Re: [Asterisk-Users] IAX2 bug in DIAX solved - Great Thanks to Steven!

[Dan]

Hi Andrew,

- Original Message - 
From: "Andrew Thompson" <[EMAIL PROTECTED]>
> Dan,
> I was trying to find out if this patch was for the problem I described
> several weeks ago:
>

No, this is only for the bug related with the following in IAX2: The phone
does not ring anymore after around 1min. and need to be restarted in order
to further accept calls.


Best regards,
Dan


___
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

Re: [Asterisk-Users] Mailing List Lag

[Chris Albertson]

Maybe the words "nine million" was not ment to be taken
literally.  What if he said "about a gazillion"  Then we'd
all be arguing if "gazillion" == 1x10^14 or 1x10^16

Have you ever set up "mailman" on a Linux system?  9,000,000
would be a real trick setup not something you'd do with a
standard PC.  You have a rack full of mail servers, raid disks
and the works.  Lets do some math:

 There are 60*60*24 seconds in a day 9E6 per messages/day
 means a little over 100 messages per second sustained 24x7.
 I really doubt it.

I think "nine milion" is a technicl term just like "sh*t load"
or "way lots".
  
--- Philipp von Klitzing <[EMAIL PROTECTED]>
wrote:
> Hi!
> 
> > 900 mails / 150 per day = 6 Subscribers..
> > Is the Asterisk community anywhere near that big??
> 
> There is more than just one list on that MLM, and you'll probably
> have to 
> count in adminstrative messages (?) and maybe even error messages.
> 
> Still that would reveal a huge number of subscribers...
> 
> Cheers, Philipp
> 
> 
> ___
> 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


=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
___
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

Re: [Asterisk-Users] Polycom Reboot Script - Please wiki-size me

[John Baker]

Oops, the cut and paste got the better of me.

This one might be a little better...

#!/usr/bin/perl
#
# PolyReboot.pl
#
# Reboots a Polycom 500 or 600 phone
#
use Net::Ping;
use Socket;

$polycompath = '/home/poly/';# Where you keep your config files
$arp = '/sbin/arp';  # Location of arp command
$sipserver   = '192.168.XXX.XXX';  # IP of asterisk server

$phone = shift;

checkphone("$phone");
touch( arp2config("$phone") );

reboot_sip_phone( "$phone", "$sipserver", "Reboot" );

sub checkphone {  # Checks for existence of phone, makes sure
 # it's in arp table
 $activephone = shift;

 # ARP table needs our phone
 print "Checking ARP table.\n";
 $p = Net::Ping->new("icmp");
 if ( $p->ping( $activephone, 2 ) ) {
 print "$activephone is ";
 print "reachable.\n";
 }
 else { die "Polycom at ", $activephone, " is not reachable!"; }
 sleep(1);
 $p->close();

 }

 sub arp2config {# Gets mac address from arp table, converts
 # to a polycom config filename, makes sure
 # the config file exists
 $arpip = shift;
 open( ARP, "$arp -an|" ) || die "Couldn't open arp table: $!\n";
 print "checking for polycom config name...", "\n";
 while () {
 chomp;
 $addr = $_;
 $ip   = $_;
 $addr =~ s/.*
([\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+).*/$1/;
 $addr =~ s/://g;
 $addr = lc($addr) . '.cfg';
 $ip =~ s/.*?(\d+\.\d+\.\d+\.\d+).*/$1/;
 if ( $ip eq $arpip ) {
 last;
 }
 }
  $polycomconfig = "$polycompath" . "$addr";

 unless ( -e "$polycomconfig" ) {
 print "sorry, polycom config file ", "$polycomconfig",
   " is not found.\n\n";
 exit;
 }

 return $polycomconfig;
 }


 sub touch {# We need to touch the config file or the phone
# won't reboot - it depends on time synchronization
 print "touching config file ", $polycomconfig, "\n";
 my $now = time;
 local (*TMP);
 foreach my $file (@_) {
 utime( $now, $now, $file )
   || open( TMP, ">>$file" )
   || die ("$0: Couldn't touch file: $!\n");
 }
 }

 sub reboot_sip_phone {# Send the phone a check-sync to reboot it
 $phone_ip = shift;

 $local_ip = shift;
 $sip_to   = shift;
 $sip_from = "0";
 $tm   = time();
 $call_id  = $tm . "msgto$sip_to";
 $httptime = `date -R`;
 $MESG = "NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
 Via: SIP/2.0/UDP $local_ip
 From: 
 To: 
 Event: check-sync
 Date: $httptime
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1300 NOTIFY
 Contact: 
 Content-Length: 0

 ";

 $proto = getprotobyname('udp');
 socket( SOCKET, PF_INET, SOCK_DGRAM, $proto );
 $iaddr = inet_aton("$phone_ip");
 $paddr = sockaddr_in( 5060, $iaddr );
 bind( SOCKET, $paddr );
 $port = 5060;

 $hisiaddr = inet_aton($phone_ip);
 $hispaddr = sockaddr_in( $port, $hisiaddr );

 if ( send( SOCKET, $MESG, 0, $hispaddr ) ) {
 print "reboot of phone ", "$phone_ip", " was successful", "\n";
 }
 else { print "reboot of phone ", "$phone_ip", " failed", "\n"; }
 }
 exit;

- Original Message - 
From: "John Baker" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 11:38 PM
Subject: [Asterisk-Users] Polycom Reboot Script - Please wiki-size me


> With my thanks to Brian West and his offering in the thread,
>
> "Subject: Re: [Asterisk-Users] Remote reload Cisco 7960"
>
> I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones
>
> PolyReboot.pl takes an IP address as a single argument and reboots the
> phone.
>
> You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg -
all
> lower case.  Further,
> you need to use ftp for your remote cfg loading, because with ftp
> synchronization, the phones
> check the ctimes of the cfg files against what's in their cache when
> receiving a check-sync in
> order to determine if rebooting is necessary.
>
> I tested this script.  It works on my Polycoms.  I have the latest bootrom
> and the
> SIP 1.1.0 software version.
>
> I'm not much of a perl person, and all of this code is heavily borrowed,
> (Thanks again, Brian), so
> someone else can prettify it as they see fit.
>
> #!/usr/bin/perl
> #
> # PolyReboot.pl
> #
> # Reboots a Polycom 500 or 600 phone
> #
> use Net::Ping;
> use Socket;
>
> $polycompath = '/home/poly/';# Where you keep your config files
> $arp = '/sbin/arp';  # Location of arp command
> $sipserver   = '192.168.XXX.XXX';  # IP of asterisk server
>
> $phone = shift;
>
> checkphone("$phone");
> touch( arp2config("$phone") );
>
> reboot_sip_phone( "$phone", "$sipserver", "Reboot" );
>
> sub checkphone {  # Checks for existence of phone, makes sure
> # it's in arp table
> $activephone = shift;
>
> # ARP table needs 

[Asterisk-Users] Polycom Reboot Script - Please wiki-size me

[John Baker]

With my thanks to Brian West and his offering in the thread,

"Subject: Re: [Asterisk-Users] Remote reload Cisco 7960"

I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones

PolyReboot.pl takes an IP address as a single argument and reboots the
phone.

You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all
lower case.  Further,
you need to use ftp for your remote cfg loading, because with ftp
synchronization, the phones
check the ctimes of the cfg files against what's in their cache when
receiving a check-sync in
order to determine if rebooting is necessary.

I tested this script.  It works on my Polycoms.  I have the latest bootrom
and the
SIP 1.1.0 software version.

I'm not much of a perl person, and all of this code is heavily borrowed,
(Thanks again, Brian), so
someone else can prettify it as they see fit.

#!/usr/bin/perl
#
# PolyReboot.pl
#
# Reboots a Polycom 500 or 600 phone
#
use Net::Ping;
use Socket;

$polycompath = '/home/poly/';# Where you keep your config files
$arp = '/sbin/arp';  # Location of arp command
$sipserver   = '192.168.XXX.XXX';  # IP of asterisk server

$phone = shift;

checkphone("$phone");
touch( arp2config("$phone") );

reboot_sip_phone( "$phone", "$sipserver", "Reboot" );

sub checkphone {  # Checks for existence of phone, makes sure
# it's in arp table
$activephone = shift;

# ARP table needs our phone
print "Checking ARP table.\n";
$p = Net::Ping->new("icmp");
if ( $p->ping( $activephone, 2 ) ) {
print "$activephone is ";
print "reachable.\n";
}
else { die "Polycom at ", $activephone, " is not reachable!"; }
sleep(1);
$p->close();

}

sub arp2config {# Gets mac address from arp table, converts
# to a polycom config filename, makes sure
# the config file exists
$arpip = shift;
open( ARP, "$arp -an|" ) || die "Couldn't open arp table: $!\n";
print "checking for polycom config name...", "\n";
while () {
chomp;
$addr = $_;
$ip   = $_;
$addr =~ s/.*
([\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+).*/$1/;
$addr =~ s/://g;
$addr = lc($addr) . '.cfg';
$ip =~ s/.*?(\d+\.\d+\.\d+\.\d+).*/$1/;
if ( $ip eq $arpip ) {
last;
}
}
 $polycomconfig = "$polycompath" . "$addr";

unless ( -e "$polycomconfig" ) {
print "sorry, polycom config file ", "$polycomconfig",
  " is not found.\n\n";
exit;
}

return $polycomconfig;
}

}

$polycomconfig = "$polycompath" . "$addr";

unless ( -e "$polycomconfig" ) {
print "sorry, polycom config file ", "$polycomconfig",
  " is not found.\n\n";
exit;
}

return $polycomconfig;
}

sub touch {# We need to touch the config file or the phone
   # won't reboot - it depends on time synchronization
print "touching config file ", $polycomconfig, "\n";
my $now = time;
local (*TMP);
foreach my $file (@_) {
utime( $now, $now, $file )
  || open( TMP, ">>$file" )
  || die ("$0: Couldn't touch file: $!\n");
}
}

sub reboot_sip_phone {# Send the phone a check-sync to reboot it
$phone_ip = shift;

$local_ip = shift;
$sip_to   = shift;
$sip_from = "0";
$tm   = time();
$call_id  = $tm . "msgto$sip_to";
$httptime = `date -R`;
$MESG = "NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP $local_ip
From: 
To: 
Event: check-sync
Date: $httptime
Call-ID: [EMAIL PROTECTED]
CSeq: 1300 NOTIFY
Contact: 
Content-Length: 0

";

$proto = getprotobyname('udp');
socket( SOCKET, PF_INET, SOCK_DGRAM, $proto );
$iaddr = inet_aton("$phone_ip");
$paddr = sockaddr_in( 5060, $iaddr );
bind( SOCKET, $paddr );
$port = 5060;

$hisiaddr = inet_aton($phone_ip);
$hispaddr = sockaddr_in( $port, $hisiaddr );

if ( send( SOCKET, $MESG, 0, $hispaddr ) ) {
print "reboot of phone ", "$phone_ip", " was successful", "\n";
}
else { print "reboot of phone ", "$phone_ip", " failed", "\n"; }
}
exit;

Enjoy,

John

___
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

Re: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)

[Siggi Langauf]

On Thu, 22 Jan 2004, Jeff Gustafson wrote:

> > Sure: Remove the SIPDefault.cnf.xml file from your TFTP server!
> > It's not required, and unless you understand what it's good for, it will
> > keep your phone rebooting in an infinite loop.
>
>   Well, it's XMLDefault.cnf.xml on my box.  It doesn't have the SIP
> firmware load.

Sorry, XMLDefault.cnf.xml is what I meant to say, of course...

> > If your phone has some stale CallManager settings that are preventing you
> > from using it, just follow these steps:
> >
> > 1) make sure you have a DHCP server on your network that will assign the
> >phone an IP address, network mask, DNS server and default router
> >It should also assign the address off your _Asterisk_ server as
> >TFTP server (DHCP: "next-server")
>
>   That kinda helped.  Now it's pointing right at Asterisk.  The problem
> is still that I have better luck with the old chan_skinny code over the
> chan_sccp.  chan_sccp still puts the phone in a config verification
> error loop.

Did you remove the XMLDefault.cnf.xml?

If so, there's still a load version conflict (although I've never seen a
7960 or 7940 care about the version communicated through SCCP):

On the phone, press "Settings", then 4 for load information.
watch out for the "App-Load-ID". On my 7940, this is "P00305000300". Yours
is most likely a smaller number...

If you have a CVS version of chan_sccp (either Jan's CVS or Theo's), just
add this exact string to your /etc/asterisk/sccp.conf in the device
definition for your phone, ie. something like:

[SEP00010203040506]
imgversion = P00305000300
type = 7940
context = default
autologin = yourline

If you still have the 0.2 release of chan_sccp, I'd recomment you to
upgrade, although you could change the value in sccp_helper.c, line 66 and
recompile.

If you have any further trouble, make sure you mention the App-Load-Id,
because it seems to be an unusual one ;-)

Cheers,
Siggi
___
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

Re: [Asterisk-Users] simple question...

[Steven Critchfield]

On Thu, 2004-01-22 at 21:55, Jess Magnaye wrote:
> it just came to my mind, and i haven't done any researches yet if
> somebody tried this one with asterisk.. :) well just in case somebody
> or someone on the list aware, i appreciate any advise.
>  
> in telco world, there's like 64kbps per channel and voice can be
> carried on a 16kbps channel.  is it possible to configure asterisk to
> make 4 extensions (ATAs example), to call out using single FXO port at
> the same time?  if that is possible, then is it also possible to make
> t1-pri to be capable of transmitting 4x23ch simultaneous calls..?

Your problem is needing to have the same software on both sides, and
your 4 into 1 would have to be going to the same destination as the
telco route 64k chunks. 

Of course the question then is if you have the same software at each
end, why are you worring about PRI when you can go full IP and not get
burned on the D channel. 

-- 
Steven Critchfield <[EMAIL PROTECTED]>

___
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

[Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway

[Jeremy Jones]

Hello folks,

I'm trying to get an 8x8 DTA-310 running mgcp to work.  I get no
dialtone & can't get it to ring.  My mgcp.conf says:

;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 0.0.0.0

[172.16.2.25]
host = 172.16.2.25
context = default
line => aaln/1

And here's the interesting bits of extensions.conf:

[globals]
...
TRUNK=H323/[EMAIL PROTECTED]
...

[default]
exten => 2084728803,1,Dial(MGCP/aaln/[EMAIL PROTECTED])

And, finally, the h323.conf:


; The NuFone Network's
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
tos=lowdelay
allow=all   ; turns on all installed codecs
disallow=g723.1; Hm...  Proprietary, don't use it...
allow=gsm   ; Always allow GSM, it's cool :)
gatekeeper = 10.0.0.202
AllowGKRouted = yes
context=default

[2084728803]
type=h323
e164=2084728803
context=default

Now, including the demo context in default w/out the mgcp extension
worked just dandy & impressed everyone.  I know my box is registering
with the gatekeeper just fine & the as5300 is getting calles to
2084728803 number to the asterisk box.

First: in my [globals] section, is that the right way to use the h323
channel as a trunk?  I couldn't find much relating to this out there.

Second, what's up with my 8x8 gear?  Anyone got it to work?  I can
attach my "mgcp debug" output if someone likes.  Here's a briefer bit
from "asterisk -c" when picking up the handset on the phone
connected to the dta-310:

*CLI> Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue:
Removing messa
ge from aaln/[EMAIL PROTECTED] tansaction 1
-- Resetting interface aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 2
Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 3
Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 4
-- Resetting interface aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 5
Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 6
Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 7
Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 8
-- Resetting interface aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 9
Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 10
Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 11
Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 12
-- Resetting interface aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 13
Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 14
Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 15
Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 16
-- Resetting interface aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
Jan 22 22:07:43 WARNING[213006]: chan_mgcp.c:2126 handle_hd_hf: Off
hook, but al
reaedy have owner on aaln/[EMAIL PROTECTED]
Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 17
Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 18
Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 19
Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing
message fro
m aaln/[EMAIL PROTECTED] tansaction 20
Jan 22 22:07:43 NOTICE[213006]: chan_m

Re: [Asterisk-Users] Background Noise

[Jess Magnaye]

yes, power is also source of background noise, but not all the time. you
must also have proper grounding.

with pc-phone, it could also be related to your phone/speaker gain controls.
so u have to be definite on what kind of background noise u are getting.

best codec for sip? it's always best to have alaw or ulaw. :) but
bw-consuming. alaw is if u are using E1s. ulaw for T1s.  don't even bother
to use gsm. better get g729.  some says u can use g723.1 with * w/o add'l
purchase of license or something.. but i couldn't make this codec work. (so
i went for g729.. :) ) im testing its quality though as digium cards are
using CPUs as DSPs.




- Original Message - 
From: "Jonathan Biggs" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 11:01 AM
Subject: Re: [Asterisk-Users] Background Noise


> Just to add some info from recent experience. May
> help,
> May not   1 X100P  2 X 4 port TDM400P
>
> Had to hook a dial-in palm PDA base for a custom
> software implementation to my * system and have the
> modem dial out and work properly.  Phone connects to
> second port on PDA Base
>
> Experienced very bad "electrical" type noise on the
> line, hum, buzz, fad in and out. would come and go.
>
> Switched ports, wires,  rxgain and txgain changes,
> phone changes, nothing helped
>
> the PDA base is also connected via serial port (or USB
> did not matter) to desktop computer for sync purposes
>
> Through trial and error.  Found noise coming from
> connection to desk top computer. On, Off did not
> matter.
>
> Resolution.  The power strip surge protector I was
> using on the desktop computer has two modes for noise
> filtration built in.  75 Hertz (or something, don;t
> remember) and 50 Hertz.  I moved the plug for the
> desktop from the 75 side to the 50 side,  All noise on
> line now gone.
>
> Not sure if this helps, constant noise on all sides
> may be power and noise filtration related...
>
>
>
>
>
> --- "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
> wrote:
> > Hi All
> > i have some background noise problem with * and a
> > diva srv 4bri + chan_capi 0.3.0 + X-Ten PRO on my
> > pc.
> > Both in incoming and outgoing call have a background
> > noise.
> >
> > there is some tuning to do?
> > where can i find documentation about capi.conf?
> > which is the best codec for sip (ulaw, alaw,
> > gsm...)?
> >
> > mark
> >
> > ___
> > 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
>
>
> __
> Do you Yahoo!?
> Yahoo! SiteBuilder - Free web site building tool. Try it!
> http://webhosting.yahoo.com/ps/sb/
> ___
> 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

___
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

Re: [Asterisk-Users] Variable to play all gsm files in a directory?

