Hi,
-Original Message-
I'm trying to pick up the account code that has been set
already earlier in the dial plan inside an AGI script.
So far what I have is:
$user = $AGI -get_variable('ACCOUNTCODE');
But it only returns a ' 0 '??
Is there a way around this?
Why don't
It's quite easy to write an LDAP interface. There are code snippets on the
web and I can send you my very quick hack, if you like.
Iain
--On Thursday, March 11, 2004 4:06 pm -0600 Brian R. Swan
[EMAIL PROTECTED] wrote:
Hi gang,
I'm looking into writing a some phone book XML/PHP software
Check:
http://www.jaredsmith.net/misc/cisco7960/Directory-0.1.tgz
Regards
Sascha
---
Sascha Knific K Systems Design
Tel. +49-8151-773260Wittelsbacherstr. 6a
Fax. +49-8151-77326282319 Starnberg, Germany
[EMAIL PROTECTED]
Hi all,
I have a problem calling from a sipset to a ISDN subscriber over
a CISCO 1760 GW.
The following setup is used.
UA --- GW --- ISDN
The UA is a sipset, the GW is the CISCO 1760 with ISDN BRI interface
and a standard ISDN subscriber.
The UA is registered with a registrar/proxy.
All numeric
try adding:
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind connect enable 8
to the dial-peer on the Cisco GW...
- Original Message -
From: Manuel Goertz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 12, 2004 2:26 AM
Subject: [Asterisk-Users] SIP
On Fri, 12 Mar 2004, Brian Capouch wrote:
I too am running 6 cards in my system, although not in a high traffic
capacity load environment.
So far my (limited) high-load simulations have shown no problems.
So - is it apocryphal that the Digium cards (drivers) won't share
interrupts?
If
Hi All,
Does anyone have Digium E1 cards in production in Australia? Are any of them
certified?
Any feedback would be appreciated.
Thaks
Alex.
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To
Sorry for double post. Wrong subject :-)
Hi All,
Does anyone have Digium E1 cards in production in Australia? Are any of them
certified?
Any feedback would be appreciated.
Thaks
Alex.
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Does anybody have experience with these units ??
http://www.udcsystems.com/
DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any attachments thereto may contain information which is
confidential and/or protected by intellectual
From time to time, during a conversation, Asterisk seems to detect a
fax tone.
It then tries to redirect it, and prints the following message :
Redirecting Zap/2-1 to fax extension
According to the source, it does this only if it matches a fax
extension in the current context.
I don't have a
Hi all,
I would like to have Asterisk bridge 2 calls with this schema
-inbound call comes in
-the caller id is passed to an external script
-the external script replies with a phone number
-an outbound call to the number provided by the script is made
-if the outgoing call is answered we have to
Hi all,
I am connecting two * (A and B) using a third * (C) as passthru and
billing control. All connections are IAX-2. So, when A wants to call
someone outside, it Dials to C. C analyzes the extension number
and redirects it to the appropriate destination at B, billing the call:
A (exten
Alex,
With Digium's agreement, I am certifying the TE410P for use in Australia.
If you want please talk to me.
At 21:57 12/03/04 +1100, you wrote:
Sorry for double post. Wrong subject :-)
Hi All,
Does anyone have Digium E1 cards in production in Australia? Are any of them
certified?
Any
I too am running 6 cards in my system, although not in a high traffic
capacity load environment.
So far my (limited) high-load simulations have shown no problems.
So - is it apocryphal that the Digium cards (drivers) won't share
interrupts?
If there is a real issue with sharing
Hi All,
I have now got my '*' server up and running quite good.
As stated in earlier posts I am no Linux guru, so a bit of hand holding
required.
First Subject.
I would now like to add h323 boxes to the '*' server, I have looked through
the wiki and followed the instructions about what I
T.38 FAX is in the short-term plans for asterisk-oh323.
Michael
T. Chan wrote:
Dear Michael
Do you foresee implementing these in the near future, one or the other or
both? Thanks
Tc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Manousos
Sent:
Hi,
Check the included README file for installation instructions.
Michael
Erick Weber V. wrote:
Hi all:
Does someone can direct me to an asterisk-oh323 how to or installation
manual
Thanks
Erick
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Hi all,
Ok. Now I know I can't bill a call when I have a native bridge
betweens *. And I do not want a Native Bridge. How could I disable
native bridge? I tried notransfer=yes but connection tries to start
a native bridge and then closes.
