Javier Rios
2003-Nov-07 06:41 UTC
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #1808 - 13 msgs archives gsm of asterisk ???
Hello.
The procedure so that it works you can find in:
http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris
k
a the files .wav
chmod 755 file.wav
sox file.wav -r 8000 file.gsm resample -ql
chmod 755 file.gsm
in extensions.conf
xxxx=> xxx,x,playback(file)
Ing Javier Rios
Ing de Proyectos
04167285748
212 2637246 /2637187
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
asterisk-users-request@lists.digium.com
Sent: jueves, 06 de noviembre de 2003 12:10
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users digest, Vol 1 #1808 - 13 msgs
Send Asterisk-Users mailing list submissions to
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When replying, please edit your Subject line so it is more specific
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Today's Topics:
1. Re: a bit frightened, guys (Andrew Kohlsmith)
2. Re: Using Asterisk as a VOIP gateway (Alejandro Ruiz)
3. Re: asterisk bandwidth management (Andrew Kohlsmith)
4. Re: Best or any VoIP provider that works with *? (Andrew
Kohlsmith)
5. Re: ISDN PBX + IVR + Voicemail Configuration -
Sanity Check ... (Klaus-Peter Junghanns)
6. RE: archives gsm of asterisk ??? (Shoval Tom)
7. Re: Red Alarm (Andrew Kohlsmith)
8. Re: IAX/SIP Client (Dan)
9. Re: USB handsets/headsets?? (Dan)
10. Re: Anyone using * in a live production environment? (Andrew
Kohlsmith)
11. Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users] Setcontext
based on CID...) (Chris Hirsch)
--__--__--
Message: 1
From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com>
Organization: Benshaw Canada
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] a bit frightened, guys
Date: Thu, 6 Nov 2003 10:34:49 -0500
Reply-To: asterisk-users@lists.digium.com
> But isn't it likely that many people call 911 simultaneously in case
of> an emergency? Maybe it's not a corner case.
Not here, anyway... Small company. :-)
"Someone else is already calling 911. If you wish to continue with your
911
call, please press 1. Otherwise, hang up and calm down." :-)
Regards,
Andrew
--__--__--
Message: 2
From: "Alejandro Ruiz" <aruiz@sputnik-ar.com.ar>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] Using Asterisk as a VOIP gateway
Date: Thu, 6 Nov 2003 12:40:40 -0300
Reply-To: asterisk-users@lists.digium.com
Hi,
I've done somthing like that with 2 X100p.
basically you connect the both end to any extension of the pbx (fxs
port).
when you dail that extension, the first machine will answer and ask for
the
extension number of the other end.
So what you actually have, is a local extension tha works as if you
picked
up an extension on the other end.
I hope this help...
----- Original Message -----
From: "Shoval Tom" <shoval@softov.co.il>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, November 05, 2003 6:12 PM
Subject: RE: [Asterisk-Users] Using Asterisk as a VOIP gateway
> How is it not economical?
> I already have the PBXs on both sides.
> If I switch to * I'll need to get a channel bank
>
> Am I wrong?
>
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
hkirrc.patrick> Sent: Wednesday, November 05, 2003 8:36 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Using Asterisk as a VOIP gateway
>
> yes you can but may not be all that economical though.
> on the other hand, if you can replace or do away with
> at least one of the pbx with * at either end,
> i think you'll be ahead of the game :-)
>
>
> Shoval Tomer wrote:
>
> > Is it possible to use * as a VOIP gateway?
> >
> > Can I connect asterisk to one of the trunks on my current PBX and on
> > the other side of the world connect another * to the trunk of
another> > regular PBX - is it possible to transfer calls from here to there?
> >
> > I guess I'll need one port FXO card for each asterisk, but I
can't
> > figure how to configure the thing.
> >
> > I know I'll need to configure the regular PBX to forward certain
calls> > to the lines connected to asterisk (by prefix, or just have everyone
> > dial 8 and get a line)
> >
> > Does this scenario make sense to anyone? Or am I barking up the
wrong> > tree?
> >
> > Shoval Tomer , MCSE
> >
> > IT Manager
> >
> > Softov Advanced System Ltd.