[JC]

- Original Message - 
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 9:09 PM
Subject: Re: [Asterisk-Users] Variable to play all gsm files in a directory?


> On Thursday 22 January 2004 16:43, jaycard wrote:
> > Is there a variable that can be used with the playback command to
> > play all gsm sound files in a directory? For example like '*.*' <-
> > which I know doesn't work. Is the only solution for this to create an
> > AGI script and use the stream_file command ?
>
> I can't see where this would be useful in more than a single situation,
> so I'd recommend that you write an AGI script which can use the opendir,
> readdir, and closedir system calls.
>
> -Tilghman
>
> ___
> 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
>

Thank you I guess AGI then is the answer to my question.
I need the technology to work for 2 projects I'm working on. I'll look in to
it. Plus I need it to be able play PREV and NEXT sound files by hiting a
digit(s), like a voice mail system in a way.

Thanks for the response.

J.C.


___
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

Re: [Asterisk-Users] Snom 200 phones not working.

[Ulexus]

I am using 2.03o with a Snom200 with two lines registered without a problem:

However, I am not using any authorization, whih could be where your problem 
is.

from my sip.conf:

[5117]
type=friend
host=dynamic
context=sip-gb
mailbox=5117
callerid="Name <5117>"


On Thursday, 22 January, 2004 16:04, Ariel Batista wrote:
> I have 2 Snom 200 and would like to get them to work properly with
> Asterisk.  With the Firmware 2.02t I am able to use the phone.  But only
> one line configured.  With there newer firmware 2.03o it will not allow
> me to make calls.  But I can get calls on the unit.  Again the 2nd line
> is not able to be registered.  Is this an issue with Asterisk or Snom?
>
> I could use some example configuration files.  I have followed the Snom
> FAQ step by step.  But it's still not working.
> -
> \
> \\_ Ariel Batista
> //
> / Red-Fone Communications, Inc.
> --
> [EMAIL PROTECTED]
> Ph: 786-544-1114
> Fx: 305-574-0212
>
> ___
> 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

___
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

[Asterisk-Users] simple question...

[Jess Magnaye]

it just came to my mind, and i haven't done any 
researches yet if somebody tried this one with asterisk.. :) well just in case 
somebody or someone on the list aware, i appreciate any advise.
 
in telco world, there's like 64kbps per channel and 
voice can be carried on a 16kbps channel.  is it possible to configure 
asterisk to make 4 extensions (ATAs example), to call out using single FXO port 
at the same time?  if that is possible, then is it also possible to make 
t1-pri to be capable of transmitting 4x23ch simultaneous calls..?
 
just curious.. nothing very serious.
 
thanks :)

Re: [Asterisk-Users] Asterisk vs. Websphere Voice Response?

[Steven Critchfield]

On Thu, 2004-01-22 at 21:32, Ted.Thomas wrote:
> 
> 
> (Embedded image moved to file: pic18467.gif)
> 
> I am a collaboration consultant doing some research for a major client who
> has a problem.  They have an existing IBM Direct Talk 2 IVR which they were
> going to upgrade to the current Websphere product, but choked on the price.
> My task is to find out the following:
> 
> 1. The spec calls for a 24 analog line system with a fairly sophisticated
> response matrix using SQL into Oracle & text-to-speech (among other
> things).  Is Asterisk in the same class of product as Websphere, or is it
> for a more straightforward "voicemail" office environment?

I don't know what all websphere does, so I can't comment on it
directly.  24 lines, no problem, but don't do it analog. SQL/oracle also
no problem. Text to speach This is where an acceptable quality is
the problem. Festival is freely available, but requires a long time and
a lot of work to make it sound good. So the answer is not out of the
box, but it can be made into soo much more.

> 2. I assume Asterisk runs on PC class servers using PCI cards for incoming
> lines.  Can the hardware side of a 24 line system be put together for less
> than $10K, or is it typically a lot more than that?

T1 card 1 port(24 lines) $500, T1 card 4 ports(96 lines) $1500. So yes.

> 3. Can Asterisk be administered and/or programmed remotely?

Do you understand how to work on a linux machine? SSH, X tunneling, VNC,
whatever, of course it can.

> 4. Is Asterisk being used by any Fortune 500 companies in mission critical
> settings that you know of?

Are you a sheep? If so, go back to choking on the cost of the IBM
product. If you are ready to do your own work, come on in. Just be aware
that while you will definately save in hardware costs, you may incur a
fair amount of cost in hours learning and building or else you hire
someone to do it for you.

> Sorry for the general nature of the questions; although I use and promote
> open source products such as Linux, I am not that familiar with IVR
> servers.  Thanks for any help you can give me...

Question 3 makes me wonder about how much you know.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

___
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

Re: [Asterisk-Users] asterisk 0.7.1 - mysql

[Tilghman Lesher]

On Thursday 22 January 2004 20:05, Warren H. Prince wrote:
> > I am glad to see I am not the only one who likes using MySQL.. :-)
> >
> > I was begining to feel that I was alone in the Asterisk/MySQL
> > world.. ;-)
>
> I'm somewhat new to * and haven't encountered the MySql connectivity.
> We use MySql and I was wondering what is achieved by installing this
> addon?

Call Detail Records to a mysql database.

-Tilghman

___
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

Re: [Asterisk-Users] Variable to play all gsm files in a directory?

[Steven Critchfield]

On Thu, 2004-01-22 at 20:09, Tilghman Lesher wrote:
> On Thursday 22 January 2004 16:43, jaycard wrote:
> > Is there a variable that can be used with the playback command to
> > play all gsm sound files in a directory? For example like '*.*' <-
> > which I know doesn't work. Is the only solution for this to create an
> > AGI script and use the stream_file command ?
> 
> I can't see where this would be useful in more than a single situation,
> so I'd recommend that you write an AGI script which can use the opendir,
> readdir, and closedir system calls.

ewww.

map{s/\.gsm$//;
print "stream file $_\n"; 
my $junk = <>;
} glob "$path/*.gsm";

-- 
Steven Critchfield <[EMAIL PROTECTED]>

___
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

Re: [Asterisk-Users] IAX2 bug in DIAX solved - Great Thanks to Steven!

[Andrew Thompson]

- Original Message -
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, January 20, 2004 2:31 AM
Subject: Re: [Asterisk-Users] IAX2 bug in DIAX solved - Great Thanks to
Steven!


> Do you mean the call is disconnected after 20-30s?
> Can you provide more details?
>
> Thank you and best regards,
> Dan
>

Dan,
I was trying to find out if this patch was for the problem I described
several weeks ago:

- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, December 10, 2003 9:28 AM
Subject: Re: [Asterisk-Users] DIAX to DIAX call and disconnecting after
50-60 sec.


> - Original Message -
> From: "Dan" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, December 08, 2003 9:14 AM
> Subject: Re: [Asterisk-Users] DIAX to DIAX call and disconnecting after
> 50-60 sec.
>
>
> > Hi,
> >
> > - Original Message -
> > From: "Andrew Thompson" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Monday, December 08, 2003 3:30 PM
> > Subject: Re: [Asterisk-Users] DIAX to DIAX call and disconnecting after
> > 50-60 sec.
> >
> >
> > > - Original Message -
> > > From: "Dan" <[EMAIL PROTECTED]>
> > > To: "Asterisk Users" <[EMAIL PROTECTED]>
> > > Sent: Monday, December 08, 2003 3:49 AM
> > > Subject: [Asterisk-Users] DIAX to DIAX call and disconnecting after
> 50-60
> > > sec.
> > >
> > >
> > > > Hi,
> > > >
> > > > There is any other user of DIAX with this problem?
> > > >
> > > > Thanks,
> > > > Dan
> > >
> > > Yes, my calls that are DIAX to asterisk to DIAX disconnect after about
a
> > > minute. These are Extension to Extension calls.
> > >
> > > HOWEVER, I tested with the IAXComm/IAXClient and had the same results.
> So,
> > I
> > > don't believe it's in your app, I think it's something in the library.
> > Can you check with IAX(1) mode too?
> > I want to know if this is related to LIBIAX2 only..
> > Can you tell me the CVS version used for this test?
> >
>
> Dan,
> Sorry it's taken so long, but I just ran these two test calls...
>
> Version info from Asterisk:
> Asterisk CVS-12/04/03-22:04:29, Copyright (C) 1999-2001 Linux Support
> Services, Inc.
> Written by Mark Spencer <[EMAIL PROTECTED]>
> =
> Connected to Asterisk CVS-12/04/03-22:04:29 currently running on bebop
(pid
> = 5145)
>
> DIAX extension 714 is 0.9.6b on Win2k Dell desktop
> DIAX extension 715 is 0.9.6 on Win2k Gateway laptop
>
> For fun, (and an attempt at re-aquanting myself) I'm running both of them
in
> Spanish.
>
> This is from the first call, IAX to IAX via Asterisk:
>
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:4569
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Accepting AUTHENTICATED call from 24.199.177.150, requested format
=
> 2, a
> ctual format = 2
> -- Executing Macro("[EMAIL PROTECTED]/1602", "ext|715|IAX/715") in new
stack
> -- Executing Dial("[EMAIL PROTECTED]/1602", "IAX/715|90|tT") in new stack
> -- Calling using options 'exten=s;callerid="Andrew Thompson"
> <714>;language=
> en;formats=2;capability=2;version=1;adsicpe=0'
> -- Called 715
> -- Call accepted by 192.168.3.100 (format GSM)
> -- Format for call is GSM
> -- IAX[715]/1603 is ringing
> -- IAX[715]/1603 answered [EMAIL PROTECTED]/1602
> -- Attempting native bridge of [EMAIL PROTECTED]/1602 and IAX[715]/1603
> -- Channel 'IAX[715]/1603' unable to transfer
> -- Channel '[EMAIL PROTECTED]/1602' unable to transfer
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036
> -- Hungup 'IAX[715]/1603'
>   == Spawn extension (macro-ext, s, 1) exited non-zero on
> '[EMAIL PROTECTED]/1602' in macro 'ext'
>   == Spawn extension (trusted, 715, 1) exited non-zero on
> '[EMAIL PROTECTED]/1602'
> -- Hungup '[EMAIL PROTECTED]/1602'
> -- Registered '715' (AUTHENTICATED) at 192.168.3.100:4569
> -- Registered '714' (AUTHENTICATED) at 24.199.177.150:4569
>
> This call lasted about 10 minutes before I hung it up.
>
>
>
>
> Th

[Asterisk-Users] voiceglo.com and dtmf

[Jeff Gustafson]

Hello all,
I've been trying to get a simple PBX up and running with
asterisk.  I decided to sign up with Voiceglo so I could have a PSTN
gateway.  The problem is that I can't seem to get Asterisk to handle
dtmf decoding reliably.  I tried inband and the rfc decoding.  inband
tried to work and the rfc mode didn't do anything.  By try to work I
mean that it rarely properly decoded the number correctly.
I saw a message a while ago that said that Voiceglo supported
IAX.  Is this still true?  I tried IAX but I was getting rejected.
SIP/voiceglo/asterisk seem to be working minus the DTMF.
Any ideas?

...Jeff

___
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

[Asterisk-Users] Asterisk vs. Websphere Voice Response?

[Ted.Thomas]

(Embedded image moved to file: pic18467.gif)

I am a collaboration consultant doing some research for a major client who
has a problem.  They have an existing IBM Direct Talk 2 IVR which they were
going to upgrade to the current Websphere product, but choked on the price.
My task is to find out the following:

1. The spec calls for a 24 analog line system with a fairly sophisticated
response matrix using SQL into Oracle & text-to-speech (among other
things).  Is Asterisk in the same class of product as Websphere, or is it
for a more straightforward "voicemail" office environment?
2. I assume Asterisk runs on PC class servers using PCI cards for incoming
lines.  Can the hardware side of a 24 line system be put together for less
than $10K, or is it typically a lot more than that?
3. Can Asterisk be administered and/or programmed remotely?
4. Is Asterisk being used by any Fortune 500 companies in mission critical
settings that you know of?

Sorry for the general nature of the questions; although I use and promote
open source products such as Linux, I am not that familiar with IVR
servers.  Thanks for any help you can give me...

Sincerely,

E.A. (Ted) Thomas, CEO
eThomasGroup.biz
"Collaboration for Business"
502-802-6130<>

Re: [Asterisk-Users] Mailing List Lag

[Philipp von Klitzing]

Hi!

> 900 mails / 150 per day = 6 Subscribers..
> Is the Asterisk community anywhere near that big??

There is more than just one list on that MLM, and you'll probably have to 
count in adminstrative messages (?) and maybe even error messages.

Still that would reveal a huge number of subscribers...

Cheers, Philipp


___
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

Re: [Asterisk-Users] Variable to play all gsm files in a directory?

[Tilghman Lesher]

On Thursday 22 January 2004 16:43, jaycard wrote:
> Is there a variable that can be used with the playback command to
> play all gsm sound files in a directory? For example like '*.*' <-
> which I know doesn't work. Is the only solution for this to create an
> AGI script and use the stream_file command ?

I can't see where this would be useful in more than a single situation,
so I'd recommend that you write an AGI script which can use the opendir,
readdir, and closedir system calls.

-Tilghman

___
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

Re: [Asterisk-Users] asterisk 0.7.1 - mysql

[Warren H. Prince]

Tilghman,

I am glad to see I am not the only one who likes using MySQL.. :-)

I was begining to feel that I was alone in the Asterisk/MySQL world.. ;-)

PS I am only kidding...

Later..

___
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
I'm somewhat new to * and haven't encountered the MySql connectivity. 
We use MySql and I was wondering what is achieved by installing this addon?
___
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

Re: [Asterisk-Users] Is there a way to # of agents logged into a queue ?

[William Suffill]

I'd be interested in the patch as well
On Thu, 2004-01-22 at 13:51, Bill Hamel wrote:
> Hi Chris,
> 
> This sounds what I am looking for, many thanks !
> 
> Also, I do not see an attachment, the patch that is :)
> 
> I dont know if the list strips attachments, perhaps send it to my email address
> [EMAIL PROTECTED]
> 
> Thanks again,
> -bh
> 
> 
> Quoting "C. Maj" <[EMAIL PROTECTED]>:
> > I attached a patch I've been using to show the # of agents
> > (members) and callers on a per queue basis.  It adds a new
> > manager command, "AgentQueues".  It returns on the manager
> > interface the following for each queue:
> > 
> > Queue: queuename
> > Agents: #
> > Callers: #
> > 
> > There's another manager command, "QueueStatus", that might be
> > what your are looking for.  There's also "Queues" but that
> > is a PITA to parse.  Fine if you just want to display it in
> > a text widget or something.
> > 
> > --Chris
> > 
> > 
> > -- 
> > 
> > Chris Maj 
> > Pronunciation Guide:  Maj == May
> > Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146
> > 
> > ___
> > 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
> > 
> > -- 
> > This message has been scanned for viruses and
> > dangerous content by The CCIS.net MailScanner, and is
> > believed to be clean.
> > 
> > 
> > -- 
> > This message has been scanned for viruses and
> > dangerous content by the Bugs.Hamel.Net MailScanner, 
> > and appears to be clean.
> > 
> > 
> 
> 
> -- 
> 
> 
> 
> This message was sent using IMP, the Internet Messaging Program.

___
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

Re: [Asterisk-Users] PSTN gateway

[Philipp von Klitzing]

Hi!

> Has anyone come across a small residential PSTN gateway?

It's all there:
http://www.voip-info.org/wiki-VOIP+Phones

Cheers, Philipp


___
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

Re: [Asterisk-Users] Standalone FXO device

[Philipp von Klitzing]

Hi!

> http://www.clipcomm.co.kr/
> 
> They were worster than any device. Device itself costed me $270/- including
> shipping but not working.

Could you be a bit more specific as to what is not working? Just 
curious...

Philipp


___
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

Re: [Asterisk-Users] New Windows IAX Client

[Ken Alker]

Please pardon my ignorance, but I've got some newbie questions for you:

1) Is your IAX client essentially identical in functionality to a software 
SIP phone?

2) I was under the impression that IAX was a protocol made up specifically 
for * boxes to pass calls to each other.  Am I mistaken about the purpose 
of IAX, or are you just using it in a different, unintended way?

3) Does IAX basically give you everything SIP does and vice-versa?  IOW, is 
it just another signaling protocol like H.323, MCGP, and SIP or is it 
totally different?