Daniel
Toying with implementing a VIC adapter in a C1750 for a pair of pstn FXO
interfaces. Any issues in doing this via * and sip? (or do I need h323?)
Anyone have a used VIC for sale?
Rich
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Anthony Law wrote:
HI,
I have successfully configured asterisk to accept SIP session from sip
phones and use oh323 to forward calls to our gateway using H323 and
eventually PSTN termination. But since some of the gateways are not in our
control, we need to send tech-prefix + phone number to
I can't find this in the archives, pardon me if it has already been
hashed out. I have recently learned of Asterisk and are trying to get
my hands around the scope.
On our University campus we have all of our users in a LDAP directory.
It would be great if we could interface with this store
The codecs are:
SIP Phone:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: G728
Asterisk:
in sip.conf
1: ulaw
2: alaw
in oh323.conf
1: G711U
Gateway:
preference 1: G711U
preference 2:
.
.
.
preference 8: G711A
That's good? Can you see where's the
Is there a way to discard voicemails that are very short or contain only
silence? I cannot find a parameter for either of these, and we are seeing a
lot of empty voicemail messages. Also, some messages have lots of silence at
the end, as if the line is not releasing when the external call hangs
Make sure your using qualify=500 in the sip.conf along with nat=yes,
make sure any firewalls allow 5060 udp and tcp and random ports
above 1 in form your PBX.
If you have all that it should work.
AstGrp wrote:
Yes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Mireia Munoz de jesus wrote:
The codecs are:
SIP Phone:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: G728
Asterisk:
in sip.conf
1: ulaw
2: alaw
in oh323.conf
1: G711U
Gateway:
preference 1: G711U
preference 2:
.
.
.
preference 8: G711A
Try with
adding the day / month augments fixed the issue. I like the suggestion
about breaking up the current config.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, March 11, 2004 3:42 PM
To: [EMAIL PROTECTED]
Subject: Re:
Andrew Gillham wrote:
Sounds good. I have not been that bothered with it when I make a
normal voice call.
It is mostly annoying when hitting the messages button on the phone.
My delay helped
that situation.
Perhaps on calls where asterisk is proxying the rtp stream we could
have an option
Hi
I have tired using the below syntax but could't go through, I wonder if my
syntax is wrong. Please kindly comment.
Btw I am using asterisk-0.7.1 with oh323-0.5.7
exten = _1613XXX,1,Dial,OH323/[EMAIL PROTECTED]
Error Below
Mar 12 09:30:01 WARNING[5126]: chan_sip.c:2365
I have the dev kit installed and the X100P answers calls and * routes
them as expected. I am not able to dial out at all:
[analog-out]
exten = _9.,1,Dial(Zap/1/$EXTEN:1)
exten = _9.,2,Congestion
included up in the default section
shouldn't this take any call beginning with 9, strip the 9 and
Wow, 1000 IRQs a second. I'm quite amazed. Does anybody know which
applications would require such a low latency ? It does seem to me that
this way of doing things is rather dangerous and prone to problems.
Anybody can comment ?
Thanks
Bruno
Nicolas Bougues wrote:
On Fri, Mar 12, 2004 at
hi,
Try
exten = _9.,1,Dial(Zap/1/${EXTEN:1})
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of randulo
Sent: 12 March 2004 14:54
To: asterisk list
Subject: [Asterisk-Users] X100P and TDM400 questions
I have the dev kit installed and the
Anthony Law wrote:
Hi
I have tired using the below syntax but could't go through, I wonder if my
syntax is wrong. Please kindly comment.
Btw I am using asterisk-0.7.1 with oh323-0.5.7
exten = _1613XXX,1,Dial,OH323/[EMAIL PROTECTED]
This should work. Provide a more detailed Asterisk log to
Looking at my /proc/interrupts:
0: 59709041 XT-PIC timer
5: 597050409 XT-PIC wcfxo
7: 597211339 XT-PIC wcfxo
10:4538876 XT-PIC eth0
11:3044608 XT-PIC aic7xxx, eth1
The voice cards generate an order of magnitude more
if you do a apt-get install asterisk you can get it all ready compiled and
everything ondibian.
I just did that last night.
hth
- Original Message -
From: Yury Bokhoncovich [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: Duane [EMAIL PROTECTED]
Sent: Thursday, March 11, 2004 11:03 PM
Subject:
David J Carter wrote:
hi,
Try
exten = _9.,1,Dial(Zap/1/${EXTEN:1})
Holy cut and paste! That should make a difference, thanks. (not at the
office to find out)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of randulo
Sent: 12 March 2004 14:54
To: asterisk
Steven Critchfield wrote:
On Fri, 2004-03-12 at 05:26, Rich Adamson wrote:
I too am running 6 cards in my system, although not in a high traffic
capacity load environment.