> >
> > Email: shoval@softov.co.il <mailto:shoval@softov.co.il>
> >
> > Mobile : 972-55-229220
> >
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 3
From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com>
Organization: Benshaw Canada
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] asterisk bandwidth management
Date: Thu, 6 Nov 2003 10:37:24 -0500
Reply-To: asterisk-users@lists.digium.com
> i am using iLBC codec and IAX.. how can i view the
> bandwidth utilization for this in linux.
I run RRD to gather bytes transferred from all my switch ports. You
could
do something similar with it and use ifconfig output or even iptables
counter output. Works _very_ well and is a breeze to set up. There are
many configuration examples and lots of documentation on using RRD with
SNMP and Linux, just google for them.
Regards,
Andrew
--__--__--
Message: 4
From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com>
Organization: Benshaw Canada
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Best or any VoIP provider that works with
*?
Date: Thu, 6 Nov 2003 10:41:18 -0500
Reply-To: asterisk-users@lists.digium.com
> Suggestions on a VoIP provider that works with *?
I am _very_ happy with voicepulse. connect.voicepulse.com. they don't
treat you like a newbie, it's all configured and billed online and their
email support is very fast and friendly. Rates ain't bad, either. :-)
Oh yes, and they support IAX2 trunking and ILBC, and you can download
the
current rates and your call history in machine-readable format without
resorting to screen-scraping. :-)
> The thought of unlimited nationwide calling is of big interest to me
and> others I am sure and I would like to know how others are handling it
on> their end.
They do not have an unlimited plan that I am aware of. That's fine
though;
unlimited long distance is not an economically viable way to run a
business, and I would like voicepulse to hang around for a good long
while.
Regards,
Andrew
--__--__--
Message: 5
Subject: Re: [Asterisk-Users] ISDN PBX + IVR + Voicemail Configuration -
Sanity Check ...
From: Klaus-Peter Junghanns <kpj@junghanns.net>
To: asterisk-users@lists.digium.com
Date: 06 Nov 2003 16:34:51 +0100
Reply-To: asterisk-users@lists.digium.com
Hi Hans,
Am Don, 2003-11-06 um 15.58 schrieb Vledder, Hans:> P.S. He who comes up with clean internal ISDN bus (point to
multi-point)> support for Asterisk, based on CologneChip based equipment receives an
18"> large Dutch cheese in the mail, right after I've wiped away my tears
of> happiness !
>
I am currently working on a zaptel driver for the hfc-s pci a based ISDN
cards and the modifications to make libpri work with BRIs. And of course
NT mode will be supported. We will also have a 4 port BRI card available
in mid/late november that works in TE and NT mode (onboard termination).
Check out www.junghanns.net/asterisk/ during the next week.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon: +49 30 79705392
fax: +49 30 79705391
iaxtel: 1-700-157-8753
email: kpj@junghanns.net
http://www.junghanns.net/asterisk
P.S. i have no idea where i should put the 18" cheese ;-) if it was
19"
i could bring it into the colo...
--__--__--
Message: 6
From: "Shoval Tom" <shoval@softov.co.il>
To: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] archives gsm of asterisk ???
Date: Thu, 6 Nov 2003 17:40:38 +0300
Reply-To: asterisk-users@lists.digium.com
Guys, it still not working.
Go here
http://www.checkdns.net/quickcheck.aspx?domain=voip-info.org&detailed=1
And see that it returns errors.
PLEASE help.
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of WipeOut
Sent: Thursday, November 06, 2003 2:14 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???
Shoval Tom wrote:
>Setting it in hosts doesn't do me any good.
>
>Trying to surf to http:// 64.65.102.50 gets me to apache test page.
>Trying to surf to http:// 64.65.102.50/tiki-index.php?page=Asterisk
>Get a 404 page doesn't exist.
>
>
>
>
Its most likely on a name based virtual server.. edit your hosts file on
your system and put somthing like..
64.65.102.50 www.voip-info.org
Then in your browser just goto http://www.voip-info.org
Later..