--On Wednesday, January 21, 2004 6:01 PM -0600 Steven Sokol 
<[EMAIL PROTECTED]> wrote:

Announcing a new Windows-based IAX/IAX2 client.  Please download it and
give it a try.  Let me know about any bugs, and any missing features.  I
have yet to come up with a catchy name for it, so at this point it calls
itself IAX Phone.  (Suggestions?  Non-derogatory suggestions,
preferably).
Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip

Reference & Support Page: http://www.sokol-associates.com/IaxPhone.htm

Features:

- Works correctly for both inbound and outbound calls!
- Registers With Multiple Servers
- 4 Line Appearances
- Direct IAX URL Dialing (user:[EMAIL PROTECTED]/[EMAIL PROTECTED])
- 20 Speed Dials
- Native IAX Blind Transfer (no more wasted pound key)
- Drag/Drop Transfer (right-click & drag a line to a Speed Dial to
transfer)
- Last number redial
- Message Waiting Indicator (Native IAX, no Manager configuration
reqired)
- Message Waiting Count *
- Registration Indicator
- Missed Calls Indicator
- Mini-Mode for low screen-real-estate usage.
- Multiple Audio Output Devices (use speakers for Ring, headset for
audio I/O)
- Supports IAX/IAX2
- Supports DSP Filters: AGC, Echo Cancellation, Denoise
- Integrated call timer
- Integrated with the Eutectics IPP200 USB handset
- Direct dialing for IAXTel users (no need for [EMAIL PROTECTED])
- Intercom Calls (auto-answer, use speakers for audio output)**
Coming Soon:

- Call Log
- Phonebook Dialing
- Outlook Integration (dialing & screen pops)
- Programmable feature buttons (DND, Forward, Direct-To-Voicemail
Transfer, etc.)
- Documentation (sorry...;-)
- Additional Screen formats (toolbar, icon-tray)
- Keyboard Shortcuts
- User-defined ring-tones (wav, midi, etc.)
- Personalized ring-tones (assigned in phonebook)
- Enhanced speed dial "dashboard" as DSS (Manager integration) for
Operator console
- Call rejection/Zapateller
- Optional IAX debugging and logging
- Auto-answer options (either all calls or by phone-book entry)
- Integrated call recording
Hopeful Additions:

- Simple text messaging
- Local conferencing
- Screen-saver awareness (auto "Away" when screen saver kicks on)
This is the very first release so there will be bugs (as always, use at
your own risk).  Please let me know what they are ASAP.  I will
hopefully have the next release (with many of the "coming soon"
features) out later in the week or over the weekend.
Thanks,

Steve Sokol

* Requires a small change to chan_iax2.c in order to return valid
message counts.  See the support page.
** Requires a special macro in extensions.conf that replaces the
caller's real number with a special Intercom indicator ID.  See the
support page.
___
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
___
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

Re: [Asterisk-Users] Grandstream transfer solution + DTMF translation possible?

[James Sharp]

>
> The solution to the problems with the Grandstream 1.0.4.39 firmware is
> to use inband (in-audio) DTMF.  Neither the RFC2833 nor INFO seem to
> work.

Don't the Grandstreams send a DTMF 'F' INFO message on a hookflash? 
Shouldn't be that hard to change chan_sip to register an 'F' as an
AST_FLASH control message.

I'd do it, but I couldn't test it...since it seems that Grandstream hasn't
found it in the goodness of their heart to add hookflash support to the
HandyTone ATA.
___
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

Re: [Asterisk-Users] ETSI PRI ISDN Signalling

[Samuel Jimenez]

  Hi,

  Assuming that the problem *is not* soft settings I would recommend you to
verify that the channel mapping is the same in your adapter as in your
E100P.   Some times they do not match, and the D channel  does not arrive
where expected at each end.

  As you said, just a clue.

  Rgds

  Sam\\\




  - Original Message - 
  From: "Daniel Bichara" <[EMAIL PROTECTED]>
  To: <[EMAIL PROTECTED]>
  Sent: Thursday, January 22, 2004 1:38 PM
  Subject: [Asterisk-Users] ETSI PRI ISDN Signalling


  > Hi All,
  >
  > I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to
  > config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN
  > Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any
clue?
  >
  > Daniel
  >
  >
  > ___
  > 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

___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[Greg Boehnlein]

On Thu, 22 Jan 2004, WipeOut wrote:

> >Personal opinion here, but if you are relying on RedHat to be your 
> >security provider, you have no business administering a system connected 
> >to the Internet. Sure, they make it easier, but common sense and a solid 
> >understanding of the applications and code that your system is based on 
> >are a hell of a lot more comforting.
>
> Dude, with all due respect take a look at point 11 on your best practice 
> PDF that you said I should read..

Yes. And if you are planning on using a RedHat 7.3 system, then turn to 
Progeny. If you use something along the lines of Debian, you don't have a 
"vendor" to deal with. Or become your own support mechanism and roll your 
own fixes by keeping up to date.

> I am not saying that I don't agree with your other points, I do, but the 
> fact still remains that the updates from the distro provider are vitaly 
> important to the running of a secure system in addition to the 
> firewalling, stopping of unused services, the removal of packages that 
> are not used and all the other things..

> Also to say that there are more vulnerabilities in the newer systems 
> seems a little odd to me since the newer systems are usually grown from 
> the older systems and generally if there is an exploit in a newer 
> package it is likely to be in the older one as well..

RedHat 8 and 9 add a lot more packages to the mix as well as use newer GCC 
and Glibc. "New" does not equal "More Secure". RedHat 8 and 9 are a pretty 
radical departure from the 7.3 train. Add new code, add new potentials for 
exploits. ;)

> Finally the fact that more exploits are discoverd in a shorter time 
> frame on the newer distro's is probably a testament to the fact that the 
> popularity of linux is spreading and growing almost exponetialy so it 
> stands to reason that more will be created and discoverd in a storter 
> time scale than before..
> 
> Anyway this is undoubtedly a topic that could go on forever with 
> everyone having an opinion, so I guess we can say that we each have out 
> own opinion about it and leave it there..

Yes.. and it's off topic, and just short of a flame war! ;) How about you 
grab the SRPMS that I posted and see if you can install / build them on 
RedHat 9.0 for us? :)

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



___
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

Re: [Asterisk-Users] disable transfer on outgoing calls?

[Philipp von Klitzing]

Hi!

> of the above scenario using 'Dial', and confirmed that it works). however,
> i can't seem to figure out how to disable transfer for outgoing calls that
> are initiated through the Manager interface (or through .call files, for that 
> matter).

Interesting problem.

Try to use a "Local" channel and an extension in your call file. That way 
you can trigger a Dial() statement from your dialplan for both legs, and 
there you should have full control over transfer rights... check the wiki 
for local channels (and pay attention to the /n option).

Cheers, Philipp


___
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

Re: [Asterisk-Users] Is there a way to # of agents logged into a queue ?

[C. Maj]

Nope, I'm an idiot.  Here's the patch :P

On Thu, 22 Jan 2004, Bill Hamel waxed:

> Hi Chris,
> 
> This sounds what I am looking for, many thanks !
> 
> Also, I do not see an attachment, the patch that is :)
> 
> I dont know if the list strips attachments, perhaps send it to my email address
> [EMAIL PROTECTED]
> 
> Thanks again,
> -bh
> 
> 
> Quoting "C. Maj" <[EMAIL PROTECTED]>:
> > I attached a patch I've been using to show the # of agents
> > (members) and callers on a per queue basis.  It adds a new
> > manager command, "AgentQueues".  It returns on the manager
> > interface the following for each queue:
> > 
> > Queue: queuename
> > Agents: #
> > Callers: #
> > 
> > There's another manager command, "QueueStatus", that might be
> > what your are looking for.  There's also "Queues" but that
> > is a PITA to parse.  Fine if you just want to display it in
> > a text widget or something.
> > 
> > --Chris
> > 
> > 
> > -- 
> > 
> > Chris Maj 
> > Pronunciation Guide:  Maj == May
> > Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146
> > 
> > ___
> > 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
> > 
> > -- 
> > This message has been scanned for viruses and
> > dangerous content by The CCIS.net MailScanner, and is
> > believed to be clean.
> > 
> > 
> > -- 
> > This message has been scanned for viruses and
> > dangerous content by the Bugs.Hamel.Net MailScanner, 
> > and appears to be clean.
> > 
> > 
> 
> 
> 

-- 

Chris Maj 
Pronunciation Guide:  Maj == May
Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146
--- ../../../../asterisk/asterisk/apps/app_queue.c  2004-01-06 16:45:20.0 
-0500
+++ app_queue.c 2004-01-22 11:05:49.0 -0500
@@ -1403,6 +1403,57 @@
return RESULT_SUCCESS;
 }
 
+static int agent_queues_show(int fd, int argc, char **argv)
+{
+   struct ast_call_queue *q;
+   struct queue_ent *qe;
+   struct member *mem;
+   time_t now;
+   char max[80];
+int agents=0;
+int callers=0;
+   
+   time(&now);
+   if (argc != 2)
+   return RESULT_SHOWUSAGE;
+   ast_mutex_lock(&qlock);
+   q = queues;
+   if (!q) {   
+   ast_mutex_unlock(&qlock);
+   ast_cli(fd, "No queues.\n");
+   return RESULT_SUCCESS;
+   }
+   while(q) {
+   ast_mutex_lock(&q->lock);
+   if (q->maxlen)
+   snprintf(max, sizeof(max), "%d", q->maxlen);
+   else
+   strcpy(max, "unlimited");
+   ast_cli(fd, "Queue: %s\r\n", q->name);
+   if (q->members) {
+mem = q->members;
+while (mem) {
+mem = mem->next;
+agents++;
+}
+}
+   ast_cli(fd, "Agents: %d\r\n", agents);
+   if (q->head) {
+qe = q->head;
+while (qe) {
+qe = q->next;
+callers++;
+}
+}
+   ast_cli(fd, "Callers: %d\r\n", callers);
+   ast_mutex_unlock(&q->lock);
+   q = q->next;
+   }
+   ast_mutex_unlock(&qlock);
+   ast_cli(fd, "\n");
+   return RESULT_SUCCESS;
+}
+
 /* JDG: callback to display queues status in manager */
 static int manager_queues_show( struct mansession *s, struct message *m )
 {
@@ -1410,6 +1461,11 @@
return queues_show( s->fd, 2, a );
 } /* /JDG */
 
+static int manager_agent_queues_show( struct mansession *s, struct message *m )
+{
+   char *a[] = { "show", "agentqueues" };
+   return agent_queues_show( s->fd, 2, a );
+}
 
 /* Dump queue status */
 static int manager_queues_status( struct mansession *s, struct message *m )
@@ -1473,6 +1529,7 @@
ast_cli_unregister(&cli_show_queues);
ast_manager_unregister( "Queues" );
ast_manager_unregister( "QueueStatus" );
+   ast_manager_unregister( "AgentQueues" );
return ast_unregister_application(app);
 }
 
@@ -1484,6 +1541,7 @@
ast_cli_register(&cli_show_queues);
ast_manager_register( "Queues", 0, manager_queues_show, "Queues" );
ast_manager_register( "QueueStatus", 0, manager_queues_status, "Queue 
Status" );
+   ast_manager_register( "AgentQueues", 0, manager_agent_queues_show, 
"AgentQueues" );
 
// [PHM 06/26/03]
ast_register_application(app_aqm, aqm_exec, app_aqm_synopsis, 
app_aqm_descrip) ;

[Asterisk-Users] Source code for Iax Phone (the new IAX client) now available

[Steven Sokol]

The source code for IAX Phone is now available.  You can get it from the
downloads page at the main Iax Phone site:

http://www.sokol-associates.com/IaxPhone.htm

The package currently contains the source for the GUI portion of the
phone, plus the ActiveX components that interface with the IAX server.
The code for the lowest-level portions of the project (the iaxClient
library that creates the WIAX1 and WIAX2 DLLs) will be released once the
iaxClient community has synchronized.

If anybody out there is a Windows coder and would like to work on
improving this version (especially the stability) please let me know.
If anybody out there would like to take on the challenge of porting this
to Linux and Mac, please also contact me.

Thanks,

Steven

P.S. - You can download the binary installation package from that same
location.

The bug/feature tracker page is here:
http://sokol-associates.com/IaxBugTracker.htm



___
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

Re: [Asterisk-Users] Standalone FXO device

[Andrew Thompson]

- Original Message -
From: "Ariel Batista" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 3:59 PM
Subject: Re: [Asterisk-Users] Standalone FXO device


> I found information on a D-link DVG-1120S (Sip unit.) It has 2 FXS ports
> and on FXO port.  This would make  a nice small office connection for
> Asterisk.  But I can not find where they are selling these units.  Even
> on Dlinks shop site it's not listed.  Has anyone here found one of these
> unit's and tested it out?
>

According to the text(for the DVG-1120 at least), their gateways have two
FXS ports and a "lifeline" port:

Two loop-start FXS RJ-11 ports
One PSTN POTS RJ-11 port for Life Line

A lifeline port is just for use if the power goes out. It hard connects one
of your analog handsets to your POTS line(which you must still maintain).


Andrew Thompson http://aktzero.com/

___
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

Re: [Asterisk-Users] R2 support

[CW_ASN - Gus]

> Maybe Telefonica (the same from .ar) is not big enough!

By the sight Telefónica in Brazil is not very serious, in Argentina offers
ISDN in all country, for all kinds of teleservices... I'm sure of that.


___
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

RE: [Asterisk-Users] New Windows IAX Client

[Steven Sokol]

I'll add these to the wish list.  I just put up a new set of pages
related to IAX Phone.  You can find them at:

http://www.sokol-associates.com/IaxPhone.htm

One of the new pages is a bug/feature request tracker.  Feel free to add
as many feature requests as you wish.

Thanks for the compliment.  It has been a fun project.  Always glad to
see others enjoy the results.

Steven

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Samuel Jimenez
> Sent: Thursday, January 22, 2004 1:52 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] New Windows IAX Client
> 
> 
>   Steven,
> 
>   This is really great!!   Very professional and features comparable
to
> what
> one find in major brands softphones.
> 
>   I wonder if it is possible/planned..
> 
>   - Programmable Directory Number buttons (DNs linked to specific
> buttons)?
>   - A special version capable to run from removable media (USB or FDD)
> with
> no need from setup installation?  This is great for frequent travelers
and
> for first contact on site demos.
>   - video phone feature?
> 
> 
>   Regards
> 
> 
>   Sam\\\
> 
> 
>   - Original Message -
>   From: "Steven Sokol" <[EMAIL PROTECTED]>
>   To: <[EMAIL PROTECTED]>;
> <[EMAIL PROTECTED]>
>   Cc: <[EMAIL PROTECTED]>
>   Sent: Wednesday, January 21, 2004 6:01 PM
>   Subject: [Asterisk-Users] New Windows IAX Client
> 
> 
>   > Announcing a new Windows-based IAX/IAX2 client.  Please download
it
> and
>   > give it a try.  Let me know about any bugs, and any missing
features.
> I
>   > have yet to come up with a catchy name for it, so at this point it
> calls
>   > itself IAX Phone.  (Suggestions?  Non-derogatory suggestions,
>   > preferably).
>   >
>   > Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip
>   >
>   > Reference & Support Page:
http://www.sokol-associates.com/IaxPhone.htm
>   >
>   >
>   > Features:
>   >
>   > - Works correctly for both inbound and outbound calls!
>   > - Registers With Multiple Servers
>   > - 4 Line Appearances
>   > - Direct IAX URL Dialing
(user:[EMAIL PROTECTED]/[EMAIL PROTECTED])
>   > - 20 Speed Dials
>   > - Native IAX Blind Transfer (no more wasted pound key)
>   > - Drag/Drop Transfer (right-click & drag a line to a Speed Dial to
>   > transfer)
>   > - Last number redial
>   > - Message Waiting Indicator (Native IAX, no Manager configuration
>   > reqired)
>   > - Message Waiting Count *
>   > - Registration Indicator
>   > - Missed Calls Indicator
>   > - Mini-Mode for low screen-real-estate usage.
>   > - Multiple Audio Output Devices (use speakers for Ring, headset
for
>   > audio I/O)
>   > - Supports IAX/IAX2
>   > - Supports DSP Filters: AGC, Echo Cancellation, Denoise
>   > - Integrated call timer
>   > - Integrated with the Eutectics IPP200 USB handset
>   > - Direct dialing for IAXTel users (no need for [EMAIL PROTECTED])
>   > - Intercom Calls (auto-answer, use speakers for audio output)**
>   >
>   > Coming Soon:
>   >
>   > - Call Log
>   > - Phonebook Dialing
>   > - Outlook Integration (dialing & screen pops)
>   > - Programmable feature buttons (DND, Forward, Direct-To-Voicemail
>   > Transfer, etc.)
>   > - Documentation (sorry...;-)
>   > - Additional Screen formats (toolbar, icon-tray)
>   > - Keyboard Shortcuts
>   > - User-defined ring-tones (wav, midi, etc.)
>   > - Personalized ring-tones (assigned in phonebook)
>   > - Enhanced speed dial "dashboard" as DSS (Manager integration) for
>   > Operator console
>   > - Call rejection/Zapateller
>   > - Optional IAX debugging and logging
>   > - Auto-answer options (either all calls or by phone-book entry)
>   > - Integrated call recording
>   >
>   > Hopeful Additions:
>   >
>   > - Simple text messaging
>   > - Local conferencing
>   > - Screen-saver awareness (auto "Away" when screen saver kicks on)
>   >
>   >
>   > This is the very first release so there will be bugs (as always,
use
> at
>   > your own risk).  Please let me know what they are ASAP.  I will
>   > hopefully have the next release (with many of the "coming soon"
>   > features) out later in the week or over the weekend.
>   >
>   > Thanks,
>   >
>   > Steve Sokol
>   >
>   > * Requires a small change to chan_iax2.c in order to return valid
>   > message counts.  See the support page.
>   >
>   > ** Requires a special macro in extensions.conf that replaces the
>   > caller's real number with a special Intercom indicator ID.  See
the
>   > support page.
>   >
>   >
>   > ___
>   > 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
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] ETSI PRI ISDN Signalling

[CW_ASN - Gus]

Please send your zaptel.conf to see what's going on.


- Original Message - 
From: "Daniel Bichara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 4:38 PM
Subject: [Asterisk-Users] ETSI PRI ISDN Signalling


> Hi All,
> 
> I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to 
> config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN 
> Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue?
> 
> Daniel
> 
> 
> ___
> 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

___
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

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[CW_ASN - Gus]

I don't know, but I can test in a very short time.
I let you know for details.