So far my (limited) high-load simulations have shown no problems.
So - is it apocryphal that the Digium cards
Ok.. Let me start by saying that SJPhone works fine through NAT and the
Cisco phones inside the internal network work fine also... It's just the
Cisco phones on the outside using NAT.
For Testing I opened the Firewall open on the IP for the * Server. I
have done, everything you recommended
I have noticed that sometimes you need to comment out profiles with
nat=yes on and then reload, then uncomment them and reload, for Asterisk
to clean out historical settings. Try that. I have run phones before on
odd port with out trouble, so I don't think that is your problem.
AstGrp wrote:
I am experiencing a strange problem and wanted to know if someone
has faced any similar issues or could provide me with a way to counter this
problem. I am in the process of experimenting with asterisk and trying to setup
a basic functional system. I have one TDM400P (single port) and one
Let me start by saying I have no cisco phones, and no idea how to
configure them. I will speak to the use of asterisk behind a NAT'ing
firewall, which I believe to be your setup.
Asterisk is pretty picky about how SIP and RTP packets are handled by
a NAT firewall. Basically you need to maintain
Maybe you could try turning off the primary sync source that you have set on
1 and 3, and let the external switches source all clocks. Don't know if
this will help, but its worth a try
regards
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email:
Hello..
I have a strange problem with one of my own
apps..
I'm dialing from NetMeeting to an * server , listen
to some prompts and then I'm hanging up my netmeeting..
My app is waitting some dtmf's and check for
channel stats (chan-_state) to see if i'm alive..
Although I've hanged up ,
On Friday 12 March 2004 05:41, Nicolas Bougues wrote:
From time to time, during a conversation, Asterisk seems to detect
a fax tone.
It then tries to redirect it, and prints the following message :
Redirecting Zap/2-1 to fax extension
According to the source, it does this only if it
On Friday 12 March 2004 07:55, Jim Sneeringer wrote:
Is there a way to discard voicemails that are very short or contain
only silence? I cannot find a parameter for either of these, and we
are seeing a lot of empty voicemail messages. Also, some messages
have lots of silence at the end, as if
My ZAP calls are being dropped after 7 seconds. The only info I can find
is:
Mar 12 14:03:08 WARNING[98311]: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Response)
Mar 12 14:03:11 WARNING[98311]: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Request)
I
Ok...
If put in the qualify=500... It says it is unreachable... But ping
times Are fine...
PING 69.133.182.77 (69.133.182.77) from 10.100.254.21 : 56(84) bytes of
data. 64 bytes from 69.133.182.77: icmp_seq=1 ttl=241 time=54.9 ms 64
bytes from 69.133.182.77: icmp_seq=2 ttl=241 time=52.0 ms
Hi,
I am having problems with answer detection on a Wildcard X100p. Here is the
issue:
When placing a call by copying a sample.call file into
/var/spool/asterisk/outgoing, the call is made, but execution of the context
continues immediately after dialing. I would like to wait until the callee
I've scoured the list about the TDM400 PCI 2.2 issue and I'm more
confused than ever. Some say the card is compatible but PCI 2.2
compatible mobo is not necessary. Note: too bad there's no good way to
remove false answers from the archives.
I just received the card and put it on a mobo that is
On Fri, Mar 12, 2004 at 07:41:32AM -0600, Jason Winget wrote:
I can't find this in the archives, pardon me if it has already been
hashed out. I have recently learned of Asterisk and are trying to get
my hands around the scope.
On our University campus we have all of our users in a LDAP
Did you plug the power into it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent: Friday, March 12, 2004 1:17 PM
To: Asterisk List
Subject: [Asterisk-Users] TDM410 final questions
I've scoured the list about the TDM400 PCI 2.2 issue and I'm
I'm about over this.. okay,, here is what I got..
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = inbound ; Default for incoming calls
tos=lowdelay
tos=184
disallow=all; Disallow
I think it's because in de [general] section you only allow=ulaw and you
shold allow=g729 to.
I'm a newbie, hope I can help
Best Regards
Erick
- Original Message -
From: Billy Huddleston [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, March 12, 2004 1:02 PM
Subject:
Bob bevins wrote:
Did you plug the power into it?
