re: [Asterisk-Users] Updated Grandstream configurator

[Neil Cherry]

Stephen R. Besch wrote:
The most recent version of GSConfigure is available at 
www.buffalo.edu/~sbesch  Several serious bugs that kept the program from 
getting started have been ferreted out and corrected with the help of 
Bruce Komito. The program is now actually running on someone's machine 
other than mine. I have built this version with the oldest copies of the 
system dll's that I could find inn an effort to solve the VB setup bug, 
so, hopefully it will no longer send anyone through multiple restarts. 
You should have at least SP3, or even better, SP4 on Win2k. I believe it 
will run on Win9x, but I have not tested it and can make no guarantees.
Thanks, I've been having real trouble with those stupid DLLs. I can't
upgrade some of them no matter what I do (WIN2K)!
--
Linux Home Automation Neil Cherry[EMAIL PROTECTED]
http://home.comcast.net/~ncherry/   (Text only)
http://linuxha.sourceforge.net/ (SourceForge)
http://hcs.sourceforge.net/ (HCS II)
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RE: [Asterisk-Users] Monitor cmd and Queues

[brian]

Get the lastest CVS head it can start monitoring from time the agent picks
up the phone.  So you get ZERO hold music.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Harold Workman
> Sent: Thursday, July 08, 2004 5:55 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Monitor cmd and Queues
>
> Hi,
>
> Id like to record a queue conversation using the Monitor command but the
> problem im running into is the way i configure it * records the music on
> hold along with the conversation.  Is there a way to start recording when
> the call is picked up by an agent?  Could someone give me a small example
> of
> how they set this up.  My current extensions.conf portion of the file
> looks
> like:
>
>
>
>
>
> [cytelbilling]  ;# Cytel Communications Billing Support
> ###
> exten => s,1,Background(/usr/src/asterisk-sounds/sounds/this-call-may-be)
> exten => s,2,Background(/usr/src/asterisk-sounds/sounds/monitored)
> exten => s,3,Background(/usr/src/asterisk-sounds/sounds/or)
> exten => s,4,Background(/usr/src/asterisk-sounds/sounds/recorded)
> exten =>
> s,5,Background(/usr/src/asterisk-sounds/sounds/for-quality-purposes)
> exten => s,6,Answer
> exten => s,7,SetMusicOnHold(default)
> exten => s,8,DigitTimeout,5
> exten => s,9,ResponseTimeout,10
> exten => s,10,Background(/usr/src/asterisk-sounds/sounds/pls-stay-on-line)
> exten => s,11,Monitor(wav,,m)
> exten => s,12,Queue(cytelcs)
>
> ---
> Harold Workman
> CCNA, CCNP
> Cytel Communications
> [EMAIL PROTECTED]
> Ph. 281-449-4000 x3098
>
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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Sathya]

> $50 is a bit steep for a book

Usually author of a book makes 10 to 15 percent of the cover price. So
whoever who wrote this book will get 5 bucks a book. Unlike a book widely
used, for an example PHP or MySQl, which will sell lot more copies, a book
on * will have a narrower readership. Therefore I think price has to be set
bit higher than a programming book. It may not be helpful for most of us,
but I guess the effort is praiseworthy, knowing that he can't make a fortune
out of it.

SW

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of mattf
> Sent: Thursday, July 08, 2004 11:47 AM
> To: '[EMAIL PROTECTED]'
> Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
>
>
> Hello,
>
> I was a reviewer of the book, and it is a good beginner book. At over 260
> pages it isn't exactly small and it does contain a lot of information that
> is not in the wiki, especially on telco topics and a very
> in-depth overview
> of Cisco IP phones.
>
> It does a good job of going over the basics of Asterisk, which in
> itself is
> a difficult task as some of the basics of Asterisk change rather
> frequently.
>
>
> It is a beginner book and doesn't cover some more advanced and more
> specialized topics, but it would be a good way for someone who had no
> understanding of the telco/VOIP world to get into Asterisk and
> set up their
> first test system.
>
> $50 is a bit steep for a book, but I can tell you that they did
> put a lot of
> effort into creating a comprehensive intro to Asterisk with a
> great deal of
> the material being written by the authors from scratch, and Asterisk is
> still quite a niche market relative to other cheaper technology books out
> there justifying the cost a bit more.
>
> MATT---
>
>
> -Original Message-
> From: Neil Cherry [mailto:[EMAIL PROTECTED]
> Sent: Thursday, July 08, 2004 11:58 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] FINALLY! a good book about Asterisk.
>
>
> Joe Babstock wrote:
> > There is finally an introductory book about Asterisk!
> > It looks like Paul Mahler at www.signate.com wrote it
> > with a lot of help from Digium. I looked at the sample
> > pages, it looks great.
>
> And how do you know it's a good book? I wouldn't mind a
> review and I may purchase the book (I doubt I qualify
> as a reviewer as I haven't yet figured this VoIP stuff
> out yet). I'm not really sure a few pages qualifies for
> a review. BTW, please excuse me if Paul is a frequent
> contributor to the mail list. I just found the method
> of announcement a bit suspect (I'm not say Paul posted
> this either).
>
> --
> Linux Home Automation Neil Cherry[EMAIL PROTECTED]
> http://home.comcast.net/~ncherry/   (Text only)
> http://linuxha.sourceforge.net/ (SourceForge)
> http://hcs.sourceforge.net/ (HCS II)
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[Asterisk-Users] asterisk to asterisk config

[Eugen Cristea]

Hi,

I would like to set two separate asterisks to talk to
each other.
Any suggestions?
I'm a "baby" asterisk fan, only started to play two
weeks ago, first managed to use kphone with asterisk
and a  X100P card that is up and running as well.

Thanks,
Eugen

Find local movie times and trailers on Yahoo! Movies.
http://au.movies.yahoo.com
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[Asterisk-Users] Repair/disassembly instruction Cisco 7960

[Kevin]

I have a Cisco 7960 with a weak hookswitch spring that works
intermittently.  Has anyone taken the back cover off the unit and can
advise suggestions or the procedure to open her up?

Thanks,

Kevin





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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

[William Boehlke]

As an interested party to the Mahler book VoIP Telephony with Asterisk, I
would like to clarify a point about it. 

We encourage anyone who is not an Asterisk beginner to send $50 to the Wiki
instead of buying the book, and we acknowledge the Wiki (that's where the
confusion began) as a source of some material. That acknowledgement is in
one of the sample pages on our site, which probably began the confusion. But
the book is the 320 page product of nine months of independent work. 

We built a dozen systems during the course of writing it to test and
implement features and options. Mark Spencer and the Digium technical staff
contributed information, advice and equipment. Eight competent engineers,
some of them active here every day, gave it a technical review. 

As a former beginner, in my opinion the index alone is worth the price
compared to the time it took me to locate information.  

Thanks for listening.

William Boehlke
Signate


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of usedcanon
Sent: Thursday, July 08, 2004 2:20 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Harold Workman
Sent: 08 July 2004 20:15
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk.


what does that have to do with an overpriced book?
and i agree with Joe.  With this book sourcing most of the documentation
directly from wiki, why pay for something thats free?  Id rather donate $49
to keeping wiki free to the enviroment.


I second that, I think a more reasonably priced book in PDF fromat would
have been better.

Umar.

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Re: [Asterisk-Users] Small Linux Distro

[Andy Powell]

On 08/07/2004 at 18:41 Philipp von Klitzing wrote:

>and you'll find a link to the "Asterisk Live! CD-ROM".
>
>If you have a moment I guess the list (and certainly me) would be 
>interested to hear about your experiences with this. :-)

Awww c'mon, it's only 29mb download it and try it for yourself

I'd be interested in anyone who has done this and if it worked ok with 
their hardware (or not) and what the system config was...

Rgds

Andy


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RE: [Asterisk-Users] E100P

[Nik Martin]

Ing. Angel Gomez wrote:
>   Thank's to all.
> 
>   - The card came WITHOUT ANY documentation, it was not buy directly
> from digium, they did not have any in stock.
>   - I usually go thru all the messages of this user list, maybe I
> overlook at one with the same question.
>   - The pictures in www.digium.com and in the product sheet are not
> good, both E1 and T1 look the same and they were taken from a
> perspective that don't show detail.
> 
>   And well, I put it on the computer and run zttool, it shows it as an
> E1 no matter how I configure /etc/zaptel.conf.
> 
>   Regards to all who answer for the help.
> 
> Jeremy McNamara wrote:
> 
>> Andres wrote:
>> 
>>> Ing. Angel Gomez wrote:
>>> 
 Hi, i just received an E100P, this is the first one I have ever
 seen, and notice that the board reads T100P. Is this right ?
>>> 
>>> 
>>> 
>>> I think this was asked just a few days ago...the answer is YES.
>> 
>> 
>> 
>> 
>> If people would read the included documentation from Digium they
>> would have known this little fact. 
>> 
>> 
>> 
>> Jeremy McNamara


FYI, I bought a T100P directly from Digium and it came wrapped in bubble
wrap in a plain white box.  No docs, NOTHING.

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[Asterisk-Users] Monitor cmd and Queues

[Harold Workman]

Hi,

Id like to record a queue conversation using the Monitor command but the
problem im running into is the way i configure it * records the music on
hold along with the conversation.  Is there a way to start recording when
the call is picked up by an agent?  Could someone give me a small example of
how they set this up.  My current extensions.conf portion of the file looks
like:





[cytelbilling]  ;# Cytel Communications Billing Support
###
exten => s,1,Background(/usr/src/asterisk-sounds/sounds/this-call-may-be)
exten => s,2,Background(/usr/src/asterisk-sounds/sounds/monitored)
exten => s,3,Background(/usr/src/asterisk-sounds/sounds/or)
exten => s,4,Background(/usr/src/asterisk-sounds/sounds/recorded)
exten =>
s,5,Background(/usr/src/asterisk-sounds/sounds/for-quality-purposes)
exten => s,6,Answer
exten => s,7,SetMusicOnHold(default)
exten => s,8,DigitTimeout,5
exten => s,9,ResponseTimeout,10
exten => s,10,Background(/usr/src/asterisk-sounds/sounds/pls-stay-on-line)
exten => s,11,Monitor(wav,,m)
exten => s,12,Queue(cytelcs)

---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098

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[Asterisk-Users] GS & DTMF in voicemail with CVS of today!?

[Philipp von Klitzing]

Hi there,

it seems that current CVS introduces a problem with DTMF (password) 
detection for VoiceMailMain. Tested with Grandstream and SIP INFO. This 
has worked nicely for the past six months and different CVS versions...

Anyone else seeing this? Only about 50% of the digits entered seem to 
arrive at Asterisk.

CVS-HEAD-07/08/04-22:40:04

Cheers, Philipp


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RE: [Asterisk-Users] Fax Detection

[Matt]

Cheers James.

Its been a while since I did some C coding.

Matt 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Golovich
Sent: 08 July 2004 21:08
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Fax Detection



On Thu, 8 Jul 2004, Ryan Courtnage wrote:

> AFAIK it's the digium card that _detects_ the fax, and allows the call 
> to jump to the 'fax' extension.  So fax _detection_ is a function of 
> the card/driver .. and using the 'fax' extension requires the use of a 
> digium card.
> 
> SpanDSP just talks fax .. I don't think it actually does any detection.