- Original Message -
From: "Thomas Haeger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 4:07 PM
Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


> Has somebody got it work at all ?
> I mean data calls (ISDN 64k) through asterisk.
>
> Regards,
>
> Thomas.
>
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Auftrag von Thomas
> Haeger
> Gesendet: Donnerstag, 22. Januar 2004 19:07
> An: [EMAIL PROTECTED]
> Betreff: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
>
>
> Hi,
>
> we tried following scenario:
>
>
> DTAG (S0) at our office Datacall with AVMFritz (PSTN) ---> Colo
> TelesSwitch --> CoLo Asterisk (---> PSTN)
>
> I think, no i know that the Teles Switch can route 64k data calls
> here is the Teles Trace:
>
> #08SETUP-->|
> 15:29:40,378 02 01 78 AE   |
>  08 02 03 90 05|
> Bearer Caps  04 02 88 90   |
> Channel Id   18 03 A1 83 9B|
> Calling PN   6C 0C 21 83 33 33 32 38   |
>  33 33 34 37 37 38 |
> Called PN70 09 C1 36 33 34 39 34   |
>  34 34 31  |
>|<--RR
> #08
>|   15:29:40,388 02 01 01 7A
>|<--SETUP ACKNOWLEDGE
> #08
>|   15:29:40,398 00 01 AE 7A
>|08 02 83 90 0D
>|   Channel Id   18 03 A9 83 9B
>|<--SETUP
> #12
>|   15:29:40,408 00 01 2A D4
>|08 02 16 60 05
>|   Bearer Caps  04 02 88 90
>|   Channel Id   18 03 A1 83 88
>|   Calling PN   6C 0C 21 80 33 33
32
> 38
>|33 33 34 37 37 38
>|   Called PN70 09 81 36 33 34
39
> 34
>|34 34 31
> #08   RR-->|
> 15:29:40,408 00 01 01 B0   |
> #12   RR-->|
> 15:29:40,418 00 01 01 2C   |
> #12SETUP ACKNOWLEDGE-->|
> 15:29:40,418 02 01 D4 2C   |
>  08 02 96 60 0D|
> Channel Id   18 03 A9 83 88|
> Progress Ind 1E 02 81 82   |
>|<--RR
> #12
>|   15:29:40,418 02 01 01 D6
> #12SETUP-->|
> 15:29:40,428 02 01 D6 2C   |
>  08 02 1A 21 05|
> Bearer Caps  04 03 88 90 A3|
> Channel Id   18 03 A1 83 81|
> Calling PN   6C 0C 41 81 33 33 32 38   |
>  33 33 34 37 37 38 |
> Called PN70 0D C1 30 33 33 32 38   |
>  33 30 37 37 37 33 33  |
>|<--RELEASE COMPLETE
> #12
>|   15:29:40,428 00 01 2C D8
>|08 02 9A 21 5A
>|08 02 80 D8
>|[Incompatible
> destinat
>|ion]
> #12   RR-->|
>
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Auftrag von CW_ASN -
> Gus
> Gesendet: Donnerstag, 22. Januar 2004 17:24
> An: [EMAIL PROTECTED]
> Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
>
>
> The incoming call request Unrestricted and 64K, and this looks like ok,
but
> in the SETUP_ACK the called number parameters shows: "Ext: 1  Progress
> Description: Called equipment is non-ISDN. (2) ]", like as is not an ISDN
> equipment.
> In the most of cases, Information transfer rate = to '64 kbit/s', and Info
> transfer capability = 'real bw required'.
>
> Are you sure that the equipment attached to * can be used in 64K?
>
> Regards,
>
> Gus
>
> - Original Message -
> From: "Thomas Haeger" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, January 22, 2004 12:28 PM
> Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
>
>
> > Hi ,
> >
> > maybe someone knows what's going wrong...
> >
> > The incoming data call will not really identified as ISDN 64k/Data
> >
> > Here my pri debug ouput
> >
> > < Protocol Discriminator: Q.931 (8)  len=39
> > < Call Ref: len= 2 (reference 5635/0x1603) (Originator)
> > < Message type: SETUP (5)
> > < Beare

Re: [Asterisk-Users] Mailing List Lag

[Tilghman Lesher]

On Thursday 22 January 2004 14:04, WipeOut wrote:
> Christian Hoffmeyer wrote:
> >- Original Message -
>
> From: "Steve Foy" <[EMAIL PROTECTED]>
>
> >To: "Asterisk-Users" <[EMAIL PROTECTED]>
> >Sent: Thursday, January 22, 2004 3:18 AM
> >Subject: Re: [Asterisk-Users] Mailing List Lag
> >
> >>On Thu, Jan 22, 2004 at 08:39:09AM +, Steve Foy wrote:
> >>>I'd be willing to host the list, I guess it just depends on how
> >>> many emails/day the Asterisk list goes through...
> >>
> >>Seems to be around 1200 emails per week, for the one week that I
> >> counted anyway...
> >
> >Last week I was complaining to John, at Digium, about the list lag. 
> > Mark walked by his office so John voiced my complaint.  Mark then
> > said that the mailing list does 9 million messages a day.
>
> I have got 685 mails in the last 5 days, that works out to 137 per
> day..
>
> Lets make it 150 per day since the last 5 days includes a Sunday..
>
> 900 mails / 150 per day = 6 Subscribers..
>
> Is the Asterisk community anywhere near that big??
>
> I am not disagreeing with Mark, its his server, it just seems that 9
> million per day is a little high..

I'd believe that the mail server handles that many messages.  Consider
that all messages are handled twice:  once coming in, thousands of times
out.  Add to that the number of bounces and mail delayed messages, due
to various mailing servers being down.  Now also, don't forget that
Digium has a dozen or more people with email accounts, each of whom
probably deal with hundreds of messages a day.  Also don't forget that
the mail server has to deal with spam, especially since there's quite a
few digium.com addresses which are publically displayed.  That adds up
to quite a lot of messages.

And given any mailing list for a community, it is probable that the vast
majority of those subscribed NEVER post to the list.  While it tends to
be the same group of people who respond to questions, the new users
asking questions seems to be different every week.

-Tilghman

___
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

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[CW_ASN - Gus]

You're still receiving [Incompatible destination], this cause is used when
bearer capabilities aren't equal.


- Original Message -
From: "Thomas Haeger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 3:07 PM
Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


> Hi,
>
> we tried following scenario:
>
>
> DTAG (S0) at our office Datacall with AVMFritz (PSTN) ---> Colo
> TelesSwitch --> CoLo Asterisk (---> PSTN)
>
> I think, no i know that the Teles Switch can route 64k data calls
> here is the Teles Trace:
>
> #08SETUP-->|
> 15:29:40,378 02 01 78 AE   |
>  08 02 03 90 05|
> Bearer Caps  04 02 88 90   |
> Channel Id   18 03 A1 83 9B|
> Calling PN   6C 0C 21 83 33 33 32 38   |
>  33 33 34 37 37 38 |
> Called PN70 09 C1 36 33 34 39 34   |
>  34 34 31  |
>|<--RR
> #08
>|   15:29:40,388 02 01 01 7A
>|<--SETUP ACKNOWLEDGE
> #08
>|   15:29:40,398 00 01 AE 7A
>|08 02 83 90 0D
>|   Channel Id   18 03 A9 83 9B
>|<--SETUP
> #12
>|   15:29:40,408 00 01 2A D4
>|08 02 16 60 05
>|   Bearer Caps  04 02 88 90
>|   Channel Id   18 03 A1 83 88
>|   Calling PN   6C 0C 21 80 33 33
32
> 38
>|33 33 34 37 37 38
>|   Called PN70 09 81 36 33 34
39
> 34
>|34 34 31
> #08   RR-->|
> 15:29:40,408 00 01 01 B0   |
> #12   RR-->|
> 15:29:40,418 00 01 01 2C   |
> #12SETUP ACKNOWLEDGE-->|
> 15:29:40,418 02 01 D4 2C   |
>  08 02 96 60 0D|
> Channel Id   18 03 A9 83 88|
> Progress Ind 1E 02 81 82   |
>|<--RR
> #12
>|   15:29:40,418 02 01 01 D6
> #12SETUP-->|
> 15:29:40,428 02 01 D6 2C   |
>  08 02 1A 21 05|
> Bearer Caps  04 03 88 90 A3|
> Channel Id   18 03 A1 83 81|
> Calling PN   6C 0C 41 81 33 33 32 38   |
>  33 33 34 37 37 38 |
> Called PN70 0D C1 30 33 33 32 38   |
>  33 30 37 37 37 33 33  |
>|<--RELEASE COMPLETE
> #12
>|   15:29:40,428 00 01 2C D8
>|08 02 9A 21 5A
>|08 02 80 D8
>|[Incompatible
> destinat
>|ion]
> #12   RR-->|
>
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Auftrag von CW_ASN -
> Gus
> Gesendet: Donnerstag, 22. Januar 2004 17:24
> An: [EMAIL PROTECTED]
> Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
>
>
> The incoming call request Unrestricted and 64K, and this looks like ok,
but
> in the SETUP_ACK the called number parameters shows: "Ext: 1  Progress
> Description: Called equipment is non-ISDN. (2) ]", like as is not an ISDN
> equipment.
> In the most of cases, Information transfer rate = to '64 kbit/s', and Info
> transfer capability = 'real bw required'.
>
> Are you sure that the equipment attached to * can be used in 64K?
>
> Regards,
>
> Gus
>
> - Original Message -
> From: "Thomas Haeger" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, January 22, 2004 12:28 PM
> Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
>
>
> > Hi ,
> >
> > maybe someone knows what's going wrong...
> >
> > The incoming data call will not really identified as ISDN 64k/Data
> >
> > Here my pri debug ouput
> >
> > < Protocol Discriminator: Q.931 (8)  len=39
> > < Call Ref: len= 2 (reference 5635/0x1603) (Originator)
> > < Message type: SETUP (5)
> > < Bearer Capability (len= 2) [ Ext: 1  Q.931 Std: 0  Info transfer
> > capability: Unrestricted digital information (8)
> > <  Ext: 1  Trans mode/rate: 64kbps,
> circuit-mode
> > (16)
> > <  Ext: 0  User information layer 1: Unknown
> > (24)
> > < Channel ID (len= 5) [ Ext: 1  IntID: Implic

Re: [Asterisk-Users] OH323 config file format

[Deepakumar JV]

Hi Tom
 
Thanks for your reply.
 
SIP: I am using for my regular 
international calls, hence i have not tested with multiple calls. i have never 
experienced any crash. its quite stable and gives good performance. my system 
config is pentium III 300Mhz with 64MB.
 
OH323: I am trying to register with www.mywebcalls.com. My basic understanding 
of H323 is, since * can act as a client (H323) as well as a gateway (H323) I 
should be able to use any of the H323 providers (even if they provide they own 
H323 based clients or web based H323 clients). So this provider has given me a 
username and password, that's all.. nothing else. Since they provide a web 
based client, they don't support any other configuration i am unable to get more 
details from them. But by using their web based client I have found out the IP 
of (i am not sure whether its a GW or GK). I am not sure where to enter the 
username and password that they have given.
 
Thanks again
Deepak

  - Original Message - 
  From: 
  T. 
  Chan 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, January 22, 2004 10:00 
  PM
  Subject: RE: [Asterisk-Users] OH323 
  config file format
  
  Hi, 
  Deepak, how are you?
   
  I 
  don't quite understand what you meant by username and password sending calls 
  to a H323 service provider, do you mean you have to register onto their 
  gatekeeper? Or otherwise, you should not need username and 
  password.
   
  Meantime, I am trying to setup up SIP calling to a service provider, 
  can you let me know what is the maximum number of calls you have experienced 
  with sending SIP calls to your service provider? Have you experienced any 
  crash? What is the configuration of your  computer?
   
  Thanks
   
  Tom
  
-Original Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]On Behalf Of Deepakumar 
JVSent: Wednesday, January 21, 2004 12:38 PMTo: 
[EMAIL PROTECTED]Subject: [Asterisk-Users] OH323 
config file format
Hello
 
I am trying to configure my 
extensions.conf and oh323.conf to termination calls to a H323 service 
provider. Can anyone send me a sample config files? or tell me where to put 
the username and password which my service provider has given? also how to 
put the Dial command in extensions.conf
 
Asterisk rocks. I have a SIP provider 
configured for all my international calls and it works absolutely fine. Its 
cool.
 
Thanks in advance
 
Regards
Deepak

[Asterisk-Users] Variable to play all gsm files in a directory?

[jaycard]

Is there a variable that can be used with the playback command to
play all gsm sound files in a directory? For example like '*.*' <- which
I know doesn't work. Is the only solution for this to create an AGI
script and use the stream_file command ?
Thanks,
J.C.
___
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

Re: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)

[Jeff Gustafson]

On Wed, 2004-01-21 at 16:23, Siggi Langauf wrote:
> Hi Jeff,
> 
> On Wed, 21 Jan 2004, Jeff Gustafson wrote:
> 
> > "Error Verifying Config Info"
> > then
> > "Registering"
> [...]
> > Any ideas?
> 
> Sure: Remove the SIPDefault.cnf.xml file from your TFTP server!
> It's not required, and unless you understand what it's good for, it will
> keep your phone rebooting in an infinite loop.

Well, it's XMLDefault.cnf.xml on my box.  It doesn't have the SIP
firmware load.

> If your phone has some stale CallManager settings that are preventing you
> from using it, just follow these steps:
> 
> 1) make sure you have a DHCP server on your network that will assign the
>phone an IP address, network mask, DNS server and default router
>It should also assign the address off your _Asterisk_ server as
>TFTP server (DHCP: "next-server")

That kinda helped.  Now it's pointing right at Asterisk.  The problem
is still that I have better luck with the old chan_skinny code over the
chan_sccp.  chan_sccp still puts the phone in a config verification
error loop.

...Jeff

___
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

[Asterisk-Users] Rtp WARNING Messages on the Cli in safe_asterisk

[Frankie Gravato]

Hello All,

Has  anyone ever seen this before. This only happens when i'm on phone
call



-- Zap/2-1 is ringing
-- SIP/2203-c48d is ringing
-- SIP/2202-f2ad is ringing
-- SIP/2204-11cd is ringing
-- SIP/2205-ce62 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- SIP/2205-ce62 answered Zap/1-1
-- Hungup 'Zap/2-1'
Jan 22 15:54:25 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource 
temporarily unavailable
Jan 22 15:54:28 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource 
temporarily unavailable
Jan 22 15:54:31 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource 
temporarily unavailable
Jan 22 15:56:48 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource 
temporarily unavailable
Jan 22 16:08:10 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource 
temporarily unavailable
Jan 22 16:08:14 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource 
temporarily unavailable
  == Spawn extension (macro-ringall, s, 2) exited non-zero on 'Zap/1-1' in macro 
'ringall'
  == Spawn extension (from-pstn, s, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1




  

-- 
Best regards,
Frankie ([EMAIL PROTECTED])  
mailto:[EMAIL PROTECTED]

___
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

[Asterisk-Users] Grandstream transfer solution + DTMF translation possible?

[Barton Hodges]

The solution to the problems with the Grandstream 1.0.4.39 firmware is
to use inband (in-audio) DTMF.  Neither the RFC2833 nor INFO seem to
work.

However, this presents another problem.  When I'm using g729 to place
a call, I get the warning "Unable to process inband DTMF" because
inband is not supposed to work with g729 (although it does seem to
work when I've tried it so far).  Can Asterisk convert between
different modes of DTMF?  For instance, my phone would use inband, and
Asterisk would convert this to rfc2833 before reaching the channel I
am connected to?  I've tried using dtmfmode=rfc2833 in the service
definition in sip.conf.

Thanks,

Barton


___
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

[Asterisk-Users] hardware compatibility

[kemal asad]

I am setting up my soho, am treating myself with some new and cool
hardware but before i go crazy on these new gadgets i wanted to check if
my idea made sense. i wanted to get http://openbrick.org/ as a router to
connect my office to adsl . then on a computer with the following
motherboard
http://www.motherboardsdirect.com/specshow.asp?itemnum=K8TM2FAR

 behing the firewall router i would put if compatible  :
http://digium.com/index.php?menu=wildcard_x100p.
i am not sure i understand what asterisk is all about but i guess i will
need to get some special handset. if you have any suggestion please
advise.
Thanks.
Kemal


___
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

RE: [Asterisk-Users] OH323 config file format

[T. Chan]

Hi, 
Deepak, how are you?
 
I 
don't quite understand what you meant by username and password sending calls to 
a H323 service provider, do you mean you have to register onto their gatekeeper? 
Or otherwise, you should not need username and password.
 
Meantime, I am trying to setup up SIP calling to a service provider, can 
you let me know what is the maximum number of calls you have experienced with 
sending SIP calls to your service provider? Have you experienced any crash? What 
is the configuration of your  computer?
 
Thanks
 
Tom

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Deepakumar 
  JVSent: Wednesday, January 21, 2004 12:38 PMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] OH323 
  config file format
  Hello
   
  I am trying to configure my 
  extensions.conf and oh323.conf to termination calls to a H323 service 
  provider. Can anyone send me a sample config files? or tell me where to put 
  the username and password which my service provider has given? also how to put 
  the Dial command in extensions.conf
   
  Asterisk rocks. I have a SIP provider 
  configured for all my international calls and it works absolutely fine. Its 
  cool.
   
  Thanks in advance
   
  Regards
  Deepak

Re: [Asterisk-Users] asterisk 0.7.1 - mysql

[WipeOut]

Jonathan Moore wrote:

MySQL support was moved out to "addons." You don't have to use ODBC
to point to MySQL, but I would say it's probably a good idea. There
is no guarantee that anyone will continue to update the mysql code
now that the license on mysql has changed.
 

Can anyone explain what has changed about the MySQL license and when it changed?
I wasn't aware of this.
 

IIRC for MySQL 4 and above the licence have changed from LGPL to GPL and 
this had a knock on affect for all software that uses any od the MySQL 
code or libs..

I am sure someone else will be able to give a more detailed explaination 
or you can search the archives for previous discussions..

Later..