Yes, sorry I *knew* when I began that I needed to say I plugged it in,
but I forgot anyway. I've even tried different power plugs (since some
have had noise from sharing power connex with the hard drives.
Steven == Steven Critchfield [EMAIL PROTECTED] writes:
Steven As I understand the PCI spec, there are 4 interrupt lines
Steven called A,B,C, and D. In slot 1, They appear in that order. In
Steven slot 2 they shift, in slot 3 they shift and again in slot 4.
That is correct, except that all
The pings are pinging the out side port on the nat device, You don't
have a
rule in your nat table to associate it with a device on the inside. You
should
reset the phone and then see if the qualify shows a return time. You will
need to make the phone register every time you change you config
Did you try using the free technical support that comes with the dev kit?
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Asterisk Learner wrote:
I am experiencing a strange problem and wanted to know if someone has
faced any similar issues or could provide me with a way to counter this
problem. I am in the process of experimenting with asterisk and trying
to setup a basic functional system. I have one TDM400P
Sounds to me that your asterisk first negotiates g729 with your phone, then
negotiates ulaw with the gateway (since it *is* the preferred codec in your
config), and on a re-invite the logic breaks up either in the phone or in
the gateway (or perhaps in the asterisk itself, I am not absolutely
I am experiencing a strange problem and wanted to know if someone has
faced any similar issues or could provide me with a way to counter this
problem. I am in the process of experimenting with asterisk and trying
to setup a basic functional system. I have one TDM400P (single port) and
When I use reinvites everything works perfectly (so phoneA--phoneB
directly works fine). When I shut off reinvites
(phoneA--asterisk--phoneB) I get the following with PhoneA initiating
the call:
Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference
is 3360, ms is 440
Mar 12
We are running 2 Asterisk boxes (ast-a and ast-b) connecting our 2
offices. We did some voice quality testing as follows:
ast-b calls ast-a externally (ie. over public phone lines), ast-a
forwards the call back to ast-b over IAX. Since ast-b can't access the
public phone network by itself it
Okay, I add allow=g729 into the [general] section of sip.conf...
I can now place calls via ulaw or g729 without any problems.. simply by
setting the allow= in the phone's sip entry..
However, INBOUND is a whole nother problem...
I get a really strange buzz sound on inbound calls.. and... here
I have two Asterisk systems running in my environment.
In between the two there is a router running NAT. One server services
extensions 90XX and the other extensions 95XX. Both boxes are running Red
Hat 9 with version 0.7.2 Asterisk.
I am running IAX and registering an IAX softphone to
Michael
Thanks alot, so native bridging will not be something that you would do
anytime soon, eh?
Thanks
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Manousos
Sent: Friday, March 12, 2004 7:51 AM
To: [EMAIL PROTECTED]
Subject: Re:
I compiled Asterisk last night from the stable cvs branch. I have two
X101P cards, and three Quicknet Phonejack ISA cards. Asterisk is able to
receive calls on both lines, and all three extensions are at least partly
working. Here's basically what it's doing:
I can pick up any extension and
Just connected my X100P to Verizon. I stumbled across a config that
works, for the moment, with this Dial command:
;this works, because it prefixes a 1 on the dialing. But why does it?...
exten = _NXX,1,Dial(Zap/1/609${EXTEN}|55)
The comment says it all. The card/SW seems to dial a 1
Well, I did some more playing around and uncommented the format=slinear
line in phone.conf. This has resolved my problems. I hope this helps
someone else out.
I compiled Asterisk last night from the stable cvs branch. I have two
X101P cards, and three Quicknet Phonejack ISA cards. Asterisk
Darrin, I had a similar (though not identical) problem. The solution
in my case was to add notransfer=yes in the iax.conf context for the
IAX softphone. It's possible that the hand off to attempt a native
transfer for you is failing because one of the servers is behind a NAT
router. Anyway,
If you want reliable progress (busy, answer, etc) you must use a digital line
like a T-1 and/or PRI. It's as simple as that. There is no way for the X100P
to know when the call has been answered since analog lines do not signal that.
callprogress tries to fake it by listening to the audio and
A few days ago the 7960 phones were delivered. Today the received the
power adapters. However, we've seen nothing about the SIP licenses
(which were bundled into the price.)
Does anyone have a tftp site that I can use to download the firmware. I
would like to use this site until Lewan Assoc.
just found that over the net.
looking forward to be able to try it with * :)
http://www.zyxel.com/product/P2000W.html
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
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[EMAIL PROTECTED]
Excellent! That did the trick! Thanks for the tip!