It's not the card that detects the fax.  Its the builtin code in asterisk
that does it (dsp.c).  chan_zap.c is currently the only channel driver that
uses the faxdetection but in theory it could be enabled/used in other
channel drivers as well

James

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Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Daryl Jones]

I've been reading drafts of this book for at least nine months and can 
assure you that its content is very different than what is available on 
the Wiki. The book is an excellent introduction to VOIP in general, and 
offers sufficient information for the novice to configure a basic 
Asterisk system.  Advanced Asterisk users will still need the Wiki and 
mailing list archives.

Perhaps a future edition of the book might cover more advanced topics, 
but the first edition is intended for beginners.

Perhaps the most important thing that this book will accomplsih is to 
increase general awareness about Asterisk being a very reliable, 
full-featured PBX.


Harold Workman wrote:
what does that have to do with an overpriced book?
and i agree with Joe.  With this book sourcing most of the documentation
directly from wiki, why pay for something thats free?  Id rather donate $49
to keeping wiki free to the enviroment.
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Re: [Asterisk-Users] internal & external SIP

[Soren Rathje]

> Hi all,
> I've got a problem with external sip clients.
> My * box has 2 nics, one to my internal network and one on a public IP.
> There are external sip clients (on public IPs) and internal clients on the 
> internal nic.
> both clients can register fine.
> I can phone external clients from the internal clients and the connection 
> works perfectly.
> But if an external client phones an internal one, the internal phone rings, 
> but when the phone is picked up the external call disappears.
> Both internal and external have canreinvite=no
> 
> Can anyone give me any ideas where to start looking into this.
> 

bindaddr = 0.0.0.0   ; Local interface
externip = xxx.xxx.xxx.xxx   ; Public IP address
localnet = 192.168.0.0/255.255.0.0   ; All RFC 1918 addresses are local networks
localnet = 10.0.0.0/255.0.0.0; Also RFC1918
localnet = 172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet = 169.254.0.0/255.255.0.0   ; Zero conf local network

Also, I saw some fixes to RTP address binding in CVS today. Hard to tell really 
without a trace..

-- Soren

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RE: [Asterisk-Users] Best VOIP PSTN Provider.

[Jay Milk]

I'd have to second broadvoice, although I don't think their support is
quite as excellent -- sometimes it takes a while to get a response from
them via email, but calling is generally pretty quick.

While I don't like Vonage as a whole, they do have more rate-centers
than others, so if you need a line someplace where you can't get BV or
anyone else, Vonage Softphone is a workable option.  You'll need to
signup for a hardline ($15/month) and then add any number of softlines
($10/month,500min) which work directly with asterisk.  One undocumented
(read: could be disabled someday), but welcome feature is that there
doesn't seem to be a limit on incoming calls over your softphone number,
making call-hunt and roll-over obsolete.

> -Original Message-
> From: Joe Babstock [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, July 08, 2004 3:12 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Best VOIP PSTN Provider.
> 
> 
> We have been using Broadvoice, www.broadvoice.com.
> Good price, excellent support. 
> 
> --- Carlos Arnt <[EMAIL PROTECTED]> wrote:
> 
> -
> 
> 
> Hi People,
> 
>  
> 
> I wondering here, who is the best VOIP PSTN Provider
> to use with my * box ?
> 
> That has good prices, good quality (Use ex. G729
> codecs) etc ?
> 
>  
> 
> I want to make calls to Europe, Asia etc with cheap
> prices and good quality in the sound ;)
> 
>  
> 
> Did anyone has a good tip for me ?
> 
>  
> 
> For use with IAX2 or SIP and with g.729 codec.
> 
>  
> 
> Thanks alot !
> 
>  
> 
>  
> 
> Carlos. ___Asterisk-Users
> mailing 
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[Asterisk-Users] Intermittent SIP 404 Not Found response?

[Bruce Komito]

I have several SIP devices (Sipuras) that are working fine with *, except
for one annoying little problem.  Occassionally, after being registered
for some period of time, the Sipura returns a 404 Not Found to (I assume)
an INVITE request.  Of course, this makes the extension appear busy.
When this happens, I check the Sipura and it is thinks it is still
registered and I check * and it shows registered.  If I reboot the Sipura
or restart *, the problem clears.  It also clears by itself eventually.

Has anyone seen this behaviour and/or know how to cure it?

TIA

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815





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Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Steve Totaro]

also, PDFs are nice and searchable.

I am sure the motive behind a paper book is to limit people "sharing" it or
it ending up on Limewire or something.


- Original Message - 
From: "Brian Weaver" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, July 08, 2004 3:27 PM
Subject: Re: [Asterisk-Users] FINALLY! a good book about Asterisk.


> When it shows up on Amazon.com, I may pick one up, but I don't
> see it there yet.
>
>
>
> Doug Harris <[EMAIL PROTECTED]> [2004-07-08 09:12:56 -0700]:
> > we should buy it and encourage everyone to do so, that will support
whoever
> > took the initiative to write a book on Asterisk,  which has been long
> > overdue.
> >
> > DH
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] Behalf Of Steve
Totaro
> > > Sent: Tuesday, June 08, 2004 8:16 AM
> > > To: [EMAIL PROTECTED]
> > > Subject: Re: [Asterisk-Users] FINALLY! a good book about Asterisk.
> > >
> > >
> > > I think I will pass.  $49 for something free on the wiki seems too
> > > expensive.  A cheaper PDF would would save a tree and probably be more
> > > reasonable in cost.
> > >
> > >
> > > - Original Message -
> > > From: "Joe Babstock" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>;
<[EMAIL PROTECTED]>;
> > > <[EMAIL PROTECTED]>
> > > Sent: Thursday, July 08, 2004 10:05 AM
> > > Subject: [Asterisk-Users] FINALLY! a good book about Asterisk.
> > >
> > >
> > > > There is finally an introductory book about Asterisk!
> > > > It looks like Paul Mahler at www.signate.com wrote it
> > > > with a lot of help from Digium. I looked at the sample
> > > > pages, it looks great.
> > > >
> > > >
> > > >
> > > > __
> > > > Do you Yahoo!?
> > > > New and Improved Yahoo! Mail - Send 10MB messages!
> > > > http://promotions.yahoo.com/new_mail
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> > >
> > >
> > >
> > >
> >
> >
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Re: [Asterisk-Users] Dialing an extension during Dial()

[programmer_ted]

I think he was referring to allowing the user to enter an extension while 
the other phone is still ringing (before it's picked up).  I don't think 
this functionality is already included (even after pressing #), so I'm 
going to add it.  I agree that it would be nice to allow extensions to be 
entered without pressing the # button first, but I think that's in there so 
that accidentally pressing a button won't transfer the call.

At 10:49 AM 7/8/2004, you wrote:
as an alternative, you can use dial with the T or t option and then have
the user press # for a blind transfer; this way you can manipulate
extensions easily.
l.
ps. would surely be nice to have Dial() be able to recognize other
extensions, just like it now doe4s with #.
l.

In data Wed, 07 Jul 2004 15:51:27 -0700, programmer_ted
<[EMAIL PROTECTED]> ha scritto:
I don't know if my last message went through, but the gist of it was
that currently, this support is lacking from the Dial command.  After
looking at the source for a few minutes, this is something I'd like to
undertake, and it shouldn't take longer than a night, so I'll be
hopefully getting back to you with a patch tonight or tomorrow.  Also,
I'll be submitting the patch to the bugfix list when it's done, so we
may see this feature in the CVS in the future.
At 08:06 AM 7/7/2004, you wrote:
I need to be able to have callers cancel a Dial() command by pressing 1
so
that they can leave a message.  I am unable to get this working.  It
seems
that Dial() will only allow the caller to transfer after the call has
been
connected or after the Dial() command itself has timed out.  Can someone
please help?
-Ryan
--
Creato con M2, il rivoluzionario client e-mail di Opera:
http://www.opera.com/m2/
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Re: [Asterisk-Users] sample config file for GS BT101?

[Steve Totaro]

I am not really sure what you are trying to accomplish.

If its the GAPS alternative, the reason why its not on there is they sell
GAPS so yeah, its reverse engineering unless you care to pay for their
system.


- Original Message - 
From: "Stephen J. Wilcox" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, July 08, 2004 4:01 PM
Subject: Re: [Asterisk-Users] sample config file for GS BT101?


> I was wondering about that too..
>
> Following the instructions on that page for config did not work for me.
Setting
> up a config file like the sample one made no difference to the phone (I
can
> confirm it did tftp it okay). Also the method references md5 checks and I
dont
> see that at all.
>
> I tried the downloads, we wouldnt do this from windows so need to know how
to
> do this to write for *nix but I couldnt get the windows app to run on
XP/2000
> machines altho apparently it will run on 98 but I wasnt able to test that
with a
> phone.
>
> So - is it literally just supposed to be a case of creating a blah=blah
style
> config file at .txt ??
>
> I note not all the options are listed in the sample, what about the
others?
>
> And finally.. why doesnt this info appear to be available from the
manufacturer,
> surely we shouldnt be reverse engineering?
>
> Steve
>
> On Thu, 8 Jul 2004, Steve Totaro wrote:
>
> > http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone
> >
> > the wiki seems to be VERY complete when it comes to GS
> > - Original Message - 
> > From: "Bruce Komito" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Thursday, July 08, 2004 9:31 AM
> > Subject: [Asterisk-Users] sample config file for GS BT101?
> >
> >
> > > If you have an example of a config file for a Grandstream BT101/102, I
> > > would appreciate if you would share it with me.
> > >
> > > Thanks
> > >
> > > Bruce Komito
> > > High Sierra Networks, Inc.
> > > www.servers-r-us.com
> > > (775) 236-5815
> > >
> > >
> > > ___
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[Asterisk-Users] Updated Grandstream configurator

[Stephen R. Besch]

The most recent version of GSConfigure is available at 
www.buffalo.edu/~sbesch  Several serious bugs that kept the program from 
getting started have been ferreted out and corrected with the help of 
Bruce Komito. The program is now actually running on someone's machine 
other than mine. I have built this version with the oldest copies of the 
system dll's that I could find inn an effort to solve the VB setup bug, 
so, hopefully it will no longer send anyone through multiple restarts. 
You should have at least SP3, or even better, SP4 on Win2k. I believe it 
will run on Win9x, but I have not tested it and can make no guarantees.

Steve Besch
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[Asterisk-Users] Re: sample config file for GS BT101?

[Stephen R. Besch]

Stephen J. Wilcox wrote:
I was wondering about that too..
Following the instructions on that page for config did not work for me. Setting 
up a config file like the sample one made no difference to the phone (I can 
confirm it did tftp it okay). Also the method references md5 checks and I dont 
see that at all.

I tried the downloads, we wouldnt do this from windows so need to know how to 
do this to write for *nix but I couldnt get the windows app to run on XP/2000 
machines altho apparently it will run on 98 but I wasnt able to test that with a 
phone.

So - is it literally just supposed to be a case of creating a blah=blah style 
config file at .txt ??