___
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

Re: [Asterisk-Users] asterisk 0.7.1 - mysql

[Brian West]

I would love to use native Mysql also but the whole reason I wrote
cdr_odbc is to sidestep the GPL issues and still have mysql without having
to update this and that all the time.

bkw

On Thu, 22 Jan 2004, WipeOut wrote:

> Tilghman Lesher wrote:
>
> >On Thursday 22 January 2004 08:01, Andrew Thompson wrote:
> >
> >
> >>- Original Message -
> >>From: "Dawid Mielnik" <[EMAIL PROTECTED]>
> >>To: <[EMAIL PROTECTED]>
> >>
> >>
> >>>Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL.
> >>>Does this new version of * only work through ODBC ? Do I have
> >>>connect to MySQL through ODBC now ?
> >>>
> >>>
> >>MySQL support was moved out to "addons." You don't have to use ODBC
> >>to point to MySQL, but I would say it's probably a good idea. There
> >>is no guarantee that anyone will continue to update the mysql code
> >>now that the license on mysql has changed.
> >>
> >>
> >
> >There's no guarantee that Asterisk will be maintained either.  There are
> >lots of people who are interested enough, though, to make sure that it
> >continues to be maintained.
> >
> >As far as cdr_addon_mysql.c support, I'm committed to maintaining it
> >for the forseeable future.  In fact, I've just uploaded a patch to the
> >bugtracker to add a CLI command to this module:
> >
> >http://bugs.digium.com/bug_view_page.php?bug_id=902
> >
> >-Tilghman
> >
> >
> >
> Tilghman,
>
> I am glad to see I am not the only one who likes using MySQL.. :-)
>
> I was begining to feel that I was alone in the Asterisk/MySQL world.. ;-)
>
> PS I am only kidding...
>
> Later..
>
> ___
> 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
>
___
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

[Asterisk-Users] MGCP Problem.

[Ricardo Martinez]

Hi. 
I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk
with the next configuration files.

'--- extensions.conf


[general]
static=yes
writeprotect=yes

[globals]
ap1 => mgcp/aaln/[EMAIL PROTECTED]

[macro-apl1]
exten => s,1,Dial(${ARG1},30,Ttmr)
;exten => s,2,Voicemail(u${MACRO_EXTEN})
;exten => s,3,Hangup
;exten => s,102,Voicemail(b${MACRO_EXTEN})
;exten => s,103,Hangup

[SIP]
include => ent

[ent]
exten => 112,1,VoiceMail(b112)
exten => 111,1,Macro(apl1,${ap1})


-

--- mgcp.conf



;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 64.76.148.185

;[dlinkgw]
;host = 192.168.0.64
;context = default
;line => aaln/2
;line => aaln/1

;; The MGCP channel supports the following service codes:
;; # - Transfer
;; *67 - Calling Number Delivery Blocking
;; *70 - Cancel Call Waiting
;; *72 - Call Forwarding Activation
;; *73 - Call Forwarding Deactivation
;; *78 - Do Not Disturb Activation
;; *79 - Do Not Disturb Deactivation
;; *8 - Call pick-up
;
; known to work with Swissvoice IP10s

[64.76.148.186]
host = 64.76.148.186
context = default
line => aaln/1

-


When i hook off the endpoint i obtain the next debug in the asterisk CLI.

from 64.76.148.186:2427Verb: 'ntfy', Identifier: '6001', Endpoint:
'aaln/0/[EMAIL PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Jan 22 18:05:11 NOTICE[49159]: chan_mgcp.c:1102 find_subchannel: Gateway
'ap1' (and thus its endpoint 'aaln/0/0') does not exist
MGCP read: 
ntfy 6001 aaln/0/[EMAIL PROTECTED] MGCP 1.0
X: 0
O: l/hd

from 64.76.148.186:2427Verb: 'ntfy', Identifier: '6001', Endpoint:
'aaln/0/[EMAIL PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Jan 22 18:05:15 NOTICE[49159]: chan_mgcp.c:1102 find_subchannel: Gateway
'ap1' (and thus its endpoint 'aaln/0/0') does not exist
MGCP read: 
ntfy 6001 aaln/0/[EMAIL PROTECTED] MGCP 1.0
X: 0
O: l/hd


Any ideas!  What i'm doing wrong?

Thanks in advance


Ricardo
___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[WipeOut]

Greg Boehnlein wrote:

On Thu, 22 Jan 2004, WipeOut wrote:

 

Ken Godee wrote:
   

 

The problem with running servers based on RH 6.x, 7.x and 8 is that RH 
is not providing errata (security specifically) updates any more.. If 
you servers are not connected to the internet then, sure stay with the 
versions that are working for you, but if you have you server live on 
the internet for ant reason then this is a big issue..
   

No it isn't. If you follow best practices for your system, remove all 
unneccessary packages, and properly firewall it, you are at no greater or 
lesser risk than any other version of RedHat.

Take a look at the following:
http://www.nacs.net/~damin/linux-best-practices.pdf
 

I realise that many vulnerabilities require local access but I am still 
not going to take the chance.. I want my servers as safe as possible, 
and if that means running the latest versions of whatever then thats 
what I am going to do.. :)
   

Take a look at the number of exploits that are available for RH 8 and 9, 
and how quickly they are mounting up, and then rethink that statement. 
There are more exploits being targeted at these platforms, in a shorter 
period of time, than 7.3 and the earlier versions.

Personal opinion here, but if you are relying on RedHat to be your 
security provider, you have no business administering a system connected 
to the Internet. Sure, they make it easier, but common sense and a solid 
understanding of the applications and code that your system is based on 
are a hell of a lot more comforting.

 

Dude, with all due respect take a look at point 11 on your best practice 
PDF that you said I should read..

I am not saying that I don't agree with your other points, I do, but the 
fact still remains that the updates from the distro provider are vitaly 
important to the running of a secure system in addition to the 
firewalling, stopping of unused services, the removal of packages that 
are not used and all the other things..

Also to say that there are more vulnerabilities in the newer systems 
seems a little odd to me since the newer systems are usually grown from 
the older systems and generally if there is an exploit in a newer 
package it is likely to be in the older one as well..

Finally the fact that more exploits are discoverd in a shorter time 
frame on the newer distro's is probably a testament to the fact that the 
popularity of linux is spreading and growing almost exponetialy so it 
stands to reason that more will be created and discoverd in a storter 
time scale than before..

Anyway this is undoubtedly a topic that could go on forever with 
everyone having an opinion, so I guess we can say that we each have out 
own opinion about it and leave it there..

Later..

___
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

[Asterisk-Users] is the mike back on?

[James H. Cloos Jr.]

It looks like digium is once again in the com. zone, although only
one nameserver is listed:

com.zone.gz: DIGIUM NS LINUX-SUPPORT.NET.

Odd that it fell out exactly 30 days before it was set to expire

Now two different registrars have fubared two different companies
here.  One more and the cons. theorists will be fnording. :-/

-JimC

P.S.  NB that cons. can stand not just for conspiracy
  but also for constipated.  :-) :-) :-)

___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[Fran Boon]

On Thu, 2004-01-22 at 18:17, Jonathan Moore wrote:
> I am researching the use of White Box Enterprise Linux. Someone else in a
> similar position with a bunch of 7.x boxes created it. He took all the SRPM
> files for REL v3 and removed all the Red Hat logos and trademarks. It is the
> same software as Enterprise but you can freely copy it. They also modded the
> update scripts to work with more generic update sources. The cool thing is the
> system is completely compatible with the REL source errata which Red Hat has
> promised to continue updating for at least five years. They have also setup a
> small system of mirrors to host the update files. It looks very promising. I am
> trying this and Debian to see which will be easier to keep updates for. Info and
> ISO file available at
> 
> http://www.beau.org/~jmorris/linux/whitebox/index.html
> 
> Someone else has a similar project going but it didn't seem to be as far along.

Personally I think that Tao Linux (http://taolinux.org/) is better
set-up than Whitebox.

Anyway, for me, RedHat Enterprise Linux & it's 'rebuild' clones are the
best development target to aim for - this is the stable environment (5
years) that people like from 7.3, brought up to date & supported by
hardware manufacturers, software developers & hence corporates.

I'm running 0.7.1 on RHEL3 without any problems so far :)

F

___
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

Re: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)

[Jeff Gustafson]

On Thu, 2004-01-22 at 02:09, Jan Czmok wrote:
> Jeff Gustafson ([EMAIL PROTECTED]) wrote:
> > Maybe it's not the new chan_sccp code that's the problem.  When I put
> > in the SEP000785532D5F.cnf.xml on the tftp server, the phone never gets
> > to a usable screen.  Instead it just tries to tftp files over and over. 
> > Th one file, P00305000300.bin, I don't have.  As far as I know I can't
> > get this file unless I buy it from cisco.   Is this file absolutely
> > required for the chan_sccp code to work?
> 
> no, its not necessary required. in this case, check that the contents of
> OS79xx.TXT if they match with your current version.
> 

I didn't have that file because I thought it would make things worse.
:)  I took the number from Settings -> Model Information -> App Load ID
and put it in the OS79XX.TXT file.  This doesn't seem to satisfy the
phone.  It still tries to pick up that .bin file.  Now at least it's
trying for the .bin file I specified.  
If I have the SEP*.cnf.xml file available, the phone just says "Opening
192.168.5.254" then "Opening 192.168.5.132" over and over again.  The
.132 is the right address for Asterisk.  If I *remove* the SEP file,
then the phone lets me dial, but doesn't let me get very far because it
complains it can't see the .bin file.
Thank you for your patience so far.  I hope this process is helping
others out there with cisco phones.

...Jeff

> Also: check your SEP*.xml resp. xmlDefault File for the software
> setting, so that this matches your current installed software.
> 
> You can find the current software load somewhere in the menus.
> 
> --jan
> 
> 
> 
> > 
> > ...Jeff
> > 
> > On Wed, 2004-01-21 at 14:05, Jan Czmok wrote:
> > > > 
> > > > Kewl,  I was apparently trying to use older chan_sccp code which didn't
> > > > work.
> > > > Okay... just tried your new code.  The phones keep resetting:
> > > > 
> > > > "Error Verifying Config Info"
> > > > then
> > > > "Registering"
> > > 
> > > 
> > > you should use the CURRENT code, which is not there as a tarball.
> > > i just posted the recent tarball to the /files directory.
> > > 
> > > use this one !
> > > 
> > > Then:
> > > 
> > > Configure your dhcp to serve it like this:
> > > 
> > >host voip-phone
> > >   { 
> > >   hardware ethernet 00:30:94:C2:89:0B;
> > >   fixed-address 212.20.150.206;
> > >   option host-name "voip-phone";
> > >   option domain-name-servers 212.20.144.98;
> > >   option routers 212.20.150.1;
> > > option tftp-server-name "193.138.116.111";
> > > }
> > > 
> > > by using your IPs.
> > > 
> > > The tftp server should contain: 
> > > 
> > > - xmlDefault.CNF.XML file.
> > > - a symlink from the xmlDefault.CNF.XM file to the SEPxx.cnf.xml file
> > > - within the sepdefault, you need to define the callmanager.
> > > - some more stuff, but we'll see it later...
> > > 
> > > this should be sufficient to bring up the asterisk withthe 79xx.
> > > 
> > > --jan
> > > 
> > 

___
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

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[Maik Schmitt]

> Has somebody got it work at all ?
> I mean data calls (ISDN 64k) through asterisk.

Yes. Works fine here with a PRI from DTAG and an Ascend.

-- 
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP

pgp0.pgp
Description: PGP signature

[Asterisk-Users] Snom 200 phones not working.

[Ariel Batista]

I have 2 Snom 200 and would like to get them to work properly with
Asterisk.  With the Firmware 2.02t I am able to use the phone.  But only
one line configured.  With there newer firmware 2.03o it will not allow
me to make calls.  But I can get calls on the unit.  Again the 2nd line
is not able to be registered.  Is this an issue with Asterisk or Snom?

I could use some example configuration files.  I have followed the Snom
FAQ step by step.  But it's still not working.
-
\
\\_ Ariel Batista
//
/ Red-Fone Communications, Inc.
--
[EMAIL PROTECTED]
Ph: 786-544-1114
Fx: 305-574-0212

___
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

Re: [Asterisk-Users] Standalone FXO device

[Ariel Batista]

> clipcomm people?
> Well, I was/am looking for a device with PSTN FXO backup.
> www.dlink.com does one like that, but is way too expensive.

I found information on a D-link DVG-1120S (Sip unit.) It has 2 FXS ports
and on FXO port.  This would make  a nice small office connection for
Asterisk.  But I can not find where they are selling these units.  Even
on Dlinks shop site it's not listed.  Has anyone here found one of these
unit's and tested it out?

>
> Ta
> SJ
>
> ___
> 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

___
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

Re: [Asterisk-Users] Asterisk and gnugk

[Lubomir Christov]

yes :)

bam wrote:

This is quite possibly a daft question, but it is possible to run * 
and gnugk on the same system with gnugk acting as a proxy for 
netmeeting endpoints and feeding everything for PSTN and SIP out 
through *?

___
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
___
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

Re: [Asterisk-Users] R2 or E&M for E1 CAS pbx to pbx link

[Samuel Jimenez]

  Hi,
   
  So, u are using a new thread...  it is better if 
  you do not change the message subject so we can trace u 
  properly.
   
  Before dealing with R2 signaling which is 
  register_signaling, you need to deal with CAS or ABCD signaling which is 
  line_signaling and be able to establish (and drop) calls correctly.  Once 
  this is done, R2 signaling can take place ---if fully available 
  in *,  and R2 tones will be able to travel between 
  switches thru the established voice channel before callers be allowed to 
  talk.
   
  So, again... you need de CAS table of the far_end switch or 
  pbx.   Get it and attach it to your post. 
  Also, if u thing u will not able to negotiate with your far 
  en party, ask him/her in advance what signaling and start arrangement should 
  you use.  Should tell you something like 4wire E&M, IMMediate.  
  This info is also crucial.
   
   
  Regards
   
  Sam\\\
   
  - Original Message - 
  
From: 
M.A. 
Ali 
To: [EMAIL PROTECTED] 

Sent: Thursday, January 22, 2004 5:01 
AM
Subject: [Asterisk-Users] R2 or E&M 
for E1 CAS pbx to pbx link


hi,
thanx for the response. I just tried to work on R2 CAS but i found that 
the libr2 has not been implemented well and tested. I think in addition to 
E&M, R2 can also be used in a pbx to pbx E1 link. what do tou suggest 
Sam ??
About the R2 implementation for asterisk i have seen in the list that 
steve has implemented 95% of that...but we dont see any release of that. any 
current info on R2 development??
and Sam you are right i don't have the CAS table of the other switch. But 
i think i can get one. 
help me out in this. I have to make a E1 pbx to pbx connection using 
CAS.
thanks in advance
janjua
 
 
 

Help STOP spam with the new MSN 
8 and get 2 months FREE* ___ 
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

Re: [Asterisk-Users] Asterisk and gnugk

[Kim Hendrikse]

Yep! Just have asterisk bind to another port than 1720. Of course, your
problem with be codecs like g723.1.

  - Kim

> This is quite possibly a daft question, but it is possible to run * and 
> gnugk on the same system with gnugk acting as a proxy for netmeeting 
> endpoints and feeding everything for PSTN and SIP out through *?
> 
> 
> ___
> 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
___
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

Re: [Asterisk-Users] Using varables in MeetMe?

[Tilghman Lesher]

On Thursday 22 January 2004 12:06, Christopher Arnold wrote:
> Hi,
>
> Im trying to enable users to enter a conference number and then do a
> calculation on this and then send them to the conference. Lokk at
> this example:
>
> exten => s,1,Read(room)
> exten => s,2,SetVar,"${room}=[${room} + 2000]";
> exten => s,3,Meetme($room|pqsd)
>
> What happens is that the conference gets setup and everything, but
> the conference number is "$room". Not really what i expected...
>
> Is this a bug of a feature?

You're using variables incorrectly.  The ONLY way to substitute
variables is to use the ${} notation.  EVERY time you use this notation
the variable is substituted prior to evaluation and execution.  So to
correct your dialplan logic:

Read(room)
SetVar(room=$[${room} + 2000])
MeetMe(${room}|pqsd)

-Tilghman

___
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

Re: [Asterisk-Users] Mailing List Lag

[WipeOut]

Christian Hoffmeyer wrote:

- Original Message - 
From: "Steve Foy" <[EMAIL PROTECTED]>
To: "Asterisk-Users" <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 3:18 AM
Subject: Re: [Asterisk-Users] Mailing List Lag

 

On Thu, Jan 22, 2004 at 08:39:09AM +, Steve Foy wrote:
   

I'd be willing to host the list, I guess it just depends on how many
emails/day the Asterisk list goes through...
 

Seems to be around 1200 emails per week, for the one week that I counted
anyway...
   

Last week I was complaining to John, at Digium, about the list lag.  Mark
walked by his office so John voiced my complaint.  Mark then said that the
mailing list does 9 million messages a day.
 

I have got 685 mails in the last 5 days, that works out to 137 per day..

Lets make it 150 per day since the last 5 days includes a Sunday..

900 mails / 150 per day = 6 Subscribers..

Is the Asterisk community anywhere near that big??

I am not disagreeing with Mark, its his server, it just seems that 9 
million per day is a little high..

Later..

___
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

Re: [Asterisk-Users] R2 support

[Daniel Bichara]

CW_ASN wrote:

  
  
  
  
  CW_ASN - Gus wrote:
  
  

  
Ok, it's old and clunky, but in some countries like Brazil, Argentina and
China is the only alternative.

  
  Only alternative??? Why is the only alternative? All mayor carriers in
Argentina and Brasil have PRI signalling, at the same price.

  

Sorry, but you are wrong. I am from Brazil and E1-ISDN is not
avaible all over the country.

Daniel

Maybe, you don't have big carriers
in all country...
 
  


Maybe Telefonica (the same from .ar) is not big enough!