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 7
From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com>
Organization: Benshaw Canada
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Red Alarm
Date: Thu, 6 Nov 2003 10:49:21 -0500
Reply-To: asterisk-users@lists.digium.com
> An E1 can be a long way from the box with the right cable. However
many> people use the wrong cable. Using a LAN cable for an E1 often gives
> errors if the cable is more than just a few metres long. Although the
> plugs look the same, the twisted pairs should be grouped differently
in> an E1 cable, and it really makes a difference. If the drop cable is
only> a couple of metres long, a LAN cable is usually adequate. This is also
> true for T1s.
Actually that's not entirely true.
standard 568A/B wired cable does not split pairs for ethernet or DSX1
wiring. The problem is that DSX1 uses pins (1,2),(4,5) and ethernet
(1,2),
(3,6) (parenthesis show pairing). DSX1 must have the (1,2) and (4,5)
pairs swapped to match the TX to the RX at each end, whereas normal
ethernet does not, as the switch is cross-wired. Using an ethernet
crossover cable does not help since it is swapping (1,2) and (3,6), not
(1,2) and (4,5).
The problem with using CAT5 for long telco runs is that the impedance is
wrong at the line clock rate (~1MHz). IIRC the impedance for telco is
specified at 600 ohms @ 1MHz, whereas for CAT5 the impedance is actually
specified at around 100MHz, where the ethernet line rate is. You can
get
away with it so long as the impedance is right, but unless you've got
the
data sheets you're playing guessing games.
Regards,
Andrew
--__--__--
Message: 8
From: "Dan" <dtoma@fx.ro>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] IAX/SIP Client
Date: Thu, 6 Nov 2003 17:53:23 +0200
Organization: Personal account
Reply-To: asterisk-users@lists.digium.com
Hi Ricky,
----- Original Message -----
From: "Asterisk" <thisemailaddressisbogus@risehigh.com>
To: <asterisk-users@lists.digium.com>
Sent: Thursday, November 06, 2003 4:54 PM
Subject: RE: [Asterisk-Users] IAX/SIP Client
>
> >DIAX will ve available as an Active X too which can be integrated in
a> web
> >page, but in a future release.
>
> This is great. How close are you on this Dan? At this time, I cann't
> think of a better application for IAX than DIAX. It really opens up
IAX> to general public.
I must first pass two important stepts:
- IAX2 support (as all further development will be based on this)
- cleaning up as much as possible the bugs from the executable version
(for
this I really need the help of all the interested users).
Then, if there is a real request for that, ActiveX version can be
next..;-)
Best regards,
Dan
--__--__--
Message: 9
From: "Dan" <dtoma@fx.ro>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] USB handsets/headsets??
Date: Thu, 6 Nov 2003 17:55:13 +0200
Organization: Personal account
Reply-To: asterisk-users@lists.digium.com
Hi,
----- Original Message -----
From: "Roy Sigurd Karlsbakk" <roy@karlsbakk.net>
To: "Asterisk Users" <asterisk-users@lists.digium.com>
Sent: Thursday, November 06, 2003 5:26 PM
Subject: Re: [Asterisk-Users] USB handsets/headsets??
> see attached lsusb file for usb out.
> this is a linux box. not windoze. and I can't use windoze for this.
....then you can try to use them for the Asterisk Console....
Sorry that I cannot help you further...
Best regards,
Dan
--__--__--
Message: 10
From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com>
Organization: Benshaw Canada
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Anyone using * in a live production
environment?
Date: Thu, 6 Nov 2003 10:56:44 -0500
Reply-To: asterisk-users@lists.digium.com
> 5) Attempt to balance the hybrid at the 2-line to 4 line
interface.
This is _precisely_ why my rollouts are all strongly recommending using
a
channel bank instead of the cheap X100P/TDM400P cards -- a lot of work
has
been put into the hybrid circuitry to dynamically adjust to the line
impedance. I've had no serious issues with the X100P/TDM400P in small
scale stuff but the echo cancel IMO should be done where it originates
--
at the hybrid.
Having said that, I do have "echocancel=32" in my zapata.conf for the
T100P
connected directly to an Adit600 FXS channel bank. I also have an old
CAC
AB1 with 12FXS and 12FXO ports I am going to deploy shortly to test
things
like far-end disconnect and other issues.