I note not all the options are listed in the sample, what about the others? 

And finally.. why doesnt this info appear to be available from the manufacturer, 
surely we shouldnt be reverse engineering?

Steve
Here's the format of the file. It works perfectly in the GSConfiguration 
program I have posted to the list. I believe that this format is complete.

Each entry is Function, length, format.
File Length, Dword, Bigendian
	The length of the file in WORDS. Length is null padded to even number 
of words

Checksum, Word, BigEndian
Value added which makes the 16-bit sum of the file equal to 0
MAC, Byte[6]
Unique hardware address of SIP device.
CRLF,CRLF
4 bytes set to "0D,0A,0D,0A"
Body
	8-bit ASCII string which contains the phones parameter data in am "&" 
delimited list. Each parameter is of the form Px=value, where x is an 
integer and value is the contents of the parameter. The parameter may be 
null or missing. If the parameter is null, the coresponding value is set 
to null. If it is missing, the currently set value in the phone is used 
(that is, only those parameters present in the file are changed). The 
order of the parameters appears to have no significance (except for 
gnkey, maybe - see below). Example: 
"...&P12=110&P17=0&P18=0&P19=0&P20=0&...". There is no separator either 
before the first parameter or following the last. In addition, some 
preliminary testing suggests that parameters which have no corresponding 
phone parameter are ignored. This theoretically makes it possible to 
include user customized data in the cfg file to be used for other 
purposes, such as storing a person's room number, etc. This could be 
quite useful in cfg management programs. Example: "...&P20=0&Room=Oval 
Office&..."

gnkey=0B82
	This must be specified as the last parameter. As with the others, it is 
specified as text, and separated from the previous parameter with the 
"&" separator. I have not tested this, but I assume that it must appear 
last in the list.

Terminal Null
	This is present only if the parameter text string ends on an odd byte 
boundary. It is added to make the file an even number of words long for 
the checksum routine. I suspect that it really can be any value you like 
(I have not tried this, by the way), as long as it is included in the 
file length calculation so that it will not turn up as a random value, 
or, worse, cause a file read overflow when the phone attempts to process it.
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RE: [Asterisk-Users] internal & external SIP

[Ian D. Wlloughby]

I am guessing the problem is that your internal clients can see the
external SIP clients but not the other way round. The clients have to be
able to make a physical connection to each other. You are not using any
NAT capabilities I guess as your internal clients have their own network
to access the server on. If you set nat on in sip.conf for one of your
internal clients and get it to register on the public network, does this
work?


R's
Ian


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Lawrence
Sent: 08 July 2004 21:00
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] internal & external SIP

Hi all,
I've got a problem with external sip clients.
My * box has 2 nics, one to my internal network and one on a public IP.
There are external sip clients (on public IPs) and internal clients on
the internal nic.
both clients can register fine.
I can phone external clients from the internal clients and the
connection works perfectly.
But if an external client phones an internal one, the internal phone
rings, but when the phone is picked up the external call disappears.
Both internal and external have canreinvite=no

Can anyone give me any ideas where to start looking into this.

Regards,
Jon

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RE: [Asterisk-Users] strange problem with oh323 loaded!

[T. Chan]

Hi,

I am the unlucky one, I have similar problem, but I am mostly using
safe_asterisk, and this "stop now"..."restart now" never works, with neither
0.6.3 nor 0.6.2

TC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anthony Law
Sent: Thursday, July 08, 2004 3:33 PM
To: Mailing List Asterisk
Subject: [Asterisk-Users] strange problem with oh323 loaded!


Hi,

As explained in my original post on June 30. When I used CVS 2004-06-16 with
oh323-0.6.3a.  I can compile and install without problem but when I am in
the asterisk console whenever I issue "stop now" or "restart now" or
"extension reload" I got stuck on the console and asterisk did not response
to either shutting down or restarting.

It stucked on

Executing last minute cleanups
  == Cleaning up OpenH323 channel driver.
  == Unregistered channel type 'OH323'
  == Destroying any remaining musiconhold processes

 The same thing will not happen if I do not load the oh323-0.6.3.a module.
Since I have this problem I have gone back to oh323-0.6.3 and it acts the
same, finally yesterday I revert it back to oh323-0.6.2a and the above did
not happen. Do you happen to know why?



Regards,



Anthony


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Re: [Asterisk-Users] advanced audio recording agi help

[Steven Critchfield]

On Thu, 2004-07-08 at 14:13, Tony Buser wrote:
> I'm thinking about doing a project using asterisk that would let someone 
> call in and save the audio to a wav file.  I know it can be done using 
> Record() or zapbarge.  However, I'd like to be able to do some more 
> complicated/interactive things such as:
> 
> 1. record some audio
> 2. play it back
> 3. pause the playback in the middle
> 4. start recording again in the middle where I had paused it
> 5. skip forward or back x number of seconds within the recording
> 6. maybe even hit a button to delete the audio from this point back or 
> this point forward
> 
> Kind of like duplicating the basic functionality of windows sound 
> recorder inside asterisk.  What we're looking to do is to rewrite our 
> custom built dictation system which is currently windows-only.  I was 
> wondering if anyone has done anything like this or if anyone has any 
> ideas on how I would go about doing such a thing?

The company I work for has already written that app and is in the
process of marketing it. We already have one clinic using it, and we are
about to move ~5000 more dictators to it now that we consider it
production quality code.

If you like, contact [EMAIL PROTECTED] for more information.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] SNMP Monitoring

[GIBERT Frédéric]

Title: SNMP Monitoring






Hello,

Does someone know how to setup snmp monitoring on asterisk. I’ve plan to deploy 50 asterisk, so I need some monitoring tools.

I try with nagios as I read in the wiki, there is some project on it, but I can’t reach the end.

Can someone help me?

Thanks.



GIBERT Frédéric

Ste VigiNetworks

Mobile: +33 6 72 08 35 16

RE: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6

[usedcanon]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Zdenek
Bouresh
Sent: 08 July 2004 22:33
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6


mattf wrote:

>Hello,
>
>I am running Asterisk on Slackware 10.0 with the 2.4 kernel(default kernel)
>and it is very happy. Don't see too much difference from 9.1 except for the
>fact that most of the binutils have been updated and several of them run
>differently now(top, ps, ...)
>
>Haven't tried the 2.6 kernel yet, but may try it later.
>
>MATT---
>
>
>-Original Message-
>From: Joe Baptista [mailto:[EMAIL PROTECTED]
>Sent: Thursday, July 08, 2004 10:01 AM
>To: [EMAIL PROTECTED]
>Subject: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6
>
>
>
>I'm installing the new Slackware 10.0 distribution - but not sure if i
>should go with the 2.4 kernal - which i think is the default install - or
>the new 2.6 kernal?  anyone running * and slackware 10.0 with 2.6 kernal?
>
>thanks
>joe
>
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>
Slackware 10 and kernel 2.6.6 and 2.6.7 run fine .

What about MySQL support for SIP friends, CDR and Voicemail ? do they work.

Umar.

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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

[usedcanon]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Harold
Workman
Sent: 08 July 2004 20:15
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk.


what does that have to do with an overpriced book?
and i agree with Joe.  With this book sourcing most of the documentation
directly from wiki, why pay for something thats free?  Id rather donate $49
to keeping wiki free to the enviroment.


I second that, I think a more reasonably priced book in PDF fromat would
have been better.

Umar.

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Re: [Asterisk-Users] multiple days on a GotoIfTime command?

[Mitchel Constantin]

I still don't see why you can't use a script and an array to simplify
this, that way you don't have to work with extensions.conf, just work
on your file, possible php and an array with a loop to check
everything.

-mitchel

On Wed, 7 Jul 2004 20:49:48 -0400, William Suffill
<[EMAIL PROTECTED]> wrote:
> well then lever it db driven and set the #'s in the db and update that
> to the proper call order as needed
> 
> 
> 
> On Wed, 07 Jul 2004 13:51:10 -0300, Gelson Dias Santos
> <[EMAIL PROTECTED]> wrote:
> > The problem is, there is no pattern. It´s not an open/close scenario.
> > This month I need to call NUMBER1, NUMBER2 and NUMBER3 on those days.
> > Next month, who knows? I´ll receive another schedule to implement on
> > asterisk.
> > I see no way to avoid changing those lines each month. What I´m trying
> > to do is reduce the number os files involved.
> >
> > Gelson
> >
> >
> >
> > brian wrote:
> > > I see the pattern.. let me think for a second.. and I'm sure I can get you
> > > something that's simpler than 31 gotoif's
> > >
> > >
> > > bkw
> > >
> > >
> > >>-Original Message-
> > >>From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > >>[EMAIL PROTECTED] On Behalf Of brian
> > >>Sent: Tuesday, July 06, 2004 5:24 PM
> > >>To: [EMAIL PROTECTED]
> > >>Subject: RE: [Asterisk-Users] multiple days on a GotoIfTime command?
> > >>
> > >>You're making this WAY too complicated its simpler than you can even
> > >>imagine.
> > >>
> > >>Mind answering my original question first?  WHAT THE HECK is the pattern
> > >>your logic?  What times are you open.. what times are you closed?  What?
> > >>
> > >>
> > >>bkw
> > >>
> > >>
> > >>>-Original Message-
> > >>>From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > >>>[EMAIL PROTECTED] On Behalf Of Roger Gulbranson
> > >>>Sent: Tuesday, July 06, 2004 4:20 PM
> > >>>To: [EMAIL PROTECTED]
> > >>>Cc: Roger Gulbranson
> > >>>Subject: Re: [Asterisk-Users] multiple days on a GotoIfTime command?
> > >>>
> > >>>On Tue, 2004-07-06 at 17:03, Gelson Dias Santos wrote:
> > >>>
> > brian wrote:
> > 
> > 
> > >What are you trying to do?  What is the end result and what hours
> > >>
> > >>are
> > >>
> > >>>you
> > >>>
> > >open?
> > 
> > 
> > Exactly what I said. Need to call a number if time and day matches
> > >>>
> > >>>what
> > >>>
> > is on the rule. This month I have to:
> > 
> > call NUMBER1 if day = 1,2,3,4,5,8,14,17,18,20,23,26,29
> > call NUMBER2 if day = 6,9,10,11,12,15,21,27,30,31
> > call NUMBER3 if day = 7,13,16,19,22,24,25,28
> > 
> > I have it working now using 31 "GotoIfTime" lines, one for each day
> > >>>
> > >>>of
> > >>>
> > month but I would like to optimize it. If I could group all days
> > >>
> > >>related
> > >>
> > to a number somehow, I would end up with just three "GotoIfTime"
> > >>
> > >>lines.
> > >>
> > >>>You are making this way too complicated.
> > >>>
> > >>>Use DBget to retrieve a number which is the extension you want and then
> > >>>dial that extension.
> > >>>
> > >>>Have a cron job (or something similar) set the extension you want via
> > >>>DBset.  You can put all of your time logic into the cron job.
> > >>>
> > >>>There may be even simpler solutions.
> > >>>
> > >>>
> > >>>
> > >>>___
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> > >>
> > >>
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> > >
> > >
> > >
> > > ___
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Re: [Asterisk-Users] Interface to generate Statements?