___
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

Re: [Asterisk-Users] New Windows IAX Client

[Samuel Jimenez]

  Steven,

  This is really great!!   Very professional and features comparable to what
one find in major brands softphones.

  I wonder if it is possible/planned..

  - Programmable Directory Number buttons (DNs linked to specific buttons)?
  - A special version capable to run from removable media (USB or FDD) with
no need from setup installation?  This is great for frequent travelers and
for first contact on site demos.
  - video phone feature?


  Regards


  Sam\\\


  - Original Message - 
  From: "Steven Sokol" <[EMAIL PROTECTED]>
  To: <[EMAIL PROTECTED]>;
<[EMAIL PROTECTED]>
  Cc: <[EMAIL PROTECTED]>
  Sent: Wednesday, January 21, 2004 6:01 PM
  Subject: [Asterisk-Users] New Windows IAX Client


  > Announcing a new Windows-based IAX/IAX2 client.  Please download it and
  > give it a try.  Let me know about any bugs, and any missing features.  I
  > have yet to come up with a catchy name for it, so at this point it calls
  > itself IAX Phone.  (Suggestions?  Non-derogatory suggestions,
  > preferably).
  >
  > Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip
  >
  > Reference & Support Page: http://www.sokol-associates.com/IaxPhone.htm
  >
  >
  > Features:
  >
  > - Works correctly for both inbound and outbound calls!
  > - Registers With Multiple Servers
  > - 4 Line Appearances
  > - Direct IAX URL Dialing (user:[EMAIL PROTECTED]/[EMAIL PROTECTED])
  > - 20 Speed Dials
  > - Native IAX Blind Transfer (no more wasted pound key)
  > - Drag/Drop Transfer (right-click & drag a line to a Speed Dial to
  > transfer)
  > - Last number redial
  > - Message Waiting Indicator (Native IAX, no Manager configuration
  > reqired)
  > - Message Waiting Count *
  > - Registration Indicator
  > - Missed Calls Indicator
  > - Mini-Mode for low screen-real-estate usage.
  > - Multiple Audio Output Devices (use speakers for Ring, headset for
  > audio I/O)
  > - Supports IAX/IAX2
  > - Supports DSP Filters: AGC, Echo Cancellation, Denoise
  > - Integrated call timer
  > - Integrated with the Eutectics IPP200 USB handset
  > - Direct dialing for IAXTel users (no need for [EMAIL PROTECTED])
  > - Intercom Calls (auto-answer, use speakers for audio output)**
  >
  > Coming Soon:
  >
  > - Call Log
  > - Phonebook Dialing
  > - Outlook Integration (dialing & screen pops)
  > - Programmable feature buttons (DND, Forward, Direct-To-Voicemail
  > Transfer, etc.)
  > - Documentation (sorry...;-)
  > - Additional Screen formats (toolbar, icon-tray)
  > - Keyboard Shortcuts
  > - User-defined ring-tones (wav, midi, etc.)
  > - Personalized ring-tones (assigned in phonebook)
  > - Enhanced speed dial "dashboard" as DSS (Manager integration) for
  > Operator console
  > - Call rejection/Zapateller
  > - Optional IAX debugging and logging
  > - Auto-answer options (either all calls or by phone-book entry)
  > - Integrated call recording
  >
  > Hopeful Additions:
  >
  > - Simple text messaging
  > - Local conferencing
  > - Screen-saver awareness (auto "Away" when screen saver kicks on)
  >
  >
  > This is the very first release so there will be bugs (as always, use at
  > your own risk).  Please let me know what they are ASAP.  I will
  > hopefully have the next release (with many of the "coming soon"
  > features) out later in the week or over the weekend.
  >
  > Thanks,
  >
  > Steve Sokol
  >
  > * Requires a small change to chan_iax2.c in order to return valid
  > message counts.  See the support page.
  >
  > ** Requires a special macro in extensions.conf that replaces the
  > caller's real number with a special Intercom indicator ID.  See the
  > support page.
  >
  >
  > ___
  > 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

___
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

[Asterisk-Users] Draytek SIP phones are broken

[Jan Janak]

Hello,

if you have a Draytek SIP phone, please check if the phone doesn't flood
your server with SIP REGISTER messages. Draytek phones are broken and
keep sending REGISTER messages after receiving 200 OK (even if expires
value is long enough).

Several such phones are flooding iptel.org public servers these days. If
you have direct contact to Draytek developers, please send it to me.

   thanks, Jan.
___
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

[Asterisk-Users] ETSI PRI ISDN Signalling

[Daniel Bichara]

Hi All,

I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to 
config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN 
Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue?

Daniel

___
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

Re: [Asterisk-Users] asterisk 0.7.1 - mysql

[WipeOut]

Tilghman Lesher wrote:

On Thursday 22 January 2004 08:01, Andrew Thompson wrote:
 

- Original Message -
From: "Dawid Mielnik" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
   

Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL.
Does this new version of * only work through ODBC ? Do I have
connect to MySQL through ODBC now ?
 

MySQL support was moved out to "addons." You don't have to use ODBC
to point to MySQL, but I would say it's probably a good idea. There
is no guarantee that anyone will continue to update the mysql code
now that the license on mysql has changed.
   

There's no guarantee that Asterisk will be maintained either.  There are
lots of people who are interested enough, though, to make sure that it
continues to be maintained.
As far as cdr_addon_mysql.c support, I'm committed to maintaining it
for the forseeable future.  In fact, I've just uploaded a patch to the
bugtracker to add a CLI command to this module:
http://bugs.digium.com/bug_view_page.php?bug_id=902

-Tilghman

 

Tilghman,

I am glad to see I am not the only one who likes using MySQL.. :-)

I was begining to feel that I was alone in the Asterisk/MySQL world.. ;-)

PS I am only kidding...

Later..

___
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

Re: [Asterisk-Users] asterisk 0.7.1 - mysql

[Jonathan Moore]

> > MySQL support was moved out to "addons." You don't have to use ODBC
> > to point to MySQL, but I would say it's probably a good idea. There
> > is no guarantee that anyone will continue to update the mysql code
> > now that the license on mysql has changed.
> 

Can anyone explain what has changed about the MySQL license and when it changed?
I wasn't aware of this.

Visit Winfield Public Schools at http://usd465.com
-
This mail sent through IMP: http://horde.org/imp/
___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[Steven Critchfield]

On Thu, 2004-01-22 at 12:17, Jonathan Moore wrote:
> I am researching the use of White Box Enterprise Linux. Someone else in a
> similar position with a bunch of 7.x boxes created it. He took all the SRPM
> files for REL v3 and removed all the Red Hat logos and trademarks. It is the
> same software as Enterprise but you can freely copy it. They also modded the
> update scripts to work with more generic update sources. The cool thing is the
> system is completely compatible with the REL source errata which Red Hat has
> promised to continue updating for at least five years. They have also setup a
> small system of mirrors to host the update files. It looks very promising. I am
> trying this and Debian to see which will be easier to keep updates for. Info and
> ISO file available at
> 
> http://www.beau.org/~jmorris/linux/whitebox/index.html
> 
> Someone else has a similar project going but it didn't seem to be as far along.

Just to point out that debian is a long matured product that is not
likely to go anywhere due to the large number of people developing it
from many countries.

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
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

[Asterisk-Users] Automatically Logging Out Queue Members

[Eric Wieling]

How do I make a queue member (added with AddQueueMember) automatically
be logged out the queue if they don't pick up?

--Eric

-- 
Go to http://www.digium.com/index.php?menu=documentation and look at
the "Unofficial Links" section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
"Asterisk Resource Pages".

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

___
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

Re: [Asterisk-Users] Using varables in MeetMe?

[Steven Critchfield]

On Thu, 2004-01-22 at 12:06, Christopher Arnold wrote:
> Hi,
> 
> Im trying to enable users to enter a conference number and then do a
> calculation on this and then send them to the conference. Lokk at this
> example:
> 
> exten => s,1,Read(room)
> exten => s,2,SetVar,"${room}=[${room} + 2000]";
> exten => s,3,Meetme($room|pqsd)
> 
> What happens is that the conference gets setup and everything, but the
> conference number is "$room". Not really what i expected...
> 
> Is this a bug of a feature?

Reread the documentation on variables again.

exten => s,1,read(room)
exten => s,2,SetVar(room=[${room} + 2000]
exten => s,3,MeetMe(${room}|pqsd)
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
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

AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[Thomas Haeger]

Has somebody got it work at all ?
I mean data calls (ISDN 64k) through asterisk.

Regards,

Thomas.

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Donnerstag, 22. Januar 2004 19:07
An: [EMAIL PROTECTED]
Betreff: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


Hi,

we tried following scenario:


DTAG (S0) at our office Datacall with AVMFritz (PSTN) ---> Colo
TelesSwitch --> CoLo Asterisk (---> PSTN)

I think, no i know that the Teles Switch can route 64k data calls
here is the Teles Trace:

#08SETUP-->|
15:29:40,378 02 01 78 AE   |
 08 02 03 90 05|
Bearer Caps  04 02 88 90   |
Channel Id   18 03 A1 83 9B|
Calling PN   6C 0C 21 83 33 33 32 38   |
 33 33 34 37 37 38 |
Called PN70 09 C1 36 33 34 39 34   |
 34 34 31  |
   |<--RR
#08
   |   15:29:40,388 02 01 01 7A
   |<--SETUP ACKNOWLEDGE
#08
   |   15:29:40,398 00 01 AE 7A
   |08 02 83 90 0D
   |   Channel Id   18 03 A9 83 9B
   |<--SETUP
#12
   |   15:29:40,408 00 01 2A D4
   |08 02 16 60 05
   |   Bearer Caps  04 02 88 90
   |   Channel Id   18 03 A1 83 88
   |   Calling PN   6C 0C 21 80 33 33 32
38
   |33 33 34 37 37 38
   |   Called PN70 09 81 36 33 34 39
34
   |34 34 31
#08   RR-->|
15:29:40,408 00 01 01 B0   |
#12   RR-->|
15:29:40,418 00 01 01 2C   |
#12SETUP ACKNOWLEDGE-->|
15:29:40,418 02 01 D4 2C   |
 08 02 96 60 0D|
Channel Id   18 03 A9 83 88|
Progress Ind 1E 02 81 82   |
   |<--RR
#12
   |   15:29:40,418 02 01 01 D6
#12SETUP-->|
15:29:40,428 02 01 D6 2C   |
 08 02 1A 21 05|
Bearer Caps  04 03 88 90 A3|
Channel Id   18 03 A1 83 81|
Calling PN   6C 0C 41 81 33 33 32 38   |
 33 33 34 37 37 38 |
Called PN70 0D C1 30 33 33 32 38   |
 33 30 37 37 37 33 33  |
   |<--RELEASE COMPLETE
#12
   |   15:29:40,428 00 01 2C D8
   |08 02 9A 21 5A
   |08 02 80 D8
   |[Incompatible
destinat
   |ion]
#12   RR-->|

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von CW_ASN -
Gus
Gesendet: Donnerstag, 22. Januar 2004 17:24
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


The incoming call request Unrestricted and 64K, and this looks like ok, but
in the SETUP_ACK the called number parameters shows: "Ext: 1  Progress
Description: Called equipment is non-ISDN. (2) ]", like as is not an ISDN
equipment.
In the most of cases, Information transfer rate = to '64 kbit/s', and Info
transfer capability = 'real bw required'.

Are you sure that the equipment attached to * can be used in 64K?

Regards,

Gus

- Original Message -
From: "Thomas Haeger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 12:28 PM
Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


> Hi ,
>
> maybe someone knows what's going wrong...
>
> The incoming data call will not really identified as ISDN 64k/Data
>
> Here my pri debug ouput
>
> < Protocol Discriminator: Q.931 (8)  len=39
> < Call Ref: len= 2 (reference 5635/0x1603) (Originator)
> < Message type: SETUP (5)
> < Bearer Capability (len= 2) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: Unrestricted digital information (8)
> <  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
> (16)
> <  Ext: 0  User information layer 1: Unknown
> (24)
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
> Dchan: 0
>  <   Ext: 1  Coding: 0   Number Specified   Channel
Type:
> 3
> < 

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[JC]

I have been using Mandrake 9.2 and it has been totally stable and haven't
had any problems with installations of asterisk. I stopped using RH9 because
of the upcoming
end of their support.

J.C.

- Original Message - 
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 11:29 AM
Subject: Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released


> On Thursday 22 January 2004 03:08, WipeOut wrote:
> > My biggest problem is that RH has basically dropped me in the "poo"
> > by killing off their free version and stopping support for all the
> > free versions as well.. I have been looking at alternatives but so
> > far nothing is going to fit the bill.. The other distro's are either
> > way off the mark or too difficult to get running in the first place
> > or to difficult to manage in a production enviroment.. also I can't
> > affort $400 for RH Enterprise Linux for each of my test/demo/dev
> > servers.. I guess there are many with the same problem.. :(
>
> Why not subscribe to Progeny?  They offer continuing support for RedHat
> 7.3 installations.
>
> -Tilghman
>
> ___
> 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
>


___
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

RE: [Asterisk-Users] Standalone FXO device

[Chris Albertson]

> > I connect with the Clipcomm device, my DSL gets down and it gets up
> > only when I switch remove the line from the device. I dunno what
> kind
> > of problem it is.

With DSL one line shares voice and data.

_Every_ device plugged into your phone line except your DSL
modem needs an in-line filter.  The filter prevents the non-modem
from either changing the line impedance or putting high frequency
noise on the line, either of which will kill the DSL signal.

So if you plug an FXO device into the same line that also carries
DSL and don't use a filter the FXO device could very well kill
your DSL connection.  But it all depends, some times you can
skip the filters.  I did but then I plugged in one more analog
phone and broke DSL.  I installed a few in-line filters and now the
DSL works better.

If you did place a filter between the clipcom device and the rj11
wall jack then something else is going on

=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
___
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

Re: [Asterisk-Users] Is there a way to # of agents logged into a queue ?

[Bill Hamel]

Hi Chris,

This sounds what I am looking for, many thanks !

Also, I do not see an attachment, the patch that is :)

I dont know if the list strips attachments, perhaps send it to my email address
[EMAIL PROTECTED]

Thanks again,
-bh


Quoting "C. Maj" <[EMAIL PROTECTED]>:
> I attached a patch I've been using to show the # of agents
> (members) and callers on a per queue basis.  It adds a new
> manager command, "AgentQueues".  It returns on the manager
> interface the following for each queue:
> 
> Queue: queuename
> Agents: #
> Callers: #
> 
> There's another manager command, "QueueStatus", that might be
> what your are looking for.  There's also "Queues" but that
> is a PITA to parse.  Fine if you just want to display it in
> a text widget or something.
> 
> --Chris
> 
> 
> -- 
> 
> Chris Maj 
> Pronunciation Guide:  Maj == May
> Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146
> 
> ___
> 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
> 
> -- 
> This message has been scanned for viruses and
> dangerous content by The CCIS.net MailScanner, and is
> believed to be clean.
> 
> 
> -- 
> This message has been scanned for viruses and
> dangerous content by the Bugs.Hamel.Net MailScanner, 
> and appears to be clean.
> 
> 


-- 



This message was sent using IMP, the Internet Messaging Program.

-- 
This message has been scanned for viruses and
dangerous content by the Bugs.Hamel.Net MailScanner, 
and appears to be clean.

___
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

[Asterisk-Users] Problem with flashing FXO callwaiting from FXS

[Chris Hirsch]

Hey all...I can't seem to figure out *exactly* what needs to be done 
when I can't flash over to an incoming callwaiting call on FXO from an 
FXS card. Right now if I get a callwaiting call from the FXO and hit 
flash nothing happens.

I've been over the archives and google and it appears that this is 
possible and I guess it can be done with *0 but I haven't found the 
config on how to glue all this together.

If anybody can help out I'd really appriecate it!

Thanks,
Chris
--
Since light travels faster than sound, isn't that why some people appear 
bright until you hear them speak? -Steven Wright

http://ccicolorado.org
Exceptional Dogs for Exceptional People - Help Out Today!
___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[Greg Boehnlein]

On Thu, 22 Jan 2004, Tilghman Lesher wrote:

> On Thursday 22 January 2004 03:08, WipeOut wrote:
> > My biggest problem is that RH has basically dropped me in the "poo"
> > by killing off their free version and stopping support for all the
> > free versions as well.. I have been looking at alternatives but so
> > far nothing is going to fit the bill.. The other distro's are either
> > way off the mark or too difficult to get running in the first place
> > or to difficult to manage in a production enviroment.. also I can't
> > affort $400 for RH Enterprise Linux for each of my test/demo/dev
> > servers.. I guess there are many with the same problem.. :(
> 
> Why not subscribe to Progeny?  They offer continuing support for RedHat
> 7.3 installations.

That was my point a bit earlier in the thread! ;)

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[Greg Boehnlein]

On Thu, 22 Jan 2004, WipeOut wrote:

> Ken Godee wrote:
> 
> >>> This is great to see.. but why RH7.3 (or RH8 for that matter) since 
> >>> it has already been EOL'ed by RH??
> >>
> >>
> >>
> >> Couple of reasons..
> >>
> >> 1. It is a stable, known quantity that uses solid components and 
> >> closely mirrors the environment that a lot of people develop Asterisk 
> >> on. It isn't going to drastically change, so those wishing to deploy 
> >> it in production may look to RedHat 7.3 as a stable platform for that 
> >> purpose.
> >>
> >
> > I agree, keep up the good work.
> >
> > I personally don't see any reason to upgrade atleast until the
> > 2.6.x kernel is well underway. Maybe that's just me, hell I'm
> > still running a 4.11 Novell server and a SCO Open server that hasn't
> > been touched since y2k upgrades.
> >
> I am guessing your systems are not connected to the internet then.. :)

I am, but I am also intelligent enough to firewall systems and properly 
secure them, no matter what distribution I run.
 