> be the only real solution. Part of the problem arises from the use of
> lower impedance telephone wiring nowdays. The typical characteristic
> impedance of Cat5 twisted pair is about 100 ohms and many line cards
are> optimized for a 600 ohm line. This is made worse if the DC resistance
of> the wiring to the CO switch is relatively low. I haven't tried this
This is a neat idea; something I have not thought of. However my ideal
PSTN
termination is digital (PRI) ... something to eliminate the hybrid
altogether, at least on my end. :-) For deployments where I am simply
providing VOIP to an existing phone system, I am recommending installing
a
T100P and a digital trunk for the existing KSU; again to eliminate the
hybrid mess, or at least push it off to someone else's problem. :-)
> 6) Try messing with Tx and Rx gains.
Something I have noticed is that on the Adit600 FXS ports, I have had to
set
its RX attenuation to -7dB!! (TX to -3dB) If my math is correct, that
means I am attenuating 85% of my incoming signal! Is this perhaps what
you
are referring to with the super-low impedance?
Thank you for this super technical and informative post. This is what
*-users needs... more tech and less running around in circles with the
same
issues over and over!
Regards,
Andrew
--__--__--
Message: 11
Date: Thu, 06 Nov 2003 08:57:28 -0700
From: Chris Hirsch <chris@base2technology.com>
To: asterisk-users@lists.digium.com
Subject: Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users]
Setcontext
based on CID...)
Reply-To: asterisk-users@lists.digium.com
This is a multi-part message in MIME format.
--------------020407020503000707090406
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Content-Transfer-Encoding: 8bit
I hate doing metoos but I tried to get ahold of Michael Baird and never
got a response....does anybody have the AGI code that Michael used for
his Anti-Ex Girlfriend as described below?
Thanks!
Chris
>is the AGI available?
>Uriel
>
>-----Original Message-----
>From: asterisk-users-admin@lists.digium.com
>[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Michael
Baird>Sent: Saturday, September 27, 2003 6:37 AM
>To: asterisk-users@lists.digium.com
>Subject: Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users]
>Setcontext based on CID...)
>
>
>I do it through AGI, I send the call to an external perl script, check
>the called-from-id against a mysql database, then send the call back to
>a context based on a ruleset I use, call-approved/call-not-approved/no
>digits received. Each context having a different voice message, so that
>the caller will know the problem, it works very well.
>
>Regards
>MIKE
>
>
>
>>Blatantly stolen from Mark's presentation:
>>
>>exten => 600/2565551212,1,Congestion
>>exten => 600,1,Dial(Zap/9,15)
>>exten => 600,2,Voicemail(u600)
>>exten => 600,102,Voicemail(b600)
>>
>>If the Caller*ID matches the ex-girlfriend (2565551212), provide
>>immediate congestion tone. Otherwise try dialing on Zap/9 for up to 15
>>seconds. If there is no answer send them to voicemail, preceeded by
>>?unavailable? message. If the interface is busy, send them to
voicemail>>with a ?busy? message.
>>
>>
>>Jeremy McNamara
>>
>>
>>
>>Matt McIntyre wrote:
>>
>>
>>
>>>I was wondering if someone might be able to offer a suggestion to me
>>>about how I might go about dropping a caller into a context specific
>>>to their CID. For example, I would like to be able to dial Asterisk
>>>from a specific number (a mobile phone) and have it drop me into a
>>>context other then the one that normal callers receive that has more
>>>options tailored to things I might want to do. I assume that
?answer?
>>>can somehow be used to do this but I thought I might ask the experts
>>>and see what they might have to say.
>>>
>>>Thanks in advance,
>>>
>>>(You guys are great)
>>>
>>>Matt
>>>
>>>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>>>! Matt McIntyre (KF4FGZ)
>>>! Certified Novell Administrator
>>>! (336) 334-1134 (Campus telephone)
>>>! (336) 215-7199 (Mobile telephone) <- Please note the change
>>>! (336) 334-1134 (Facsimile)
>>>! E-MAIL: mamcinty@uncg.edu <mailto:mamcinty@uncg.edu>
>>>! AIM: MixMANJaVa
>>>! ICQ: 11956085
>>>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>>>
>>>
>>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users@lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
--
Only in America are there handicap parking places in front of a skating
rink
http://ccicolorado.org
Exceptional Dogs for Exceptional People - Help Out Today!