[Philipp von Klitzing]

Hi!

> > I am not sure what "statements" you want, but look here:
> > http://www.voip-info.org/wiki-Asterisk+GUI
> 
> Yeah, I've looked there already. I guess I'm just thinking of a basic
> interface where you can generate invoices for customers.
> 
> I guess I better start coding :)

Or go and look at the "See also" part on the bottom of this page:
http://www.voip-info.org/wiki-Asterisk+billing

Cheers, Philipp


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Re: [Asterisk-Users] Fax Detection

[James Golovich]

On Thu, 8 Jul 2004, Ryan Courtnage wrote:

> AFAIK it's the digium card that _detects_ the fax, and allows the call 
> to jump to the 'fax' extension.  So fax _detection_ is a function of 
> the card/driver .. and using the 'fax' extension requires the use of a 
> digium card.
> 
> SpanDSP just talks fax .. I don't think it actually does any detection.

It's not the card that detects the fax.  Its the builtin code in asterisk
that does it (dsp.c).  chan_zap.c is currently the only channel driver
that uses the faxdetection but in theory it could be enabled/used in other
channel drivers as well

James

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[Asterisk-Users] Repair/disassembly instruction Cisco 7960

[Kevin]

I have a Cisco 7960 with a weak hookswitch spring that works
intermittently.  Has anyone taken the back cover off the unit and can
advise suggestions or the procedure to open her up?

Thanks,

Kevin





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RE: [Asterisk-Users] H323 channel

[T. Chan]

Dear All,

There is a question about the H323 channels (H323 driver, not OH323), it is
not passing CallerID. If a call comes in on ZAP and out H323 to another
gateway, the other gateway does not see the ANI, and if Asterisk is used as
a passthrough, it receives callerID from the other gateway, but when sent
out to the terminating endpoint, the terminating endpoint is not seeing it.

Is there anywhere we should configure so that it is passing the callerID to
a terminating endpoint on the outbound H323 channels. Please enlighten me,
thanks

TC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris A.
Icide
Sent: Tuesday, July 06, 2004 6:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 channel


On 03:23 AM 7/6/2004, administrator tootai wrote:
 >Hello everybody,
 >
 >my * box is connected to gnugk with H323 channel. If I call from an H323
 >EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio
 >start but noisy (scratch) , then became ok for callee (SIP EP) but still
 >scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323
 >EP and it's ok. And from now, it's also ok when H323 EP call SIP one's!
 >
 >No need to say that H323<->H323 is working, as well as SIP<->SIP.
 >Running CVS version from yesterday. Used codecs are G711U & A, G723.1
 >and G729. If I just use G711 it's the same. SIP EP has to call first
 >when * is started to make it work. Any hint?
 >
 >Also, H323 is still broken and working without FastStart. Is there a
 >workaround existing?

Just to help troubleshooting, which h323 implementation for asterisk did
you use?


-Chris

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[Asterisk-Users] displaying call progress with SendText on a Snom

[Brady Alleman]

Is there a list of phones (hard or soft) that support the Asterisk
SendText or SendURL applications?  I have been trying to make this work
with a Snom 200 and a Cisco 7960, to display call progress information,
such as which trunk a outbound call is routed to, but my attempts have
been unsuccessful.  The Snom claims to support SMS somehow, but searches
reveal no useful information on how it is to be done.  Any tips?

Thanks in advance.
-- 
Brady Alleman
Network Engineer
Cumberland Technologies International

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Re: [Asterisk-Users] Interface to generate Statements?

[Darren Bentley]

On Thu, 2004-07-08 at 13:15, Philipp von Klitzing wrote:
> Hi!
> 
> > Is there any downloadable software to generate Statements from the mysql
> > call log?
> 
> I am not sure what "statements" you want, but look here:
> http://www.voip-info.org/wiki-Asterisk+GUI
> 
> Cheers, Philipp
> 

Yeah, I've looked there already. I guess I'm just thinking of a basic
interface where you can generate invoices for customers.

I guess I better start coding :)

Thanks,

- Darre

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Re: [Asterisk-Users] Using Cisco AS5350 as pstn GW .. one-way audio problem

[Glen Hinkle]

I assume the pstn is your * system.  
Can you get audio both ways if you send the traffic back to *?  

pstn -> as5350 -> pstn ?

-g


On Thu, 2004-07-08 at 14:09, [EMAIL PROTECTED] wrote:
> Hi all.
> 
> I have a strange problem, I've got a AS5350 hooked up to a telco using
> two trunked E1's
> 
> The 5350 should only act as a GW to a sipproxyserver. 
> 
> THe thing is it seems to be only oneway audio?
> 
> There are no firewall at all, and the audio still only get one-way
> 
> When I call from pstn --> as5350 --> sip-sip-phone   I can here the
> sip-phone  ,, but the sipphone cannot her the pstn-phone.
> 
> What could be the reason for this ?
> 
> /Mike
> 
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Re: [Asterisk-Users] E100P

[Ing. Angel Gomez]

Thank's to all.
	- The card came WITHOUT ANY documentation, it was not buy directly from 
digium, they did not have any in stock.
	- I usually go thru all the messages of this user list, maybe I 
overlook at one with the same question.
	- The pictures in www.digium.com and in the product sheet are not good, 
both E1 and T1 look the same and they were taken from a perspective that 
don't show detail.

	And well, I put it on the computer and run zttool, it shows it as an E1 
no matter how I configure /etc/zaptel.conf.

Regards to all who answer for the help.
Jeremy McNamara wrote:
Andres wrote:
Ing. Angel Gomez wrote:
Hi, i just received an E100P, this is the first one I have ever seen, 
and notice that the board reads T100P. Is this right ?

I think this was asked just a few days ago...the answer is YES.


If people would read the included documentation from Digium they would 
have known this little fact.


Jeremy McNamara
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Re: [Asterisk-Users] Best VOIP PSTN Provider.

[Joe Babstock]

We have been using Broadvoice, www.broadvoice.com.
Good price, excellent support. 

--- Carlos Arnt <[EMAIL PROTECTED]> wrote:

-


Hi People,

 

I wondering here, who is the best VOIP PSTN Provider
to use with my * box ?

That has good prices, good quality (Use ex. G729
codecs) etc ?

 

I want to make calls to Europe, Asia etc with cheap
prices and good quality in the sound ;)

 

Did anyone has a good tip for me ?

 

For use with IAX2 or SIP and with g.729 codec.

 

Thanks alot !

 

 

Carlos.
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RE: [Asterisk-Users] Interface to generate Statements?

[Timothy R. McKee]

Darren:

It's not that simple.  You have to decode the other end of the call into a
RateCenter and then have $$ rates in a different database for each rate
center.  I believe this process is called 'rating the call'.  Lest you take
the easy way out and assign all (xxx) xxx- numbers a flat rate, be aware
the some area codes actually connect at overseas rates to places such as the
Carribean area (it is the North American Numbering Plan, after all) and cost
you LOTS of , so you had better charge more for those. 




Timothy R. McKee


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Sent: Thursday, July 08, 2004 12:49
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Interface to generate Statements?

Hello,

Is there any downloadable software to generate Statements from the mysql
call log?

I'm considering developing a PHP/MySql interface but I'm thinking that
someone must have already done this?

Thanks,

- Darren

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RE: [Asterisk-Users] Fax Detection

[Matt]

Sorry it's currently set to both.it's been a long old day!
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington
Sent: 08 July 2004 19:45
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Fax Detection

Hi Ryan,

Faxdetect should be set to one of the following in zapata.conf:

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

I don't believe that faxdetect=yes is valid but I didn't read the source,
just the sample config file.

-Seth

On Thu, 2004-07-08 at 12:28, Matt wrote:
> Hi Ryan,
> 
> Telappliant only offer SIP/IAX inbound I do have an X100P in the PBX 
> and faxdetect=yes in the zapata.conf.
> I suppose my question is: does SpanDSP detect faxes over SIP or is it 
> only over Zap?
> 
> Matt
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ryan 
> Courtnage
> Sent: 08 July 2004 09:47
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Fax Detection
> 
> On July 7, 2004 09:19 pm, Matt wrote:
> > Hi all
> >
> > I've tried Google, wiki and mailing list and IRC but still haven't 
> > gotten to the bottom of this.  Hopefully someone might be able to help.
> >
> > I'm using telappliant to provide my inbound and outbound calls. 
> 
> I'm not familiar with teleppliant.  Do you use a digium (zap) card?  
> AFAIK, you need one and need faxdetect=yes in zapata.conf.
> 
> When a fax comes in, does anything relevant get written to * console?
> 
> > * plays
> > host to 30 cisco's and they are all working great using G711 A-law.  
> > I've managed to get SpanDSP to compile and install and I can send a 
> > receive a fax on a dedicated extension.  What I'm trying to do now 
> > and can't seem to nail is getting an inbound fax to be detected and 
> > then
> handled.
> >
> > I've tried the examples from the wiki and the sites linked on the 
> > wiki; messed about trying my own weird and wonderful methods but 
> > still no
> joy.
> >
> > All the calls are using G711 A-law.
> >
> > Here is the test context I'm using
> >
> > XXX = hiden
> >
> > Exten => 08700686XXX,1,Goto(textextension,7000,1)
> >
> > [testextension]
> > Exten => 7000,1,Answer
> > Exten => 7000,2,Dial(SIP/104)
> > Exten => fax,1,rxfax(/var/spool/asterisk/incoming/testfax.tif)
> > Exten => fax,2,congestion
> > Exten => fax,102,congestion
> >
> > Calls hit the testextension contect but don't get detected as a fax.
> >
> > Cheers
> >
> > Matt
> >
> > ___
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> > [EMAIL PROTECTED]
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> ..
> Ryan Courtnage
> Coalescent Systems Inc
> 403.244.8089
> www.voxbox.ca
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--
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661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559

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RE: [Asterisk-Users] Using Windows Messenger+Video in *

[Florian Overkamp]

Hi, 

> -Original Message-
> Has anybody used Windows Messenger with asterisk?
> All documents around (google - wiki - bugs.digium.com) say 
> that asterisk supports windows messenger with video but i 
> have no succes yet!
> I can establish connection with audio but no video yet.
> I've used a range of windows messengers from version 4.7 to 5.0.0482.

This is a little brief to say. I have had this working properly with recent
asterisk boxes. A few things: Check if the [general] section has
'videosupport=yes' and if the sip peers are allowed to use h261 and h263
codecs.

Best regards,
Florian

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Re: [Asterisk-Users] sample config file for GS BT101?

[Stephen J. Wilcox]

I was wondering about that too..

Following the instructions on that page for config did not work for me. Setting 
up a config file like the sample one made no difference to the phone (I can 
confirm it did tftp it okay). Also the method references md5 checks and I dont 
see that at all.

I tried the downloads, we wouldnt do this from windows so need to know how to 
do this to write for *nix but I couldnt get the windows app to run on XP/2000 
machines altho apparently it will run on 98 but I wasnt able to test that with a 
phone.

So - is it literally just supposed to be a case of creating a blah=blah style 
config file at .txt ??

I note not all the options are listed in the sample, what about the others? 