> The problem with running servers based on RH 6.x, 7.x and 8 is that RH 
> is not providing errata (security specifically) updates any more.. If 
> you servers are not connected to the internet then, sure stay with the 
> versions that are working for you, but if you have you server live on 
> the internet for ant reason then this is a big issue..

No it isn't. If you follow best practices for your system, remove all 
unneccessary packages, and properly firewall it, you are at no greater or 
lesser risk than any other version of RedHat.

Take a look at the following:
http://www.nacs.net/~damin/linux-best-practices.pdf

> I realise that many vulnerabilities require local access but I am still 
> not going to take the chance.. I want my servers as safe as possible, 
> and if that means running the latest versions of whatever then thats 
> what I am going to do.. :)

Take a look at the number of exploits that are available for RH 8 and 9, 
and how quickly they are mounting up, and then rethink that statement. 
There are more exploits being targeted at these platforms, in a shorter 
period of time, than 7.3 and the earlier versions.

Personal opinion here, but if you are relying on RedHat to be your 
security provider, you have no business administering a system connected 
to the Internet. Sure, they make it easier, but common sense and a solid 
understanding of the applications and code that your system is based on 
are a hell of a lot more comforting.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



___
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

[Asterisk-Users] Asterisk and gnugk

[bam]

This is quite possibly a daft question, but it is possible to run * and 
gnugk on the same system with gnugk acting as a proxy for netmeeting 
endpoints and feeding everything for PSTN and SIP out through *?

___
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

Re: [Asterisk-Users] Strange Zaptel Modprobe driver failure

[Tilghman Lesher]

On Wednesday 21 January 2004 15:53, Mark Rizzo wrote:
> Hello, my first post to the list.  I have started to install and play
> with Asterisk.  I was following some basic instructions to
> 'jump-start' my system.  I have a TDM400P with one port and a X100P.
>
> I am running the latest CVS versions (from today).
>
> Following the steps I found for basic jump-start I did the following:
> Modprobe zaptel
> Modprobe wcfxo
> Modprobe wcfxs
>
> Edited /etc/zaptel.conf and added the following lines:
> fxsks=1
> fxoks=2
> loadzone=us
> defaultzone=us
>
> I then ran ztcfg -vv and receive a good response.
>
> Asterisk started and running just fine!  YES!
>
> I then rebooted my computer for other reasons.  Now the following
> happens:
> Modprobe zaptel
> (works)
>
> Modprobe wcfxo
> ZT_CHANCONFIG failed on channel 2: No such device or address (6)
> /lib/modules/2.4.20-gentoo-r9/misc/wcfxo.o: post-install wcfxo failed
> /lib/modules/2.4.20-gentoo-r9/misc/wcfxo.o: insmod wcfxo failed
>
> Modprobe wcfxs
> (works)
>
> lsmod shows that both modules are loaded.  Asterisk runs, though I
> have yet to try and use both boards.  If I remove my changes to the
> zaptel.conf file, then run modprobe, then re-add my changes to
> zaptel.conf file I am fine.

The problem is not that the modules aren't loading, but that when the
first module is loaded, the entry in modules.conf attempts to configure
the second channel, for which the driver is not yet loaded.  This is the
error that you see about the second channel.

I have found three possible solutions:

1)  Remove the post-install wcfxo line from /etc/modules.conf.  This is
the easiest solution, but it is not permanent, as everytime you
recompile and reinstall the drivers, the line will be readded.

2)  Create /etc/zaptel.conf.0 and /etc/zaptel.conf.1, the first config
file with only the configuration for your wcfxo, and the second for
all channels.  Before each modprobe, symlink (or copy) the
appropriate file into place, e.g.

modprobe zaptel
ln -sf /etc/zaptel.conf.0 /etc/zaptel.conf
modprobe wcfxo
ln -sf /etc/zaptel.conf.1 /etc/zaptel.conf
modprobe wcfxs

The benefit here is that you don't ever have to bother with this again
until you add more hardware.  The downside is that you need a script to
do this.

3)  Ignore the error.  Upside:  no action is needed.  Downside:  if you
get in the habit of ignoring errors, sometime, you're going to miss
something important.

-Tilghman

___
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

Re: [Asterisk-Users] Mailing List Lag

[Christian Hoffmeyer]

- Original Message - 
From: "Steve Foy" <[EMAIL PROTECTED]>
To: "Asterisk-Users" <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 3:18 AM
Subject: Re: [Asterisk-Users] Mailing List Lag


> On Thu, Jan 22, 2004 at 08:39:09AM +, Steve Foy wrote:
> > I'd be willing to host the list, I guess it just depends on how many
> > emails/day the Asterisk list goes through...
>
> Seems to be around 1200 emails per week, for the one week that I counted
> anyway...

Last week I was complaining to John, at Digium, about the list lag.  Mark
walked by his office so John voiced my complaint.  Mark then said that the
mailing list does 9 million messages a day.

Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL

(iax)  700.859.4508

___
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

Re: [Asterisk-Users] asterisk 0.7.1 - mysql

[Tilghman Lesher]

On Thursday 22 January 2004 08:01, Andrew Thompson wrote:
> - Original Message -
> From: "Dawid Mielnik" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> >
> > Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL.
> > Does this new version of * only work through ODBC ? Do I have
> > connect to MySQL through ODBC now ?
>
> MySQL support was moved out to "addons." You don't have to use ODBC
> to point to MySQL, but I would say it's probably a good idea. There
> is no guarantee that anyone will continue to update the mysql code
> now that the license on mysql has changed.

There's no guarantee that Asterisk will be maintained either.  There are
lots of people who are interested enough, though, to make sure that it
continues to be maintained.

As far as cdr_addon_mysql.c support, I'm committed to maintaining it
for the forseeable future.  In fact, I've just uploaded a patch to the
bugtracker to add a CLI command to this module:

http://bugs.digium.com/bug_view_page.php?bug_id=902

-Tilghman

___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[Jonathan Moore]

I am researching the use of White Box Enterprise Linux. Someone else in a
similar position with a bunch of 7.x boxes created it. He took all the SRPM
files for REL v3 and removed all the Red Hat logos and trademarks. It is the
same software as Enterprise but you can freely copy it. They also modded the
update scripts to work with more generic update sources. The cool thing is the
system is completely compatible with the REL source errata which Red Hat has
promised to continue updating for at least five years. They have also setup a
small system of mirrors to host the update files. It looks very promising. I am
trying this and Debian to see which will be easier to keep updates for. Info and
ISO file available at

http://www.beau.org/~jmorris/linux/whitebox/index.html

Someone else has a similar project going but it didn't seem to be as far along.
-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Tilghman Lesher <[EMAIL PROTECTED]>:

> On Thursday 22 January 2004 03:08, WipeOut wrote:
> > My biggest problem is that RH has basically dropped me in the "poo"
> > by killing off their free version and stopping support for all the
> > free versions as well.. I have been looking at alternatives but so
> > far nothing is going to fit the bill.. The other distro's are either
> > way off the mark or too difficult to get running in the first place
> > or to difficult to manage in a production enviroment.. also I can't
> > affort $400 for RH Enterprise Linux for each of my test/demo/dev
> > servers.. I guess there are many with the same problem.. :(
> 
> Why not subscribe to Progeny?  They offer continuing support for RedHat
> 7.3 installations.
> 
> -Tilghman
> 
> ___
> 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
> 


Visit Winfield Public Schools at http://usd465.com
-
This mail sent through IMP: http://horde.org/imp/
___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[WipeOut]

Tilghman Lesher wrote:

On Thursday 22 January 2004 03:08, WipeOut wrote:
 

My biggest problem is that RH has basically dropped me in the "poo"
by killing off their free version and stopping support for all the
free versions as well.. I have been looking at alternatives but so
far nothing is going to fit the bill.. The other distro's are either
way off the mark or too difficult to get running in the first place
or to difficult to manage in a production enviroment.. also I can't
affort $400 for RH Enterprise Linux for each of my test/demo/dev
servers.. I guess there are many with the same problem.. :(
   

Why not subscribe to Progeny?  They offer continuing support for RedHat
7.3 installations.
-Tilghman

 

Didn't know it existed.. Looks very interesting..

Thanks..

___
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

[Asterisk-Users] Using varables in MeetMe?

[Christopher Arnold]

Hi,

Im trying to enable users to enter a conference number and then do a
calculation on this and then send them to the conference. Lokk at this
example:

exten => s,1,Read(room)
exten => s,2,SetVar,"${room}=[${room} + 2000]";
exten => s,3,Meetme($room|pqsd)

What happens is that the conference gets setup and everything, but the
conference number is "$room". Not really what i expected...

Is this a bug of a feature?


/Chris
___
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

AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[Thomas Haeger]

Hi,

we tried following scenario:


DTAG (S0) at our office Datacall with AVMFritz (PSTN) ---> Colo
TelesSwitch --> CoLo Asterisk (---> PSTN)

I think, no i know that the Teles Switch can route 64k data calls
here is the Teles Trace:

#08SETUP-->|
15:29:40,378 02 01 78 AE   |
 08 02 03 90 05|
Bearer Caps  04 02 88 90   |
Channel Id   18 03 A1 83 9B|
Calling PN   6C 0C 21 83 33 33 32 38   |
 33 33 34 37 37 38 |
Called PN70 09 C1 36 33 34 39 34   |
 34 34 31  |
   |<--RR
#08
   |   15:29:40,388 02 01 01 7A
   |<--SETUP ACKNOWLEDGE
#08
   |   15:29:40,398 00 01 AE 7A
   |08 02 83 90 0D
   |   Channel Id   18 03 A9 83 9B
   |<--SETUP
#12
   |   15:29:40,408 00 01 2A D4
   |08 02 16 60 05
   |   Bearer Caps  04 02 88 90
   |   Channel Id   18 03 A1 83 88
   |   Calling PN   6C 0C 21 80 33 33 32
38
   |33 33 34 37 37 38
   |   Called PN70 09 81 36 33 34 39
34
   |34 34 31
#08   RR-->|
15:29:40,408 00 01 01 B0   |
#12   RR-->|
15:29:40,418 00 01 01 2C   |
#12SETUP ACKNOWLEDGE-->|
15:29:40,418 02 01 D4 2C   |
 08 02 96 60 0D|
Channel Id   18 03 A9 83 88|
Progress Ind 1E 02 81 82   |
   |<--RR
#12
   |   15:29:40,418 02 01 01 D6
#12SETUP-->|
15:29:40,428 02 01 D6 2C   |
 08 02 1A 21 05|
Bearer Caps  04 03 88 90 A3|
Channel Id   18 03 A1 83 81|
Calling PN   6C 0C 41 81 33 33 32 38   |
 33 33 34 37 37 38 |
Called PN70 0D C1 30 33 33 32 38   |
 33 30 37 37 37 33 33  |
   |<--RELEASE COMPLETE
#12
   |   15:29:40,428 00 01 2C D8
   |08 02 9A 21 5A
   |08 02 80 D8
   |[Incompatible
destinat
   |ion]
#12   RR-->|

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von CW_ASN -
Gus
Gesendet: Donnerstag, 22. Januar 2004 17:24
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


The incoming call request Unrestricted and 64K, and this looks like ok, but
in the SETUP_ACK the called number parameters shows: "Ext: 1  Progress
Description: Called equipment is non-ISDN. (2) ]", like as is not an ISDN
equipment.
In the most of cases, Information transfer rate = to '64 kbit/s', and Info
transfer capability = 'real bw required'.

Are you sure that the equipment attached to * can be used in 64K?

Regards,

Gus

- Original Message -
From: "Thomas Haeger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 12:28 PM
Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


> Hi ,
>
> maybe someone knows what's going wrong...
>
> The incoming data call will not really identified as ISDN 64k/Data
>
> Here my pri debug ouput
>
> < Protocol Discriminator: Q.931 (8)  len=39
> < Call Ref: len= 2 (reference 5635/0x1603) (Originator)
> < Message type: SETUP (5)
> < Bearer Capability (len= 2) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: Unrestricted digital information (8)
> <  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
> (16)
> <  Ext: 0  User information layer 1: Unknown
> (24)
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
> Dchan: 0
>  <   Ext: 1  Coding: 0   Number Specified   Channel
Type:
> 3
> <   Ext: 1  Channel: 30 ]
> < Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> <   Presentation: Presentation permitted, user
> number not screened (0) '3328334778' ]
> < Called Number (len=11) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
> ISDN/Tele

[Asterisk-Users] Integrating * with a legacy Nec NEAX 1400

[Eric W. Hatch]

Title: Message



 
 
I have searched the 
lists and the wiki for some info regarding Integrating * with a legacy Nec NEAX 
1400. 
 
I am trying to build 
a test PBX and eventually integrate it with the current PBX here at my 
office. Do further testing and if all goes well replace the neax with *. I have 
a T1 line coming in from the Phone company, but as far as I can tell the most 
expensive part of this is going to be replacing the telephones. If * could talk 
to my existing phones (Dterm Series II by NEC) I would be able to break the cost 
down into modules eventually replacing the phones for VOIP models. I have about 
70 multiline terminals, 20 single line, and 3 conference rooms. 

 
What is the best way 
to start off for this project?
Are there any 
companies that have done something similar to this that I could talk to about 
their experiences with * ? 
Is there anyone else 
in the Wisconsin area that has an * system running that I could talk to about 
their experiences with * ?
 
I am very interested 
in hearing any comments and suggestions you may have about this. Thank 
you.

Re: [Asterisk-Users] Re: Digium X100P for $43

[John Baker]

This sounds like a good business.  Get a no-name X100P-alike that retails
everywhere else for about $15 and then put it on e-bay for $43 to fools that
don't know any better.

I love America.

John
- Original Message - 
From: "Alfred R. Nurnberger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 9:59 AM
Subject: RE: [Asterisk-Users] Re: Digium X100P for $43


>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of
> [EMAIL PROTECTED]
> Sent: Thursday, January 22, 2004 3:48 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Re: Digium X100P for $43
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Sean Cheesman
> > Sent: Wednesday, January 21, 2004 11:04 PM
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] Re: Digium X100P for $43
> >
> >
> > for the record, mine has the same fcc id number as the
> > Digiums.  Is this typical for copied hardware, or is there
> > something a little fishy going on here?
>
> --
--
> -
> I looked at the site of www.digitnetworks.com today.
>
> The thing I noticed was that nowhere on the site they listed the real
> (registered) company name or mention their address.
> One line has a (801) phone number listed otherwise only email addresses.
>
> Whois revealed the following:
>
> Registrant:
> Domains by Proxy, Inc.
> 15111 N Hayden Rd., Suite 160
> PMB353
> Scottsdale, Arizona 85260
> United States
>
> Again no name, no address, sucessful way of hiding their identity.
>
> I stick with Digium, I know who they are, where they are and what I am
> getting,
> and I am supporting the developmnet of *.
>
> Alfred.
>
> ___
> 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
>

___
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

Re: [Asterisk-Users] Gsm + snom phones

[Detlef Wengorz]

Matteo Brancaleoni wrote:
> 
> Hi.
> 
> > About a month ago I made a test with snom200b.
> > At least then it worked ok with *.
> > At the moment  I'm using mainly g711a. So, there is always a possibility
> > something
> 
> but you also tested gsm ?

It works for me with gsm :-)
6 snom 200 and one snom 105 with gsm over german t-dsl (128 Kbit)

sound is really good :-)

> 
> Greets,Matteo.
> 
> --
> Matteo Brancaleoni
> Espia System Administrator
> Email : [EMAIL PROTECTED]
> Web   : http://www.espia.it
> Phone : +39 02 70633354  - ext 201
> IAX(2): [EMAIL PROTECTED] - ext 201
> Iaxtel: 1-700-56-62458   - ext 201
> 
> ___
> 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

-- 
Best regards 
Detlef Wengorz <[EMAIL PROTECTED]>
Detlef Wengorz <[EMAIL PROTECTED]>
___
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

RE: [Asterisk-Users] Standalone FXO device

[Senad Jordanovic]

Kannaiyan Natesan wrote:
> SJ,
> 
> I'm also dealing with Andrew, they were good at telling you
> stories but nothing professional with the product. 
> 
> I registered with fwd and started dialling 14551 my fauvorite
> where i get clear voice. It gave me with completely noisy sound,
> I tried to reduce and increase the gain, but nothing works. 
> 
> I primarily want to share my DSL connected PSTN line to other
> members, so other members can use my PSTN minutes for free. But when
> I connect with the Clipcomm device, my DSL gets down and it gets up
> only when I switch remove the line from the device. I dunno what kind
> of problem it is.
> 
>  I left with that, Next I want to try to other networks by
>  connecting ATA 186 FXS port to it. It works sometimes and just
> holds the line without hanging it up. I need to switch it off to get
> the line hooked on. Something very strange.  
> 
>  The overall performance of the devices just sucks,
> 
>   I use the model CG-101E.
> 
>   If you need the device, I can ship you the one which I have got
> with me, since I'm no more interested in having it. 
> 
>   I don't mind paying higher, but I'm looking for a quality
> device. If you can suggest anything, please share it to all in the
> list.  
> 
> Kannaiyan

Hmm..

Very odd and strange problems. Have you pointed these problems to
clipcomm people?
Well, I was/am looking for a device with PSTN FXO backup. 
www.dlink.com does one like that, but is way too expensive.

Ta
SJ

___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[Tilghman Lesher]

On Thursday 22 January 2004 03:08, WipeOut wrote:
> My biggest problem is that RH has basically dropped me in the "poo"
> by killing off their free version and stopping support for all the
> free versions as well.. I have been looking at alternatives but so
> far nothing is going to fit the bill.. The other distro's are either
> way off the mark or too difficult to get running in the first place
> or to difficult to manage in a production enviroment.. also I can't
> affort $400 for RH Enterprise Linux for each of my test/demo/dev
> servers.. I guess there are many with the same problem.. :(

Why not subscribe to Progeny?  They offer continuing support for RedHat
7.3 installations.

