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for? People will allways hit 1 g
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not really sure how to troubleshoot this, any ideas?
Thanks,
Enable register_globals in php.
You can also put an extract($_GET); in the top of the php file.
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simultaneous in google or the wiki
(http://www.voip-info.org)
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://www.asternic.org and
look at the Flash Operator Panel. It can do that and more..
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going to POTS. I did not try the Sipura
SPA-3000 yet, but it seems to be a cheap alternative to a gateway,
providing you with one FXO and one FXS for $130 or so. the echo
cancellation in the sipura works well for fxs, it might work well to for
fxo.
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Andrew Kohlsmith wrote:
On Sunday 15 August 2004 12:03, Nicolas Gudino wrote:
If you already have the analog telephone wiring in place, and you are on
a budget, I recomend you to use sipura spa-2000 adapters. They are a
whole lot better than GS phones. You can have 3way conferences and
attendant
, and send DTMF 9876543 before
bridging the call with the calling party.
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Hola Horacio,
Comentarios en línea...
Horacio J. Peña wrote:
Hola!
I'm using asterisk as H.323 - PRI gateway. First call goes
thru ok, second concurrent call fails with:
Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to
context [ip2pri]
-- Executing
this, but you need to open it
on a web browser and use your mouse to drag the manager extension to any
leg of an already bridged call, with some extensions logic and meetme
in the mix. I'm not sure if it will fit your needs, but it might help...
http://www.asternic.org
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#' I can't do that because the Cisco Phone start busy
signal.
How can I start using all DTMF features using Cisco Phone?
Did you try by dialing just '*8' ?
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did not try this, but I know that ActionID is implemented in some
manager commands. Best regards,
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asterisk from CVS. It does not work if I transfer with the ATA
or phone transfer feature.
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, or even call queues. If you can afford
the hardware, you can try with a high end cisco phone.
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using X-Lite, I just
would like to transfer the call to another SIP extension; Just a
Flash+Extension+Hangup CALL...
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application cann fill the gap for sip devices that are not
capable of consultative transfers by themselves...
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not have time to investigate yourself search for Asterisk
consultants on http://www.voip-info.org
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Holger Schurig wrote:
What I'm thinking of is giving each GUI a slot of 10-15 minutes for
a presentation and then a panel discussion on the GUI theme.
No chance for me to pay flight + entry to conference. My wife would hack
me in little pieces :-)
Me neither...
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, just goto
/channels/chan_sip.c and change
#define DEFAULT_USERAGENT Asterisk PBX to whatever user agent you want
, even their own .
Thats it.
You don't need to modify the source to change the useragent. Just put:
useragent=cisco_super_phone
in the general section of sip.conf
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or not), but not soft phones as fas as I know. Some
companies are developing SIP addons to their phones also. Search for
asterisk gui on the wiki.
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Hi John,
John Todd wrote:
At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote:
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
output snip
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Reachable
Time: 81
some more snip
supports manager notifications:
http://bugs.digium.com/bug_view_page.php?bug_id=759
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,
its possible using a combination of meetme and the manager interface...
The Flash Operator Panel I made supports barge in using dragdrop in
combination with the meetme E parameter. http://www.asternic.org
Best regards,
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) will not be removed when reverting back to
the previous version and you will have problems. And just issue a 'make
install' (not a 'make samples'!)
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it to match a valid number and send it inmediatly.
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via dragdrop
* Originate calls via dragdrop
* Barge in on a call using dragdrop
* Set the caller id when transferring or originating a call
* Automatically pop up web page with customer details
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;
In extension.conf add the disa context like this:
[disa]
exten = s,1,disa,no-password|disa
This way, if an error happens with DISA, it will be displayed at the
asterisk console (it will not be hidden inside AGI).
Good luck,
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the voicemail context, so the manager
notifications allways return 0 messages. I will submit a bug/patch to
the bugtracker for this (as it affects the MWI in my flash operator
panel), and I will try to look also at your problem.
Best regards,
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Voicemail email notifications are fixed on CVS as of now (thanks to
citats).