And finally.. why doesnt this info appear to be available from the manufacturer, 
surely we shouldnt be reverse engineering?

Steve

On Thu, 8 Jul 2004, Steve Totaro wrote:

> http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone
> 
> the wiki seems to be VERY complete when it comes to GS
> - Original Message - 
> From: "Bruce Komito" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, July 08, 2004 9:31 AM
> Subject: [Asterisk-Users] sample config file for GS BT101?
> 
> 
> > If you have an example of a config file for a Grandstream BT101/102, I
> > would appreciate if you would share it with me.
> > 
> > Thanks
> > 
> > Bruce Komito
> > High Sierra Networks, Inc.
> > www.servers-r-us.com
> > (775) 236-5815
> > 
> > 
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[Asterisk-Users] internal & external SIP

[Jon Lawrence]

Hi all,
I've got a problem with external sip clients.
My * box has 2 nics, one to my internal network and one on a public IP.
There are external sip clients (on public IPs) and internal clients on the 
internal nic.
both clients can register fine.
I can phone external clients from the internal clients and the connection 
works perfectly.
But if an external client phones an internal one, the internal phone rings, 
but when the phone is picked up the external call disappears.
Both internal and external have canreinvite=no

Can anyone give me any ideas where to start looking into this.

Regards,
Jon

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RE: [Asterisk-Users] Small Linux Distro

[Nik Martin]

Philipp von Klitzing wrote:
> Hi!
> 
>> Does anyone have a current, stripped linux distro which has only
>> asterisk and net drivers?
> 
> Look here: http://www.voip-info.org/wiki-Asterisk+installation+tips
> 
> and you'll find a link to the "Asterisk Live! CD-ROM".
> 
> If you have a moment I guess the list (and certainly me) would be
> interested to hear about your experiences with this. :-)
> 
> Cheers, Philipp
> 
> P.S.: You might also want to look at this:
> http://www.voip-info.org/wiki-Asterisk+linux+distributions 
> 

IMHO, A Slackware setup CD is as good as it gets.  You can install the dev
tools, bison, a shell, ssh, and kernel headers and have a nice, lean system
without all the fluff.  I got my asterisk server up in < 2 hours from
scratch that way, and am a newcomer to Linux.

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[Asterisk-Users] strange problem with oh323 loaded!

[Anthony Law]

Hi,

As explained in my original post on June 30. When I used CVS 2004-06-16 with
oh323-0.6.3a.  I can compile and install without problem but when I am in
the asterisk console whenever I issue "stop now" or "restart now" or
"extension reload" I got stuck on the console and asterisk did not response
to either shutting down or restarting.

It stucked on

Executing last minute cleanups
  == Cleaning up OpenH323 channel driver.
  == Unregistered channel type 'OH323'
  == Destroying any remaining musiconhold processes

 The same thing will not happen if I do not load the oh323-0.6.3.a module.
Since I have this problem I have gone back to oh323-0.6.3 and it acts the
same, finally yesterday I revert it back to oh323-0.6.2a and the above did
not happen. Do you happen to know why?



Regards,



Anthony


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[Asterisk-Users] advanced audio recording agi help

[Tony Buser]

I'm thinking about doing a project using asterisk that would let someone 
call in and save the audio to a wav file.  I know it can be done using 
Record() or zapbarge.  However, I'd like to be able to do some more 
complicated/interactive things such as:

1. record some audio
2. play it back
3. pause the playback in the middle
4. start recording again in the middle where I had paused it
5. skip forward or back x number of seconds within the recording
6. maybe even hit a button to delete the audio from this point back or 
this point forward

Kind of like duplicating the basic functionality of windows sound 
recorder inside asterisk.  What we're looking to do is to rewrite our 
custom built dictation system which is currently windows-only.  I was 
wondering if anyone has done anything like this or if anyone has any 
ideas on how I would go about doing such a thing?

Thanks!
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Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Brian Weaver]

When it shows up on Amazon.com, I may pick one up, but I don't 
see it there yet.



Doug Harris <[EMAIL PROTECTED]> [2004-07-08 09:12:56 -0700]:
> we should buy it and encourage everyone to do so, that will support whoever
> took the initiative to write a book on Asterisk,  which has been long
> overdue.
> 
> DH
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf Of Steve Totaro
> > Sent: Tuesday, June 08, 2004 8:16 AM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] FINALLY! a good book about Asterisk.
> >
> >
> > I think I will pass.  $49 for something free on the wiki seems too
> > expensive.  A cheaper PDF would would save a tree and probably be more
> > reasonable in cost.
> >
> >
> > - Original Message -
> > From: "Joe Babstock" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]>;
> > <[EMAIL PROTECTED]>
> > Sent: Thursday, July 08, 2004 10:05 AM
> > Subject: [Asterisk-Users] FINALLY! a good book about Asterisk.
> >
> >
> > > There is finally an introductory book about Asterisk!
> > > It looks like Paul Mahler at www.signate.com wrote it
> > > with a lot of help from Digium. I looked at the sample
> > > pages, it looks great.
> > >
> > >
> > >
> > > __
> > > Do you Yahoo!?
> > > New and Improved Yahoo! Mail - Send 10MB messages!
> > > http://promotions.yahoo.com/new_mail
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> >
> >
> 
> 
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[Asterisk-Users] HOW ASTERISK WORKS

[Giscard Fernandes Faria]

Hy guys, I cannot understand How the asterisk works. I
would like know how the h323.conf, sip.conf and
extension.conf works. I don't understand the
parameters and the [sections].

What I need to the asterisk get a SIP call and forward
them to a H323 terminal. I working at the h323.conf
and extension.conf but I cannot understand!!! Please
someone can help me.

I your can send me a example (with comments) of a
simple example working with sip and h323.

Thanks.

Giscard





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Re: [Asterisk-Users] Call queing

[C. Maj]

On Mon, 5 Jul 2004, Jeremy Kenney waxed:

> Hello all I have a issue I am wondering if someone can help me
> 
> Here is my problem,
> 
> I have several queues setup for different numbers I want each queue to play
> a custom message to the caller when calling in and then to the called
> extention when the person answers how is this done and can I specify a
> customer directory for each one

This is possible, just Playback different messages before
Queue'ing them.  Change the music on hold class if you want
to play different stuff while they are waiting.

--Chris


-- 
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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Re: [Asterisk-Users] Asterisk Book

[pat munis]

does anyone know anything about this Book?
- Original Message -
From: [EMAIL PROTECTED]
Date: Mon, 5 Jul 2004 00:15:44 +0600 (MDT)
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk Book

> If anyone is interested in getting a book on asterisk I would recommend
> checking out  http://www.saww.net/asterisk/
> 
> 
> 
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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Harold Workman]

what does that have to do with an overpriced book?
and i agree with Joe.  With this book sourcing most of the documentation
directly from wiki, why pay for something thats free?  Id rather donate $49
to keeping wiki free to the enviroment.


---
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy Kenney
Sent: Thursday, July 08, 2004 12:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FINALLY! a good book about Asterisk.


This is true but school cost money

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, June 08, 2004 11:16 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

I think I will pass.  $49 for something free on the wiki seems too
expensive.  A cheaper PDF would would save a tree and probably be more
reasonable in cost.


- Original Message -
From: "Joe Babstock" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]>;
<[EMAIL PROTECTED]>
Sent: Thursday, July 08, 2004 10:05 AM
Subject: [Asterisk-Users] FINALLY! a good book about Asterisk.


> There is finally an introductory book about Asterisk!
> It looks like Paul Mahler at www.signate.com wrote it
> with a lot of help from Digium. I looked at the sample
> pages, it looks great.
>
>
>
> __
> Do you Yahoo!?
> New and Improved Yahoo! Mail - Send 10MB messages!
> http://promotions.yahoo.com/new_mail
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Re: [Asterisk-Users] Access Bank 2 <---> T100P T1 Cable.

[Steven Critchfield]

On Thu, 2004-07-08 at 11:19, Anton Tinchev wrote:
> I just got my Access Bank 2 (of course i'm happy now:)).
> Just need the wiring scheme

Dude you could have asked google.

google search terms, t1 crossover cable

First link I receive was
http://www.gcom.com/home/support/t1crossover.html

Or you could have used the wiki and found
http://www.voip-info.org/wiki-crossover+T1+cable

Or even a short trip through the mailing list archive with google would
have provided you
http://lists.digium.com/pipermail/asterisk-users/2003-July/015190.html

Don't be a support drain because you are not willing to look for the
information yourself.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Fax Detection

[Ryan Courtnage]

On 8-Jul-04, at 10:28 AM, Matt wrote:
Hi Ryan,
Telappliant only offer SIP/IAX inbound I do have an X100P in the PBX 
and
faxdetect=yes in the zapata.conf.
I suppose my question is: does SpanDSP detect faxes over SIP or is it 
only
over Zap?
AFAIK it's the digium card that _detects_ the fax, and allows the call 
to jump to the 'fax' extension.  So fax _detection_ is a function of 
the card/driver .. and using the 'fax' extension requires the use of a 
digium card.

SpanDSP just talks fax .. I don't think it actually does any detection.

Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan 
Courtnage
Sent: 08 July 2004 09:47
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Fax Detection

On July 7, 2004 09:19 pm, Matt wrote:
Hi all
I've tried Google, wiki and mailing list and IRC but still haven't
gotten to the bottom of this.  Hopefully someone might be able to 
help.

I'm using telappliant to provide my inbound and outbound calls.
I'm not familiar with teleppliant.  Do you use a digium (zap) card?  
AFAIK,
you need one and need faxdetect=yes in zapata.conf.

When a fax comes in, does anything relevant get written to * console?
* plays
host to 30 cisco's and they are all working great using G711 A-law.
I've managed to get SpanDSP to compile and install and I can send a
receive a fax on a dedicated extension.  What I'm trying to do now and
can't seem to nail is getting an inbound fax to be detected and then
handled.
I've tried the examples from the wiki and the sites linked on the
wiki; messed about trying my own weird and wonderful methods but 
still no
joy.
All the calls are using G711 A-law.
Here is the test context I'm using
XXX = hiden
Exten => 08700686XXX,1,Goto(textextension,7000,1)
[testextension]
Exten => 7000,1,Answer
Exten => 7000,2,Dial(SIP/104)
Exten => fax,1,rxfax(/var/spool/asterisk/incoming/testfax.tif)
Exten => fax,2,congestion
Exten => fax,102,congestion
Calls hit the testextension contect but don't get detected as a fax.
Cheers
Matt
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..
Ryan Courtnage
Coalescent Systems Inc
403.244.8089
www.voxbox.ca
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[Asterisk-Users] outgoing caller id from SIP to isdn (p2p)

[Tomaz]

hi,
how to set caller ID for internal SIP users when dialing out on telco 
ISDN p2p (hfc card) line?
I need to setup numbers from 0 - > 9 (10  sip users).internal caller id 
is working correctly .. from 0 to 9 ,but when I dial on isdn telco line 
-> gsm show only our prefix number ( xxY ) , only 6 (x) numbers of  
7 (Y).

here is extension 1 dialing on isdn line but on gsm is only 6 numbers , 
7 number  SIP/1 is not shown!