--------------020407020503000707090406
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="Content-Type"
content="text/html;charset=ISO-8859-1">
<title></title>
</head>
<body text="#000000" bgcolor="#ffffff">
I hate doing metoos but I tried to get ahold of Michael Baird and never
got a response....does anybody have the AGI code that Michael used for
his Anti-Ex Girlfriend as described below?<br>
<br>
Thanks!<br>
Chris<br>
<blockquote type="cite"
cite="mid008001c3856d$c34e8ba0$650aa8c0@uriel01">
<pre wrap="">is the AGI available?
Uriel
-----Original Message-----
From: <a class="moz-txt-link-abbreviated"
href="mailto:asterisk-users-admin@lists.digium.com">asterisk-users-admin
@lists.digium.com</a>
[<a class="moz-txt-link-freetext"
href="mailto:asterisk-users-admin@lists.digium.com">mailto:asterisk-user
s-admin@lists.digium.com</a>]On Behalf Of Michael Baird
Sent: Saturday, September 27, 2003 6:37 AM
To: <a class="moz-txt-link-abbreviated"
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digiu
m.com</a>
Subject: Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users]
Setcontext based on CID...)
I do it through AGI, I send the call to an external perl script, check
the called-from-id against a mysql database, then send the call back to
a context based on a ruleset I use, call-approved/call-not-approved/no
digits received. Each context having a different voice message, so that
the caller will know the problem, it works very well.
Regards
MIKE
</pre>
<blockquote type="cite">
<pre wrap="">Blatantly stolen from Mark's presentation:
exten => 600/2565551212,1,Congestion
exten => 600,1,Dial(Zap/9,15)
exten => 600,2,Voicemail(u600)
exten => 600,102,Voicemail(b600)
If the Caller*ID matches the ex-girlfriend (2565551212), provide
immediate congestion tone. Otherwise try dialing on Zap/9 for up to 15
seconds. If there is no answer send them to voicemail, preceeded by
´unavailable¡ message. If the interface is busy, send them
to voicemail
with a ´busy¡ message.
Jeremy McNamara
Matt McIntyre wrote:
</pre>
<blockquote type="cite">
<pre wrap="">I was wondering if someone might be able to
offer a
suggestion to me
about how I might go about dropping a caller into a context specific
to their CID. For example, I would like to be able to dial Asterisk
from a specific number (a mobile phone) and have it drop me into a
context other then the one that normal callers receive that has more
options tailored to things I might want to do. I assume that
´answer¡
can somehow be used to do this but I thought I might ask the experts
and see what they might have to say.
Thanks in advance,
(You guys are great)
Matt
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
! Matt McIntyre (KF4FGZ)
! Certified Novell Administrator
! (336) 334-1134 (Campus telephone)
! (336) 215-7199 (Mobile telephone) <- Please note the change
! (336) 334-1134 (Facsimile)
! E-MAIL: <a class="moz-txt-link-abbreviated"
href="mailto:mamcinty@uncg.edu">mamcinty@uncg.edu</a> <a
class="moz-txt-link-rfc2396E"
href="mailto:mamcinty@uncg.edu"><mailto:mamcinty@uncg.edu></a>
! AIM: MixMANJaVa
! ICQ: 11956085
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
</pre>
</blockquote>
<pre wrap="">
_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated"
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digiu
m.com</a>
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://li
sts.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<pre wrap=""><!---->
_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated"
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digiu
m.com</a>
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://li
sts.digium.com/mailman/listinfo/asterisk-users</a>
_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated"
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digiu
m.com</a>
<a class="moz-txt-link-freetext"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://li
sts.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<br>
<br>
<pre cols="72" class="moz-signature">--
Only in America are there handicap parking places in front of a skating
rink
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Exceptional Dogs for Exceptional People - Help Out Today!
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