On Thu, 2004-07-01 at 15:16, Nicolas Gudino wrote:
Hi Rich,
On Thu, 2004-07-01 at 11:36, Rich Adamson wrote:
Just upgraded to cvs Head this morning and noticed our voicemail
notification (via email) is failing
Hi,
I have just submited bug 1962. If you are using the Flash Operator Panel
(and maybe other swtichboard/manager applications with MWI) with current
CVS-HEAD, you might need to apply the patch to get MWI working.
Best regards,
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On Thu, 2004-07-01 at 18:03, Nicolas Gudino wrote:
Hi,
I have just submited bug 1962. If you are using the Flash Operator Panel
(and maybe other swtichboard/manager applications with MWI) with current
CVS-HEAD, you might need to apply the patch to get MWI working.
Best regards,
Fixed
to the operator panel mailing list to continue
this thread. Best regards,
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},
+ { CANADA, 0, 0, 0, 0, 0, 0x3, 0, 0 },
super big snip
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)
'cvs -D' is incomplete, you have to specify the date of the version you
are requesting after the 'D'. Anyways, it seems that the problem is
fixed on CVS. Do a 'cvs update'
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'
Segmentation fault
Are you running Redhat or Fedora? If so, read this thread for a solution:
http://lists.digium.com/pipermail/asterisk-users/2004-January/031953.html
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to create channel). Maybe you have to
revert to a previous version till the bug is fixed. ( cvs -D )
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solve your problem.
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
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users registered.
4. Update a BdD.
This is possible? There are any best way to implement this?
Thanks a lot.
It can be done, in fact it's already done. Look here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20GUI
Monastery does exactly what you describe and a bit more.
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Hi Brian,
Brian Cuthie wrote:
BTW, anyone know how to get the SPA-2000 do drop loop current
momentarily when the other end hangs up?
-brian
There is a web configuration option to reverse the polarity in the
latest 2.0 firmware.
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for it, but I also have a web
page that access the asterisk manager port in a regular basis (for
agents login/logout), and I don't have problems or crashes. I'm running
CVS-HEAD.
Best regards,
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version.
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searched the wiki and the list archives. Stock Debian
3.0 stable installation. Any advice? Thanks.
I do not have an TDM400P, but read reports about it in this very list.
Try replacing channel = 2 to 3 in zapata conf. The order of the modules
seems to be relevant...
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when the machine boots. You can
write to it for configuration data... 1 million times will last much
more than a regular hard disk this way. But you will still need a hard
disk for voicemails...
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with ${CALLERIDNUM} instead of ${EXTEN}
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seems to works fine.
You have to add the proper .gsm sounds: 20 thru 29, cien, 100, 200, 300,
400, 500, 600, 700, 800, 900, mil, millon, millones, y
You have different sounds for 100: 100.gsm (ciento) and cien.gsm (cien)
Best regards,
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- Original Message -
From: Altus Snyman [EMAIL PROTECTED]
Good day all
Did someone get the new ver0.5 flash panel working
Is it suppose not to show who the caller is calling,like on ver0.2?
And how do I change the language
Thanks
Altus
Hi Altus,
There is a mailing list for the
more information I could provide?
Kind regards,
Matt Riddell
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these actions:
* Hang-up a channel
* Transfer a call leg via dragdrop
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On Sat, 2004-04-10 at 15:50, Thomas Gallaway wrote:
I run 4 X100P's in our asterisk box. Just make sure you give each card
it's own IRQ.
Paul,
Is the own IRQ per card a strict rule ? Becasue a I have a X100P +
TDM400P on a SMP PIII box, the X100P is sharing IRQ 11 with usb-uhci and
no
, please
send them to me directly! I wont release the .fla source for now, maybe in
the future.
New versions of the application will be posted in
http://sip.house.com.ar/operator , I'm cleaning some bugs in the server and
in the flash applet also. Thanks,
- Original Message -
From: Nicolas
Hi Eric,
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 11:17 AM
Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
Being able to have more buttons as well as changing the button size
would be useful.
What screen resolutions do you
to flash clients. It
might give you ideas on how to implement the betabrite interface. Best
regards,
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supported.
I looked through your code to see if I could make some changes,
unfortunatly I can't speak Italian! :)
Me neither! I speak spanish..LOL.
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in string at ./op_server.pl line 68, CONFIG
line 35.