Executing Dial("SIP/1-4388", "Zap/g3/BYEXTENSION|130|t|r") in new stack
   -- Called g3/041629939
   -- Registered SIP '1' at 194.249.91.197 port 5060 expires 3600
   -- Zap/1-1 is ringing

where to look for this ? any solutions?
tnx
Tomaz
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Re: [Asterisk-Users] tdm400p static - out of ideas

[Ryan Courtnage]

On 8-Jul-04, at 9:37 AM, Jorge Mendoza wrote:
We have not experience with Digium cards. However we had similar 
problems when installing legacy pbx. The problem: local ground. One 
easy way to test the local ground is with a voltmeter measure the 
voltage between the CO tip wire (in open loop state) and local ground. 
This must be less than 2 VDC and must be stable (not oscillations). 
Correcting the local ground all problems were fixed.
When you plug a regular phone, the phone is floating and have not 
electrical contact with other devices, then there are not ground 
currents.
My guess is that there are some hardware design that are more 
sensitive than others to ground currents.
This is very good advice - I will certainly check for this.
Digium has suggested that loading the driver like this will resolve the 
static noise problems I've been seeing:

modprobe wcfxs lowpower=1
I'll report back in a few days with the results.
I'd like to know exactly how this will fix the prob (ie: the chain of 
events that occur to create the problem).


Hope this helps.
Jorge
Ryan Courtnage wrote:
On July 8, 2004 03:22 am, Nicholas Bachmann wrote:
Ryan Courtnage wrote:
Hello,
Over the past several weeks, we have been having an intermittant 
problem
with our deployment of a TDM400P card (4 fxo).  We have tried many
things, and the problem still re-occurs.

The Problem:
Occasionally (every 48 hours), the TDM400p card will stop answering
incoming calls on ALL fxo ports.  Attempts to send outbound calls 
on any
Zap channel will result in hearing a loud 'static' noise on the 
line.
Let's look at some possibilities of line problems:
What time does it stop answering? Is it ever during ALIT times 
(usually
very early morning)?
It's totally random - morning/evening/afternoon.  Once it stops 
answering, that's it, a reboot or module-reload is needed.  If ALIT 
for some reason prevents the card from answering, it should be able 
to recover and begin answering after the ALIT is complete.
Have you tried calling the telco to see if it could be their problem?
When the card goes into the non-functional state, I can plug a 
regular phone into any of the lines and make calls just fine.  After 
verifying working lines and plugging them back into the tdm400p card, 
I still can't dial out (the Zap channel will answer, but I will hear 
only static, and the call to the pstn is never placed).  As well, 
incoming calls will not be answered (* console will not even show the 
'started simple switch on zap/x' message).
How far away from the CO/mux are you?
Not too sure - it's in downtown Calgary - so probably not far.
There is the possibility that _something_ with the phone line is 
triggering the problem.  Maybe it's some noise, an unexpected signal, 
some crosstalk ...  things that will cause unexpected behavior ... 
but also things that shouldn't put the entire card into a 
non-functioning state.
Have you tried a new/different card?  If you didn't want to fork out 
the
cash for a new one, you could try a X100P/knockoff* on one of the 
lines
to see if that fixes the problem... if so you can deduce a bad card.
I may have to push for a replacement tdm400p card from Digium.
Nick
*I usually don't recommend the knockoffs, but for a day of testing 
$10
sure beats $100... everybody else should support Digium! :-)
An acquaintance who is having the same problem has reluctantly 
replaced his card with an openline4.  I would like nothing more than 
to stick with Digium hardware - this thread and obtaining a 
replacement card is my last kick at the cat.
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[Asterisk-Users] IAX2 problems transfering back and forth between pbxes

[Bartek Kania]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi!
I have two pbxes connected using IAX2 and trunk=yes.
I tried the following today: From pbx1 (using a phone connected to a
TDM400P) I call pbx2 and log in via DISA.
Then I continue by calling back to pbx1. This works, and pbx2 seems to
transfer the call just fine and hangs up the channel.
If I then call pbx2 again (ie the call would be
pbx1->pbx2->pbx1->pbx2) it does not work anymore.

The effects of this are that pbx2 uses 100% cpu until the call
disconnects (which happens in about 20sec or so).

I am running CVS head as of today on both boxes.

The relevant error-messages are:
pbx1:  Lots and lots of:
 Jul  8 20:12:35 WARNING[9225]: chan_iax2.c:4893 socket_read: Received
 trunked frame before first full voice frame

pbx2:
 Jul  8 20:12:57 WARNING[9225]: chan_iax2.c:1409 attempt_transmit: Max
 retries exceeded to host x.x.x.x on [EMAIL PROTECTED]/16387
 (type = 2, subclass = 2, ts=65, seqno=2)


I tested this since I will probably be setting up a system using two
pbxes where we will be doing a lot of transfers between pbxes.
I.e. person1 on pbx1 answers and talks, transfers to person2 on
pbx2. Person2 talks some and then transfers back to person1.

Is this a known bug? An unknown bug? Or user(me) error?

/B

- -- 
A: Because we read from top to bottom, left to right.
Q: Why should i start my reply below the quoted text?
- -- http://www.i-hate-computers.demon.co.uk/

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)

iD8DBQFA7ZbXWYjaxM2wIe4RAkTNAJ42p2LX5RgCzq4yceIeZOJmJ/QxVgCeIp/g
xkocU0GxrFLl4Vm8F1ai8ys=
=9jaJ
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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

[mattf]

Hello,

I was a reviewer of the book, and it is a good beginner book. At over 260
pages it isn't exactly small and it does contain a lot of information that
is not in the wiki, especially on telco topics and a very in-depth overview
of Cisco IP phones.

It does a good job of going over the basics of Asterisk, which in itself is
a difficult task as some of the basics of Asterisk change rather frequently.


It is a beginner book and doesn't cover some more advanced and more
specialized topics, but it would be a good way for someone who had no
understanding of the telco/VOIP world to get into Asterisk and set up their
first test system.

$50 is a bit steep for a book, but I can tell you that they did put a lot of
effort into creating a comprehensive intro to Asterisk with a great deal of
the material being written by the authors from scratch, and Asterisk is
still quite a niche market relative to other cheaper technology books out
there justifying the cost a bit more.

MATT---


-Original Message-
From: Neil Cherry [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 08, 2004 11:58 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FINALLY! a good book about Asterisk.


Joe Babstock wrote:
> There is finally an introductory book about Asterisk!
> It looks like Paul Mahler at www.signate.com wrote it
> with a lot of help from Digium. I looked at the sample
> pages, it looks great. 

And how do you know it's a good book? I wouldn't mind a
review and I may purchase the book (I doubt I qualify
as a reviewer as I haven't yet figured this VoIP stuff
out yet). I'm not really sure a few pages qualifies for
a review. BTW, please excuse me if Paul is a frequent
contributor to the mail list. I just found the method
of announcement a bit suspect (I'm not say Paul posted
this either).

-- 
Linux Home Automation Neil Cherry[EMAIL PROTECTED]
http://home.comcast.net/~ncherry/   (Text only)
http://linuxha.sourceforge.net/ (SourceForge)
http://hcs.sourceforge.net/ (HCS II)
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[Asterisk-Users] Re: i or s or whatever the invalid_exten is HELP !!!!!

[Stefan Tichy]

You are using SIP phones and in your sip.conf there should be some
context definition for incoming calls. You may jump (goto) to some
other context later, but the context defined in sip.conf should
contain extensions catching all numbers you may dial from that phone.


On Thu, Jul 08, 2004 at 04:09:55PM +0100, Simon wrote:
> Or can i test the first 2 digits of a number and send it to the right
> context ?

Yes, that is what you have to do:

exten => i,1,goto(busy,1,1)
exten => _65.,1,goto(office1,${EXTEN},1)
exten => _66.,1,goto(office2,${EXTEN},1)


-- 
Stefan Tichy   <[EMAIL PROTECTED]>
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Re: [Asterisk-Users] Shady dial anyone??

[C. Maj]

On Thu, 8 Jul 2004, Nauman Farooq waxed:

> wondering if anybody knows this..does shady dial work only with a zap
> interface or can it be configured to be used with SIP or IAX. 

It is interface ambivalent.

--Chris


-- 
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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Re: [Asterisk-Users] Re: tdm400p static - out of ideas (Jorge Mendoza)

[Ryan Courtnage]

On 8-Jul-04, at 10:49 AM, David Cook wrote:
Ryan, from the console what does "zap show channel 1" or 2/3/4 in your
case say.
I have X100P's and I seem to be having similar sounding problems. I
noticed that the above command shows the channel to be off-hook at all
times when a phone line is plugged in.
My is the same way (I've tried on 2 servers with tdm400p cards) - my 
guess is that this is normal.

I don't know why or if it is a bug in the application reporting the 
status.

dbc.
Ryan Courtnage wrote:
On July 8, 2004 03:22 am, Nicholas Bachmann wrote:
Ryan Courtnage wrote:
Hello,
Over the past several weeks, we have been having an intermittant 
problem
with our deployment of a TDM400P card (4 fxo).  We have tried many
things, and the problem still re-occurs.

The Problem:
Occasionally (every 48 hours), the TDM400p card will stop answering
incoming calls on ALL fxo ports.  Attempts to send outbound calls 
on any
Zap channel will result in hearing a loud 'static' noise on the 
line.
Let's look at some possibilities of line problems:
What time does it stop answering? Is it ever during ALIT times 
(usually
very early morning)?

It's totally random - morning/evening/afternoon.  Once it stops 
answering,
that's it, a reboot or module-reload is needed.  If ALIT for some 
reason
prevents the card from answering, it should be able to recover and 
begin
answering after the ALIT is complete.


Have you tried calling the telco to see if it could be their problem?

When the card goes into the non-functional state, I can plug a regular
phone
into any of the lines and make calls just fine.  After verifying 
working
lines and plugging them back into the tdm400p card, I still can't 
dial out
(the Zap channel will answer, but I will hear only static, and the 
call to
the pstn is never placed).  As well, incoming calls will not be
answered (*
console will not even show the 'started simple switch on zap/x' 
message).


How far away from the CO/mux are you?

Not too sure - it's in downtown Calgary - so probably not far.
There is the possibility that _something_ with the phone line is
triggering
the problem.  Maybe it's some noise, an unexpected signal, some
crosstalk ...
things that will cause unexpected behavior ... but also things that
shouldn't
put the entire card into a non-functioning state.

Have you tried a new/different card?  If you didn't want to fork out 
the
cash for a new one, you could try a X100P/knockoff* on one of the 
lines
to see if that fixes the problem... if so you can deduce a bad card.

I may have to push for a replacement tdm400p card from Digium.

Nick
*I usually don't recommend the knockoffs, but for a day of testing 
$10
sure beats $100... everybody else should support Digium! :-)

An acquaintance who is having the same problem has reluctantly
replaced his
card with an openline4.  I would like nothing more than to stick with
Digium
hardware - this thread and obtaining a replacement card is my last 
kick at
the cat.
--
David Cook
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RE: [Asterisk-Users] Fax Detection

[Seth Remington]

Hi Ryan,

Faxdetect should be set to one of the following in zapata.conf:

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

I don't believe that faxdetect=yes is valid but I didn't read the
source, just the sample config file.