Darrin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hadar Pedhazur
Sent: Friday, March 12, 2004 3:13 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hang-ups when using IAX
Darrin, I had a
Hi all,
Has anyone found a way of extracting callerID information from a BT pstn line.
I'm currently using a x100p which as we all know can't detect the callerID on
a UK BT pstn line.
Has anyone found any hardware which can detect the callerID which could be
interfaced to the * system in some
Hi,
I'm having a bit of a problem. I have two Asterisk servers, one serving SIP
clients on the outside of a NAT, the other on the inside. The internal one
also serves PSTN and IAX clients.
When I call someone (who is on SIP) from any phone registered with the
internal Asterisk, I get through to
Tor Houghton wrote:
PHONES1=IAX/[EMAIL PROTECTED]
Did you try IAX2/[EMAIL PROTECTED] ?
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
Check the codecs allowed. Cisco supports most, but in my tests I
limited
myself to G.711U.
Another possibility, and one I seem to remember having is that the IP
address of the gateway did not match the * server, and as such did not
have access to the correct Calling Search Space and Media
At 4:06 PM -0600 on 3/11/04, Brian R. Swan wrote:
Hi gang,
I'm looking into writing a some phone book XML/PHP software for my Cisco
phones. Specifically, I'd like to be able to use a web interface (on the
computer) to maintain a contact list, and then dial from it on the phone.
Maybe using
Date: Fri, 12 Mar 2004 20:31:24 -0500
To: [EMAIL PROTECTED]
From: John Todd [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] XML Phone book software.
At 4:06 PM -0600 on 3/11/04, Brian R. Swan wrote:
Hi gang,
I'm looking into writing a some phone book XML/PHP software for my Cisco
phones.
Can anyone post an example of the blacklist command
in a dial plan?
I followed the instructions from wiki, but cant get
it to work.
Basically I want to be able to block off (to
another extension) certain tel#'s that dial in.
Thanks everyone,
Juan Cardenas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen R.
Besch
Sent: Saturday, March 13, 2004 12:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Strange Problem
Asterisk Learner wrote:
I am experiencing a strange problem and wanted to
It does not dial a 1. The '1' denotes the Zap channel number which in
this case is probably your X100P. Zap channels are assigned to Zap ports
depending on the order in which you do a modprobe on them.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
On some telco systems, the person who makes the call is the only one who
can 'release' the call. As such, even if the callee (person who was
called) hangs up, and then picks up again (say minutes later) the caller
will still be on the line. As soon as the caller hangs up, both ends are
Cool, thanks! I must have missed this in my searching.
Thanks,
Brian
On Friday 12 March 2004 7:12 am, stan wrote:
On Thu, Mar 11, 2004 at 04:06:41PM -0600, Brian R. Swan wrote:
I'm looking into writing a some phone book XML/PHP software for my Cisco
phones. Specifically, I'd like to be
Hi,
I am looking for some assistance here with what I think is an IVR
requirement...however I could be wrong - I am not just a newbie when it
comes to this stuff, I am perhaps the newbie of newbies at telephony stuff.
So I would really appreciate it if anyone could give me a general steer just
Michael Shuler wrote:
When I use reinvites everything works perfectly (so phoneA--phoneB
directly works fine). When I shut off reinvites
(phoneA--asterisk--phoneB) I get the following with PhoneA initiating
the call:
Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference
is
On Sat, 2004-03-13 at 14:40, Andrew Braae wrote:
Hi,
I am looking for some assistance here with what I think is an IVR
requirement...however I could be wrong - I am not just a newbie when it
comes to this stuff, I am perhaps the newbie of newbies at telephony stuff.
So I would really
Update...
I did some more testing today.. And with the same setup but one box
behind a Linksys router and another box behind a Pix firewall.. The
linksys works with no problems... So it appears to be how the PIX is
handling NAT SIP... If any one has any thoughts on this , it would be
greatly
Nathaniel Powning wrote:
Did you try using the free technical support that comes with the dev kit?
I will do that, but I thought the huge number of configs of those of you
on this list using this stuff in the real world would be a bigger
knowledge base. I also was hoping to find someone close
Adam,
That is a good point. I will have to check it out if it is indeed the
case and let you guys know.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Saturday, March 13, 2004 7:36 AM
To: [EMAIL PROTECTED]
Subject: RE:
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