Try removing line 35 on your op_server.cfg, maybe its a blank line and
the server does not handle that gracefuly. Its not harmfull anyways.
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time (no refreshing necessary), and its graphically appealing.
It's a work in progress... so expect some bugs. I appreciate any
feedback you can give me.
Best regards,
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are running RedHat or Fedora, start asterisk
with LD_ASSUME_KERNEL=2.4.1 Good luck,
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Hi,
On this subject - has anybody managed to implement a method
of warning the caller that their call will expire? I've
Two questions;
Has anybody successfully implemented this, either by way of
source changes or by using the T extension (possibly
something obvious I've missed?)
I made
Hi,
As I'm doing this, I'm considering installing an asterisk box at my
office (about 6-10 different phone stations) and would like to get
opinions on the best quality and/or most well-supported SIP hard phones
and SIP soft phone clients.
I had great luck with sipura spa-2000 adapters. They
Hi Jan,
Try this:
exten = _3XX,1,SetVar(FAXFILE=/tmp/faxfor-${EXTEN}-${TIMESTAMP}.tif)
exten = _3XX,2,rxfax(${FAXFILE})
Good luck,
- Original Message -
From: Jan Baumann [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 28, 2004 7:09 AM
Subject: [Asterisk-Users]
faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4
tifflzw tiffpack
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Maybe this helps. I have 4 sipuras on the same network as Asterisk. I had to
make sure each line on the sipura uses a different sip port: 5060/5061 on
the first one, 5062/5063 on the second, and so on.
Best regards,
- Original Message -
From: Matt McIntyre
To: [EMAIL PROTECTED]
Sent:
- Original Message -
From: Gelson Dias Santos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Does it mean * supports tome based disconnect? How can I turn it
ok? That what my original question (i´m the original poster).
Try with:
busydetect=yes
busycount=7
in zapata.conf
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Subject: [Asterisk-Users] Use of Alert_Info with C7960?
Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around
with distinctive ringing, trying to make it work.
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 20, 2004 8:55 PM
Subject: Re: [Asterisk-Users] Use of Alert_Info with C7960?
On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old
Style
and Synth Low. The first three
Hi Hans,
http://bugs.digium.com/bug_view_page.php?bug_id=773
This patch plays a tone 40,30,20 and 10 seconds before absolutetimeout.
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- Original Message -
From: Hans-Henrik Andresen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday
to have ring
differently for internal calls vs external calls.
Thanks guys,
Matt
Hi Matt,
Try with:
exten = 1000,1,SetVar(ALERT_INFO=Bellcore-r3)
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- Original Message -
From: Chris Lee [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, March 02, 2004 6:42 AM
Subject: [Asterisk-Users] Does it exist - DNS TX record?
When handed a URL type address for telephony, is there a DNS TX record
(like MX but for telephone/Video) that
=0.0
txgain=0.0
group=1
pickupgroup=1
immediate=yes
musiconhold=default channel = 1
^^^
is this a typo? If not, the channel = 1 should go on a line of its own.
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Hi,
I had the exact same problem, and it was caused by my crappy ADSL
connection. I had great download and upload speeds too, but inspecting it
closer, there was a great deal of lost packets. The problem went away when I
changed my ADSL provider.
- Original Message -
From: yair hakak
You can use AGI, the example below uses asterisk-perl:
---
#!/usr/bin/perl -w
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
$number = $AGI-get_data(input-number, 1, 8);
$AGI-say_number($number);
exit 0;
sub mycallback {
the time spent entering and
validating data but I've sat and timed it with a stopwatch and the CDR
is always longer than reality.
-Tim
Hi Tim,
In my case, the CDRs are longer because asterisk last inbetween 5 an 10
seconds to detect the hangup. You should take that into account.
--
Nicolas
Look into bugs.digium.com. I think there is a patch for doing what you want.
- Original Message -
From: Scott Bennett [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 19, 2004 11:01 PM
Subject: RE: [Asterisk-Users] Calls with incoming distinctive ring
So am I to assume
On Mon, 2004-01-19 at 18:38, Olle E. Johansson wrote:
LQ (Asterisk) wrote:
Hi guys,
I was reading that Steve Underwood is working on Asterisk R2 signalling
support, and has the 95% of the work done.
What is R2? I'm curious.