-Seth

On Thu, 2004-07-08 at 12:28, Matt wrote:
> Hi Ryan,
> 
> Telappliant only offer SIP/IAX inbound I do have an X100P in the PBX and
> faxdetect=yes in the zapata.conf.
> I suppose my question is: does SpanDSP detect faxes over SIP or is it only
> over Zap?
> 
> Matt
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Courtnage
> Sent: 08 July 2004 09:47
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Fax Detection
> 
> On July 7, 2004 09:19 pm, Matt wrote:
> > Hi all
> >
> > I've tried Google, wiki and mailing list and IRC but still haven't 
> > gotten to the bottom of this.  Hopefully someone might be able to help.
> >
> > I'm using telappliant to provide my inbound and outbound calls. 
> 
> I'm not familiar with teleppliant.  Do you use a digium (zap) card?  AFAIK,
> you need one and need faxdetect=yes in zapata.conf.
> 
> When a fax comes in, does anything relevant get written to * console?
> 
> > * plays
> > host to 30 cisco's and they are all working great using G711 A-law.  
> > I've managed to get SpanDSP to compile and install and I can send a 
> > receive a fax on a dedicated extension.  What I'm trying to do now and 
> > can't seem to nail is getting an inbound fax to be detected and then
> handled.
> >
> > I've tried the examples from the wiki and the sites linked on the 
> > wiki; messed about trying my own weird and wonderful methods but still no
> joy.
> >
> > All the calls are using G711 A-law.
> >
> > Here is the test context I'm using
> >
> > XXX = hiden
> >
> > Exten => 08700686XXX,1,Goto(textextension,7000,1)
> >
> > [testextension]
> > Exten => 7000,1,Answer
> > Exten => 7000,2,Dial(SIP/104)
> > Exten => fax,1,rxfax(/var/spool/asterisk/incoming/testfax.tif)
> > Exten => fax,2,congestion
> > Exten => fax,102,congestion
> >
> > Calls hit the testextension contect but don't get detected as a fax.
> >
> > Cheers
> >
> > Matt
> >
> > ___
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> 
> --
> ..
> Ryan Courtnage
> Coalescent Systems Inc
> 403.244.8089
> www.voxbox.ca
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-- 
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559

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Re: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6

[Zdenek Bouresh]

mattf wrote:
Hello,
I am running Asterisk on Slackware 10.0 with the 2.4 kernel(default kernel)
and it is very happy. Don't see too much difference from 9.1 except for the
fact that most of the binutils have been updated and several of them run
differently now(top, ps, ...)
Haven't tried the 2.6 kernel yet, but may try it later.
MATT---
-Original Message-
From: Joe Baptista [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 08, 2004 10:01 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6

I'm installing the new Slackware 10.0 distribution - but not sure if i
should go with the 2.4 kernal - which i think is the default install - or
the new 2.6 kernal?  anyone running * and slackware 10.0 with 2.6 kernal?
thanks
joe
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Slackware 10 and kernel 2.6.6 and 2.6.7 run fine .
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Re: [Asterisk-Users] x100p and two hfc isdn cards

[Tomaz]

aha ,
yes i try to put x100p out from box and everything is ok :) but once 
again .. put together slowly all 3 cards (2xhfc + 1 x100p)
and get working 2 hfc cards and loading x100p without any errors!

less /proc/zaptel/3
Span 3: WCFXO/0 "Wildcard X101P Board 1" RED
  7 WCFXO/0/0 FXSKS
but:
-- Executing Dial("SIP/101-f09f", "Zap/7/BYEXTENSION|130|t|r") in new stack
Jul  8 22:35:14 NOTICE[311316]: app_dial.c:559 dial_exec: Unable to 
create channel of type 'Zap'
 == Everyone is busy at this time

so anyone has maybe solution  to this  problem ? 
thankyou


[EMAIL PROTECTED] wrote:
hi
I had the saem trouble, so I just took my x100p card out
and the problem went away:)
I know its not the ultimate solution, but I decided to use
an ATA with my analgue phone instead.
I would suggest trying to put the analogue lines as channel
7 and the isdn lines as channels 1-6
Good luck
regards
Clive

On Thu, 08 Jul 2004 11:52:23 +0200
Tomaz <[EMAIL PROTECTED]> wrote:
 

hello,
i have a problem starting asterisk with one x100p digium
and two hfc chipset isdn cards with bri-stuff.0.0.2.
"ztcfg -vv" shows me a this  info:
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves:
02)
Channel 03: Individual Clear channel (Default) (Slaves:
03)
Channel 04: D-channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves:
05)
Channel 06: Individual Clear channel (Default) (Slaves:
06)
Channel 07: D-channel (Default) (Slaves: 07)
7 channels configured.
ZT_SPANCONFIG failed on span 1: Invalid argument (22)
   

-
 

cat  /etc/zaptel.conf
loadzone=nl
defaultzone=nl
fxsks=1

loadzone=nl  defaultzone=nl
span=1,1,3,ccs,ami
bchan=2-3,5-6
dchan=4,7  

and
# cat /etc/asterisk/zapata.conf
[channels]
switchtype = euroisdn
; p2p TE mode
signalling = bri_cpe
;
prilocaldialplan=national
pridialplan = unknown
;
echocancel=yes
group = 1
context=isdn
channel => 2-3,5-6
group = 2
context=gsm
signalling=fxs_ks
channel => 1
-
but when i start asterisk i got this errors:
Parsing '/etc/asterisk/zapata.conf': Found
Jul  8 13:53:58 WARNING[16384]: chan_zap.c:682 zt_open:
Unable to specify channel 2: No such device or address
Jul  8 13:53:58 ERROR[16384]: chan_zap.c:5397 mkintf:
Unable to open channel 2: No such device or address
here = 0, tmp->channel = 2, channel = 2
Jul  8 13:53:58 ERROR[16384]: chan_zap.c:7668 setup_zap:
Unable to register channel '2-3'
Jul  8 13:53:58 WARNING[16384]: loader.c:313
ast_load_resource: chan_zap.so: load_module failed,
returning -1
 == Unregistered channel type 'Tor'
 == Unregistered channel type 'Zap'
   -- Unregistered channel 1
Jul  8 13:53:58 WARNING[16384]: loader.c:408
load_modules: Loading module chan_zap.so failed!
Segmentation fault
what to do? i have latest CVS asterisk ..
thank you,
Tomaz
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_
For super low premiums ,click here http://www.dialdirect.co.za/quote
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[Asterisk-Users] Re: Shady dial anyone??

[Jason Stewart]

On 08/07/04 19:04 +0500, Nauman Farooq wrote:
> wondering if anybody knows this..does shady dial work only with a zap
> interface or can it be configured to be used with SIP or IAX. 
> 
> Nauman

--- Unecessary reply to asterisk-users digest snipped out---

It should work with any type of channel, seeing as the only files modified are
app_queue.c and chan_agent.c

I cannot vouch for this, as I've never used shady dial, but I will be
and I'll be sure to give my review of it in the near future.

Cheers,
Jason

P.S. Please do not reply to an existing message; this wastes bandwidth
and screws up threaded mail user agents such as pine and mutt. just
send a new mail to [EMAIL PROTECTED]
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[Asterisk-Users] Best VOIP PSTN Provider.

[Carlos Arnt]

Hi People,
 
I wondering here, who is the best VOIP PSTN Provider to use with my * box ?
That has good prices, good quality (Use ex. G729 codecs) etc ?
 
I want to make calls to Europe, Asia etc with cheap prices and good quality in the sound ;)
 
Did anyone has a good tip for me ?
 
For use with IAX2 or SIP and with g.729 codec.
 
Thanks alot !
 
 
Carlos.


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[Asterisk-Users] Using Cisco AS5350 as pstn GW .. one-way audio problem

[micke]

Hi all.

I have a strange problem, I've got a AS5350 hooked up to a telco using
two trunked E1's

The 5350 should only act as a GW to a sipproxyserver. 

THe thing is it seems to be only oneway audio?

There are no firewall at all, and the audio still only get one-way

When I call from pstn --> as5350 --> sip-sip-phone   I can here the
sip-phone  ,, but the sipphone cannot her the pstn-phone.

What could be the reason for this ?

/Mike

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[Asterisk-Users] rxfax - mISDN - missing logs

[Stefan Tichy]

Hi,

using HFC cards, mISDN/chan_misdn and spandsp lib fax retrieval works,
but some log file entries are missing. There should be one of the lines:
Fax successfully received.
Fax receive not successful.


Dail Plan config used:

[fax]
exten => _.,1,SetVar(FAXFILE=.)
exten => _.,2,SetVar(LOCALSTATIONID=..)
exten => _.,3,rxfax(${FAXFILE})
exten => _.,4,NoOp,XYZ
exten => _.,5,Hangup


Log file:

   -- Executing RxFAX("mISDN/2", "..") in new stack
   Got hangup
   Extension .., priority . returned normally even though call was hung up
   cdr_mysql: inserting a CDR record.

The file ${FAXFILE} is available and seems to be complete, but isn't
this hangup to early?

Is this a rxfax/spandsp or a mISDN problem?

-- 
Stefan Tichy   <[EMAIL PROTECTED]>
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RE: [Asterisk-Users] Audiocodes -> Asterisk Implementation

[Brian J. Rathman]

Here are the setting off of the box. It is my understanding that anything over version 
4.20 is ok. Any help would be greatly appreciated.

Version ID:4.20.299.4192 
DSP Type:48104 
DSP Software Version:20231 
DSP Software Name:105IM4  
Flash Version:192


-Original Message-
From: Anton Tinchev [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 08, 2004 11:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Audiocodes -> Asterisk Implementation


Brian J. Rathman wrote:

> Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get 
> the channels to registers with Asterisk, but anytime I try and send a call I receive 
> these error messages:
> 
> Jul  6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission 
> on '[EMAIL PROTECTED]' of Response 20587: Found
> Jul  6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:11 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:13 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:18 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:20 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:24 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of 
> order packet 20589 (expecting 20588)
> Jul  6 15:12:25 DEBUG[1133742896]: chan_sip.c:706 __sip_autodestruct: Auto 
> destroying call '[EMAIL PROTECTED]' 
> 
> I am using CVS version Asterisk CVS-HEAD-06/18/04-11:53:43. I have tried changing 
> just about every config option I can think of in both Asterisk and the Audiocodes 
> box without any success. Any ideas? I have checked the web for documentation on this 
> setup, and all I have found is that some people have it working, but that is about 
> it, no details. Any help would be greatly appreciated.
> 
> Thanks,
> Brian
> 
Firmwire version?

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[Asterisk-Users] Variable for extension that transferred the call?

[Paul Zimm]

Is there a variable available to use in my dial plan, that identifies 
the extension or callerid that
transferred the call?

I use valet parking to stack calls for an extension in their own parking 
lot.
I'd like the extension they use for valet park to be the same for all 
phones. That way I can
"program" a memory button on the phone without worrying if it's hooked 
up to the
correct extension.