-Tilghman

___
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

Re: [Asterisk-Users] need help configuring IAX to make outbound calls through a remote server

[C. Maj]

On Wed, 21 Jan 2004, Paul Mahler waxed:

> I am trying to make outbound calls from my Asterisk client through a remote
> Asterisk server with IAX. 
> 
> In iax.conf on both sides 
> 
> [dar]
> context=trusted
> secret=xx
> type=friend
> host=192.168.1.1

I'm not going to try and fix all of this, but if you've got
the same hostname on both hosts, one of them doesn't know
about the other.  You need to set the host differently on
each of the hosts.  Ie, on 192.168.1.1, you need to set
"host=192.168.1.2" and on 192.168.1.2, you need to set
"host=192.168.1.1".

> in extensions.conf  at the client making the call
> 
> Exten=_1NXXNXX,1,Dial(IAX2/dar:[EMAIL PROTECTED]/)
> 
> What goes in extensions.conf at the remote server? What is needed for the
> remote server to accept the call from my client, figure out the dialed
> number and then dial it outbound on some line? 

You'll need to have a "trusted" context in each, for
starters.  But there's a lot more dialplan work you'll need
to do, depending on where your outbound lines are, what
numbers they can dial without incurring toll charges, etc.

--Chris


-- 

Chris Maj 
Pronunciation Guide:  Maj == May
Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146

___
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

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[CW_ASN - Gus]

The incoming call request Unrestricted and 64K, and this looks like ok, but
in the SETUP_ACK the called number parameters shows: "Ext: 1  Progress
Description: Called equipment is non-ISDN. (2) ]", like as is not an ISDN
equipment.
In the most of cases, Information transfer rate = to '64 kbit/s', and Info
transfer capability = 'real bw required'.

Are you sure that the equipment attached to * can be used in 64K?

Regards,

Gus

- Original Message -
From: "Thomas Haeger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 22, 2004 12:28 PM
Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


> Hi ,
>
> maybe someone knows what's going wrong...
>
> The incoming data call will not really identified as ISDN 64k/Data
>
> Here my pri debug ouput
>
> < Protocol Discriminator: Q.931 (8)  len=39
> < Call Ref: len= 2 (reference 5635/0x1603) (Originator)
> < Message type: SETUP (5)
> < Bearer Capability (len= 2) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: Unrestricted digital information (8)
> <  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
> (16)
> <  Ext: 0  User information layer 1: Unknown
> (24)
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
> Dchan: 0
>  <   Ext: 1  Coding: 0   Number Specified   Channel
Type:
> 3
> <   Ext: 1  Channel: 30 ]
> < Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> <   Presentation: Presentation permitted, user
> number not screened (0) '3328334778' ]
> < Called Number (len=11) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '63494441' ]
> -- Making new call for cr 5635
> -- Processing Q.931 Call Setup
> -- Processing IE 4 (Bearer Capability)
> -- Processing IE 24 (Channel Identification)
> -- Processing IE 108 (Calling Party Number)
> -- Processing IE 112 (Called Party Number)
> > Protocol Discriminator: Q.931 (8)  len=14
> > Call Ref: len= 2 (reference 38403/0x9603) (Terminator)
> > Message type: SETUP ACKNOWLEDGE (13)
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
> Dchan: 0
> >ChanSel: Reserved
> >   Ext: 1  Coding: 0   Number Specified   Channel
Type:
> 3
> >   Ext: 1  Channel: 30 ]
> > Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0)
0:
> 0   Location: Private network serving the local user (1)
> >   Ext: 1  Progress Description: Called
> equipment is non-ISDN. (2) ]
> -- Accepting call from '3328334778' to '63494441' on channel 30, span
2
> -- Executing GotoIf("Zap/61-1", "0?50:100") in new stack
> -- Goto (pri2,63494441,100)
> -- Executing Dial("Zap/61-1", "Zap/g2/033283077733SPEECH") in new
stack
> -- Making new call for cr 39439
> > Protocol Discriminator: Q.931 (8)  len=50
> > Call Ref: len= 2 (reference 6671/0x1A0F) (Originator)
> > Message type: SETUP (5)
> > Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: Speech (0)
> >  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
> (16)
> >  Ext: 1  User information layer 1: A-Law
(35)
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
> Dchan: 0
> >ChanSel: Reserved
> >   Ext: 1  Coding: 0   Number Specified   Channel
Type:
> 3
> >   Ext: 1  Channel: 1 ]
> > Calling Number (len=14) [ Ext: 0  TON: Subscriber Number (4)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> >   Presentation: Presentation permitted, user
> number passed network screening (1) '3328334778' ]
> > Called Number (len=21) [ Ext: 1  TON: Subscriber Number (4)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '033283077733SPEECH' ]
> -- Called g2/033283077733SPEECH
> < Protocol Discriminator: Q.931 (8)  len=10
> < Call Ref: len= 2 (reference 39439/0x9A0F) (Terminator)
>
> < Message type: SETUP ACKNOWLEDGE (13)
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
> Dchan: 0
>  <   Ext: 1  Coding: 0   Number Specified   Channel
Type:
> 3
> <   Ext: 1  Channel: 1 ]
> -- Processing IE 24 (Channel Identification)
> beroasterisk*CLI>
> < Protocol Discriminator: Q.931 (8)  len=9
> < Call Ref: len= 2 (reference 5635/0x1603) (Originator)
> < Message type: DISCONNECT (69)
> < Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location:
> User (0)
> <  Ext: 1  Cause: Normal Clearing (16), class = Normal
Event
> (1) ]
> -- Processing IE 8 (Cause)
> -- Channel 30, span 2 got hangup
> NEW_HANGUP DEBUG: Calling q931_hangup,

RE: [Asterisk-Users] Call Queue with no agents -> Congestion or voicebox instead of MOH?

[B. J. Bomar]

Do not define any members in the queues.conf.  Instead have them login to
the queue using the AddQueueMember application.  If there is no one logged
into the queue when a call comes in, it will go to the priority in the
context.  Hope this helps.

B. J.





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jan Baumann
Sent: Thursday, January 22, 2004 5:05
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call Queue with no agents -> Congestion or
voicebox instead of MOH?



Hi all,

I have successfully set up a call queue with agents and agentCallbackLogin.

Works fine, but if no agent is logged in incoming callers get
music-on-hold forever (or until some timeout).

Is it possible to play congestion tone without answering the call (and
thus causing costs to PSTN callers) or send them to unvailable-mailbox
directly to leave a message if no agents are logged into the queue?

Thanks and best regards,
Jan



___
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

___
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

Re: [Asterisk-Users] Is there a way to # of agents logged into a queue ?

[C. Maj]

On Wed, 21 Jan 2004, Bill Hamel waxed:

> Hi,
> 
> Looking around I can't seem to find a way to show the number of agents currently
> logged into a queue and if possible who they are. Is there a way to do this ?
> 
> Thanks
> -b

I attached a patch I've been using to show the # of agents
(members) and callers on a per queue basis.  It adds a new
manager command, "AgentQueues".  It returns on the manager
interface the following for each queue:

Queue: queuename
Agents: #
Callers: #

There's another manager command, "QueueStatus", that might be
what your are looking for.  There's also "Queues" but that
is a PITA to parse.  Fine if you just want to display it in
a text widget or something.

--Chris


-- 

Chris Maj 
Pronunciation Guide:  Maj == May
Fingerprint: 43D6 799C F6CF F920 6623  DC85 C8A3 CFFE F0DE C146

___
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

Re: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[Steven Critchfield]

What hardware is on the other side of the call(initiating)? Is it set up
to send as voice to avoid data call costs? 

I remember at one point that was a neat trick to keep the telco from
charging their data premium, the data would be passed over the circuit
as voice. 

I think it had to do with the PSCs trying to keep ISDN in the US from
being metered by the minute, but giving ground on data calls.

On Thu, 2004-01-22 at 09:28, Thomas Haeger wrote:
> Hi ,
> 
> maybe someone knows what's going wrong...
> 
> The incoming data call will not really identified as ISDN 64k/Data

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
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

Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released

[WipeOut]

Ken Godee wrote:

This is great to see.. but why RH7.3 (or RH8 for that matter) since 
it has already been EOL'ed by RH??


Couple of reasons..

1. It is a stable, known quantity that uses solid components and 
closely mirrors the environment that a lot of people develop Asterisk 
on. It isn't going to drastically change, so those wishing to deploy 
it in production may look to RedHat 7.3 as a stable platform for that 
purpose.

I agree, keep up the good work.

I personally don't see any reason to upgrade atleast until the
2.6.x kernel is well underway. Maybe that's just me, hell I'm
still running a 4.11 Novell server and a SCO Open server that hasn't
been touched since y2k upgrades.
I am guessing your systems are not connected to the internet then.. :)

The problem with running servers based on RH 6.x, 7.x and 8 is that RH 
is not providing errata (security specifically) updates any more.. If 
you servers are not connected to the internet then, sure stay with the 
versions that are working for you, but if you have you server live on 
the internet for ant reason then this is a big issue..

I realise that many vulnerabilities require local access but I am still 
not going to take the chance.. I want my servers as safe as possible, 
and if that means running the latest versions of whatever then thats 
what I am going to do.. :)

Later..

___
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

RE: [Asterisk-Users] OT: Canada's Primus introduces SIPlocalservice

[Asterisk Users]

Two different companies with two different platforms.

Primus U.S. uses a SIP based service. Primus Canada's new service, Talk
Broadband, is pure MGCP.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Liu
Sent: Wednesday, January 21, 2004 6:36 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] OT: Canada's Primus introduces
SIPlocalservice

I am sure Primus has a SIP platform because we have played with it.  We
managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2
hard phones.  Their PC-Phone app is also a SIP soft phone.  If you are
registering to sip.iprimus.net then it is definitely their SIP platyform
not MGCP.

David

>>> [EMAIL PROTECTED] 1/21/2004 6:39:34 AM >>>

I'm not sure Primus uses SIP. I think it's MGCP.

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of David Liu
Sent: Tuesday, January 20, 2004 9:16 PM
To: [EMAIL PROTECTED] 
Subject: Re: [Asterisk-Users] OT: Canada's Primus introduces SIP
localservice

Hey Colin,

Do let me know if Primus' SIP service can work with Asterisk.  We
tried
setting it up like how you would for iconnecthere   However, we even
failed to register in the first place!  (Of course password and
username
are correct).

Anyone else on the list successfully used Primus' SIP with Asterisk?

David

>>> [EMAIL PROTECTED] 1/20/2004 12:25:50 PM >>>
Primus in Canada has launched a SIP-based service to replace your
business
and residential POTS lines with a VoIP version. It's called
TalkBroadband
and it looks killer:

http://www.primus.ca/en/residential/talkbroadband/index.html 

Basic service for $20 Cdn a month!!

Local number portability!!

Cheapo Primus LD rates!!

They don't care where geographically you plug it in!!

When you sign up, they ship you this Dlink puppy for free:

ftp://ftp10.dlink.com/pdfs/products/DVG-1120/DVG-1120_ds.pdf 

It has 2 FXS ports + ethernet + POTS backup port

My order's in already, I'll be pleased to tell Telus where to put
their
"value pricing" once I get it installed. If anyone in Canada wants to
know
my experiences with it, email me off-list next month.
___
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 
___
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 


___
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
___
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


___
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

Re: [Asterisk-Users] Background Noise

[Jonathan Biggs]

Just to add some info from recent experience. May
help,
May not   1 X100P  2 X 4 port TDM400P

Had to hook a dial-in palm PDA base for a custom
software implementation to my * system and have the
modem dial out and work properly.  Phone connects to
second port on PDA Base

Experienced very bad "electrical" type noise on the
line, hum, buzz, fad in and out. would come and go.

Switched ports, wires,  rxgain and txgain changes,
phone changes, nothing helped

the PDA base is also connected via serial port (or USB
did not matter) to desktop computer for sync purposes

Through trial and error.  Found noise coming from
connection to desk top computer. On, Off did not
matter.

Resolution.  The power strip surge protector I was
using on the desktop computer has two modes for noise
filtration built in.  75 Hertz (or something, don;t
remember) and 50 Hertz.  I moved the plug for the
desktop from the 75 side to the 50 side,  All noise on
line now gone.

Not sure if this helps, constant noise on all sides
may be power and noise filtration related...





--- "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
wrote:
> Hi All
> i have some background noise problem with * and a
> diva srv 4bri + chan_capi 0.3.0 + X-Ten PRO on my
> pc.
> Both in incoming and outgoing call have a background
> noise.
> 
> there is some tuning to do?
> where can i find documentation about capi.conf?
> which is the best codec for sip (ulaw, alaw,
> gsm...)?
> 
> mark
> 
> ___
> 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


__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
___
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

RE: [Asterisk-Users] Re: Digium X100P for $43

[Alfred R. Nurnberger]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 3:48 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Re: Digium X100P for $43


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Sean Cheesman
> Sent: Wednesday, January 21, 2004 11:04 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Re: Digium X100P for $43
>
>
> for the record, mine has the same fcc id number as the
> Digiums.  Is this typical for copied hardware, or is there
> something a little fishy going on here?


-
I looked at the site of www.digitnetworks.com today.

The thing I noticed was that nowhere on the site they listed the real
(registered) company name or mention their address.
One line has a (801) phone number listed otherwise only email addresses.

Whois revealed the following:

Registrant:
Domains by Proxy, Inc.
15111 N Hayden Rd., Suite 160
PMB353
Scottsdale, Arizona 85260
United States

Again no name, no address, sucessful way of hiding their identity.

I stick with Digium, I know who they are, where they are and what I am
getting,
and I am supporting the developmnet of *.

Alfred.

___
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

Re: [Asterisk-Users] Gsm + snom phones

[Pertti Pikkarainen]

Yes, absolutely.
sorry,  I was unclear ..
-- Pertti

Matteo Brancaleoni wrote:

Hi.

 

About a month ago I made a test with snom200b.
At least then it worked ok with *.
At the moment  I'm using mainly g711a. So, there is always a possibility 
something
   

but you also tested gsm ?

Greets,Matteo.

 

___
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

RE: [Asterisk-Users] OT: Canada's Primus introduces SIP localserv ice

[Colin Anderson]

If you look at the specs on the Dlink box that Primus gives you, you will
see that it is SIP.



I am sure Primus has a SIP platform because we have played with it.  We
managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2
hard phones.  Their PC-Phone app is also a SIP soft phone.  If you are
registering to sip.iprimus.net then it is definitely their SIP platyform
not MGCP.

David

>>> [EMAIL PROTECTED] 1/21/2004 6:39:34 AM >>>

I'm not sure Primus uses SIP. I think it's MGCP.

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of David Liu
Sent: Tuesday, January 20, 2004 9:16 PM
To: [EMAIL PROTECTED] 
Subject: Re: [Asterisk-Users] OT: Canada's Primus introduces SIP
localservice

Hey Colin,

Do let me know if Primus' SIP service can work with Asterisk.  We
tried
setting it up like how you would for iconnecthere   However, we even
failed to register in the first place!  (Of course password and
username
are correct).

Anyone else on the list successfully used Primus' SIP with Asterisk?

David

>>> [EMAIL PROTECTED] 1/20/2004 12:25:50 PM >>>
Primus in Canada has launched a SIP-based service to replace your
business
and residential POTS lines with a VoIP version. It's called
TalkBroadband
and it looks killer:

http://www.primus.ca/en/residential/talkbroadband/index.html 

Basic service for $20 Cdn a month!!

Local number portability!!

Cheapo Primus LD rates!!

They don't care where geographically you plug it in!!

When you sign up, they ship you this Dlink puppy for free:

ftp://ftp10.dlink.com/pdfs/products/DVG-1120/DVG-1120_ds.pdf 

It has 2 FXS ports + ethernet + POTS backup port

My order's in already, I'll be pleased to tell Telus where to put
their
"value pricing" once I get it installed. If anyone in Canada wants to
know
my experiences with it, email me off-list next month.
___
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 
___
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 


___
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
___
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
___
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

[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2588 - 11 msgs

[George Bean]

Message: 5
To: [EMAIL PROTECTED]
From: Doug Meredith <[EMAIL PROTECTED]>
Date: Wed, 21 Jan 2004 20:05:19 -0400
Organization: Skyridge Systems Inc.
Subject: [Asterisk-Users] Re: What technology could my phone company be
using?
Reply-To: [EMAIL PROTECTED]

>>Mark Hazlewood <[EMAIL PROTECTED]> wrote:

>>Sounds like Centrex services, we had it from Telus in Alberta a few
years
>>ago.

>I believe this is used for Centrex.  I thought Centrex was basically a
>CO-hosted PBX.  Is it also a local-loop technology?  Are there PCI
>cards or SIP gateway boxes available?

You can think of Centrex as a virtual PBX residing within a LEC or CLEC
switch. It gives you most of the functionality of an on premise PBX
delivered over local loops from the LEC or CLEC switch. This allows
three or four digit dialing between extensions and no charges incurred
for calls unless the user dials 9 for an outside line. 

Centrex can use POTS or ISDN desksets or a mixture of both. For
instance, individual users can have POTS desksets but a receptionist
could have an ISDN deskset to provide programmable keys for multiline
answer and transfer. ADSI desksets can also be used to provide this
functionality.

I assume you are questioning the availability of Centrex compatible
hardware for a system that doesn't employ Asterisk, as Centrex would be
redundant when connected to Asterisk. Any voice capable PCI modem card
can be used to terminate a Centrex POTS loop. ADSI capability may exist,
as a "Winmodem" DSP could be programmed to handle voice and ADSI, but I
haven't seen any applicable drivers/software. ISDN Centrex loops can be
terminated on a PCI ISDN card and software is available to provide voice
functionality. If you want a SIP gateway, you might as well use Asterisk
and non-Centrex loops. There is no sense in duplicating PBX
functionality and paying the monthly Centrex charges.

Regards,
George Bean
Puwaba Technologies




___
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