A type of signaling for E1 lines.
--
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people working on this, we might join efforts and work together and came
up with a small linux version with asterisk included, that can boot from
a pendrive or a cdrom.
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before launching
asterisk (with stock redhat 9 kernels):
export LD_ASSUME_KERNEL=2.4.1
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On Mon, 2004-01-12 at 12:23, Maciek Kaminski wrote:
Hi,
What linux SIP UAs do You successfully use with Asterisk?
Maciej Kaminski
kphone work ok, but its very basic.
http://www.wirlab.net/kphone/
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Andy Powell wrote:
Nicolas,
I'd appreciate a copy of this if possible... got a url where I can
grab it?
Thanks
You can grab a copy from the bugtracker:
http://bugs.digium.com/bug_view_page.php?bug_id=773
I've already sent the disclaimer to Digium..
Best regards,
Andy Powell wrote:
I'd be nice to be able to play a tone (or message) at AbsoluteTimeout - N
where N is a number os seconds before the cut-off... a bit like pay phones
(used?) to do...
I have implemented an 'horrible' patch that sort of works. I'm not very good
at C, and I'm new to asterisk.
I still have the problem, but I have noticed one interesting fact. I
have choppy sound from SIP to PSTN, but the voicemail prompts sound
great (asterisk generated sounds are working well)... I will keep trying
and keep you informed.
On Mon, 2004-01-05 at 13:22, WipeOut wrote:
Michael Van
I have a similar problem, with GS phones, X-Lite or Kphone. I tried all
the codecs with the same result. Choppy sound in the direction SIP-Phone
- pstn, but crystal clear sound the other way around. The only
difference in my case is that I have two asterisks servers connected
together via IAX2,
regards,
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is prety
bad. Are there any documents on how to tune jitter buffers? Thanks!
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Did you try with this line before launching asterisk (with stock redhat
9 kernels):
export LD_ASSUME_KERNEL=2.4.1
Best regards,
On Tue, 2003-12-30 at 20:07, JR Richardson wrote:
Thanks for all your help Martin,
Guys,
This is a good find and hopefully could help someone else.
I've been
Richardson wrote:
-Original Message-
No I didn't, I don't have a clue what that is or does. Please explain, I'll
try it and let you know.
Did you try with this line before launching asterisk (with stock redhat
9 kernels):
export LD_ASSUME_KERNEL=2.4.1
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this context in
the proper place. Best regards,
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Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.
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Nicolas Gudino [EMAIL PROTECTED]
House
I'm not a GPL expert, so I have a few questions: Does an AGI script needs to
be distributed in source form? Maybe this application/script is using
Asterisk unmodified. They can sell just their AGI scripts and provide only
asterisk with full source?
- Original Message -
From: Brian West
- Original Message -
From: Michael Rowley [EMAIL PROTECTED]
So, the docs say no more than 2 x100p cards sane, has anyone done it?
put 5 or 6 in one box?
I'm using 4 of them, it works.
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? Thanks!!
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Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.
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continue processing when the call is hung up or terminated or
would I have to use another AGI on the h extension to process post
call operations?
Good question. I can't answer.
This is an important question I need answered for my system..
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Nicolas Gudino
House Internet S.R.L.
Buenos
you do your post call operations?? do you use another AGI
script on the h extension?
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina
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to the dial command as dst in the CDR.
I'm sure there is a proper way to handle this.. and I'm sure someone can
help me figure it out. Thanks!
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina
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scripts)
I'm desparatly trying to get my employer to let me use Asterisk. So I must
get this to work.
I've posted about this before, I'm sorry, but I'm desperate.
I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated)
I'm using Netmeeting to test
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Nicolas Gudino
Hi,
I have the same problem, Im also running RH 9. But Im using SIP only
with Cisco ATAs. There are reports of asterisk not doing well with
RedHat because of the new threads handling in RH kernel. Maybe compiling
a fresh rpm from kernel.org will solve the problem.
Testing my AGI script (writen
G729 themselves, without needing it on * ?
And in the first scenario, if the SIP provider supports G729 and the ATA has
a public IP, do I need to license the codec in *?
Thanks in advance,
Nicolas Gudino
Buenos Aires - Argentina
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Nicolas Gudino
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