My extensions are 3 digit and all begin with 8.
current . . . . . . . . . .
exten => _5XX,1,ValetParkCall(auto|8${EXTEN:1}|120|8${EXTEN:1}|1|default)
exten => *66,1,ValetUnParkCall(fifo|${CALLERIDNUM})
what I want . . . . . . . . . .
exten => 
*55,1,ValetParkCall(auto|${extension_that_transferred}|120|${extension_that_transferred}|1|default)
exten => *66,1,ValetUnParkCall(fifo|${CALLERIDNUM})

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Re: [Asterisk-Users] Dialing an extension during Dial()

[lenz]

as an alternative, you can use dial with the T or t option and then have  
the user press # for a blind transfer; this way you can manipulate  
extensions easily.
l.
ps. would surely be nice to have Dial() be able to recognize other  
extensions, just like it now doe4s with #.
l.


In data Wed, 07 Jul 2004 15:51:27 -0700, programmer_ted  
<[EMAIL PROTECTED]> ha scritto:

I don't know if my last message went through, but the gist of it was  
that currently, this support is lacking from the Dial command.  After  
looking at the source for a few minutes, this is something I'd like to  
undertake, and it shouldn't take longer than a night, so I'll be  
hopefully getting back to you with a patch tonight or tomorrow.  Also,  
I'll be submitting the patch to the bugfix list when it's done, so we  
may see this feature in the CVS in the future.

At 08:06 AM 7/7/2004, you wrote:
I need to be able to have callers cancel a Dial() command by pressing 1  
so
that they can leave a message.  I am unable to get this working.  It  
seems
that Dial() will only allow the caller to transfer after the call has  
been
connected or after the Dial() command itself has timed out.  Can someone
please help?

-Ryan

--
Creato con M2, il rivoluzionario client e-mail di Opera:  
http://www.opera.com/m2/
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RE: [Asterisk-Users] odd behavior - adtran ta 850 + t100p

[Bisker, Scott (7805)]

I've never used an 850, but I had similar problem on the 750 when I had the channel 
configured wrong in the 750 console.  Have you tried reseting the config and making 
sure everything is FXS Loopstart.

Also, have you tried another AMP-50 cable with your bank.  I had a bad cable that was 
crossing signal with all channels.

-sb

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeff Roberts
Sent: Thursday, July 08, 2004 11:11 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] odd behavior - adtran ta 850 + t100p


[EMAIL PROTECTED] wrote:

>I've been working with an adtran ta 850 hooked to a t100p pretty much all
>day today, and I haven't gotten past configuring zaptel.conf and
>zapata.conf.  For some reason, when I pick up analog phone hooked up to
>the first module of a quad fxs card in the second slot of the ta 850,
>asterisk thinks that all four of the fxs modules in that card are going
>off hook.  If I pick up a phone hooked to module 2 of the same fxs card
>then asterisk (correctly) only sees that module go off hook.
>
>When plugging a phone into any of the fxs pairs, I only get dial tone for
>a second or two and then I get silence.  However, I can still dial
>extensions and get through.  I'm not sure but maybe it is a config problem
>with the ta 850, as it takes a little more manual configuration than the
>ta 750 I worked with before.  Anybody have any pointers?
>
>Here is the output on the console when I pick up a phone on module 1, and
>module 2, respectively:
>
>[EMAIL PROTECTED]:~# asterisk -r
>Asterisk CVS-HEAD-07/06/04-12:37:58, Copyright (C) 1999-2004 Digium.
>Written by Mark Spencer <[EMAIL PROTECTED]>
>=
>Connected to Asterisk CVS-HEAD-07/06/04-12:37:58 currently running on
>slack1 (pid = 702)
>- Remote UNIX connection
>-- Starting simple switch on 'Zap/5-1'
>-- Starting simple switch on 'Zap/6-1'
>-- Starting simple switch on 'Zap/7-1'
>-- Starting simple switch on 'Zap/8-1'
>-- Hungup 'Zap/5-1'
>-- Hungup 'Zap/6-1'
>-- Hungup 'Zap/7-1'
>-- Hungup 'Zap/8-1'
>-- Starting simple switch on 'Zap/5-1'
>Jul  6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook
>failed: Device or resource busy
>Jul  6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook
>failed: Device or resource busy
>-- Starting simple switch on 'Zap/6-1'
>-- Starting simple switch on 'Zap/7-1'
>Jul  6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook
>failed: Device or resource busy
>-- Starting simple switch on 'Zap/8-1'
>-- Hungup 'Zap/5-1'
>-- Hungup 'Zap/6-1'
>-- Hungup 'Zap/7-1'
>-- Hungup 'Zap/8-1'
>-- Starting simple switch on 'Zap/6-1'
>-- Hungup 'Zap/6-1'
>
>
>Here is zaptel.conf:
>
>span=1,0,0,esf,b8zs
>loadzone = us
>defaultzone=us
>fxsks=1
>fxoks=5-24
>
>And here is zapata.conf:
>
>[channels]
>transfer=yes
>
>context=default
>
>language = en
>usecallerid = no
>hidecallerid = no
>echocancel=yes
>echocancelwhenbridged=yes
>rxgain=0.0
>txgain=0.0
>immediate=no
>signalling=fxs_ks
>echotraining=yes
>
>group = 0
>channel => 1
>
>context=trusted
>group = 1
>
>signalling = fxo_ks
>rxwink = 300
>
>channel => 5-24
>
>Any help appreciated,
>-Jeff
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>  
>
Well I tried setting up the the unused fxo ports, tried setting them to 
unused, and even moved the fxs cards around in the bank to see if it 
would make any difference. No joy though.  Anybody know how to run some 
self tests on this bank to be sure its the problem?  I'm pretty sure 
adtran will fix or replace the bank, but I'm sure they are going to want 
me to explain the problem but I'm not sure what info they'll need.
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Re: [Asterisk-Users] Sip Peer Status

[Karl Brose]

The times are measured as a result of qualify=yes and involve sending an 
OPTION SIP request to the peer which in turn has to reply
to the request.  So this is an application layer "ping" which naturally 
has much greater latency than an ICMP ping.

Brent Franks wrote:
Hello,
I am cruious what exactly status shows.  If I do a sip show peers, I get
this table:
2133/213310.10.60.9   D  255.255.255.255  5060 OK
(95 ms)
2120/212010.10.60.2   D  255.255.255.255  5060 OK
(112 ms)
Now, if I exit asterisk, and ping from the same server, response times are
never greater than 2ms.  Interestingly enough, the one that shows up at a
lower 95 ms is actually on a different switch with higher ping times than
the 2120 peer.
What gives, is this a bug?
Thanks,
- Brent
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RE: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Jeremy Kenney]

This is true but school cost money

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, June 08, 2004 11:16 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

I think I will pass.  $49 for something free on the wiki seems too
expensive.  A cheaper PDF would would save a tree and probably be more
reasonable in cost.


- Original Message - 
From: "Joe Babstock" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]>;
<[EMAIL PROTECTED]>
Sent: Thursday, July 08, 2004 10:05 AM
Subject: [Asterisk-Users] FINALLY! a good book about Asterisk.


> There is finally an introductory book about Asterisk!
> It looks like Paul Mahler at www.signate.com wrote it
> with a lot of help from Digium. I looked at the sample
> pages, it looks great.
>
>
>
> __
> Do you Yahoo!?
> New and Improved Yahoo! Mail - Send 10MB messages!
> http://promotions.yahoo.com/new_mail
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[Asterisk-Users] Re: tdm400p static - out of ideas (Jorge Mendoza)

[David Cook]

Ryan, from the console what does "zap show channel 1" or 2/3/4 in your
case say.

I have X100P's and I seem to be having similar sounding problems. I
noticed that the above command shows the channel to be off-hook at all
times when a phone line is plugged in.

I don't know why or if it is a bug in the application reporting the status.

dbc.

Ryan Courtnage wrote:
> On July 8, 2004 03:22 am, Nicholas Bachmann wrote:
>
>>Ryan Courtnage wrote:
>>
>>>Hello,
>>>
>>>Over the past several weeks, we have been having an intermittant problem
>>>with our deployment of a TDM400P card (4 fxo).  We have tried many
>>>things, and the problem still re-occurs.
>>>
>>>The Problem:
>>>
>>>Occasionally (every 48 hours), the TDM400p card will stop answering
>>>incoming calls on ALL fxo ports.  Attempts to send outbound calls on any
>>>Zap channel will result in hearing a loud 'static' noise on the line.
>>
>>Let's look at some possibilities of line problems:
>>What time does it stop answering? Is it ever during ALIT times (usually
>>very early morning)?
>
>
> It's totally random - morning/evening/afternoon.  Once it stops answering,
> that's it, a reboot or module-reload is needed.  If ALIT for some reason
> prevents the card from answering, it should be able to recover and begin
> answering after the ALIT is complete.
>
>
>>Have you tried calling the telco to see if it could be their problem?
>
>
> When the card goes into the non-functional state, I can plug a regular
phone
> into any of the lines and make calls just fine.  After verifying working
> lines and plugging them back into the tdm400p card, I still can't dial out
> (the Zap channel will answer, but I will hear only static, and the call to
> the pstn is never placed).  As well, incoming calls will not be
answered (*
> console will not even show the 'started simple switch on zap/x' message).
>
>
>>How far away from the CO/mux are you?
>
>
> Not too sure - it's in downtown Calgary - so probably not far.
>
> There is the possibility that _something_ with the phone line is
triggering
> the problem.  Maybe it's some noise, an unexpected signal, some
crosstalk ...  
> things that will cause unexpected behavior ... but also things that
shouldn't
> put the entire card into a non-functioning state.
>
>
>>Have you tried a new/different card?  If you didn't want to fork out the
>>cash for a new one, you could try a X100P/knockoff* on one of the lines
>>to see if that fixes the problem... if so you can deduce a bad card.
>
>
> I may have to push for a replacement tdm400p card from Digium.
>
>
>>Nick
>>
>>*I usually don't recommend the knockoffs, but for a day of testing $10
>>sure beats $100... everybody else should support Digium! :-)
>
>
> An acquaintance who is having the same problem has reluctantly
replaced his
> card with an openline4.  I would like nothing more than to stick with
Digium
> hardware - this thread and obtaining a replacement card is my last kick at
> the cat.

-- 
David Cook
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[Asterisk-Users] Interface to generate Statements?

[Darren]

Hello,

Is there any downloadable software to generate Statements from the mysql
call log?

I'm considering developing a PHP/MySql interface but I'm thinking that
someone must have already done this?

Thanks,

- Darren

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Re: [Asterisk-Users] Small Linux Distro

[Philipp von Klitzing]

Hi!

> Does anyone have a current, stripped linux distro which has only 
> asterisk and net drivers?

Look here:
http://www.voip-info.org/wiki-Asterisk+installation+tips

and you'll find a link to the "Asterisk Live! CD-ROM".

If you have a moment I guess the list (and certainly me) would be 
interested to hear about your experiences with this. :-)

Cheers, Philipp

P.S.: You might also want to look at this:
http://www.voip-info.org/wiki-Asterisk+linux+distributions


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