of the agreement on my
departure from Mexuar,
the Corraleta applet source code Westhawk Ltd wrote for them has been
released under the GPL.
it is available for download at :
http://www.mexuar.com/files/corraleta_sdk.rar
Tim.
On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote
On Wed, 2009-01-14 at 17:38 +, Tim Panton wrote:
On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote:
Thank you for getting that code contributed to the community. Is
there
a spec somewhere of the features supported by that applet? A version
history? Docs of the SDK it's
and Cisco 79xx phones?
Michiel van Baak
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other evident thrashing. I
have restarted the machine, and the Manager performance isn't changing.
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, or at least 10?
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On Tue, 2008-05-13 at 22:46 -0400, Steve Totaro wrote:
On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein [EMAIL PROTECTED]
wrote:
I have over a half-dozen different SATA hard drives, each with
different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each
one's
into the USB of a single server with a merged dialplan and a
few other tweaks to point at several different mounted drives, rather
than one per host/IP#.
Col
- Original Message -
From: Matthew Rubenstein [EMAIL PROTECTED]
To: Asterisk -Users asterisk-users@lists.digium.com
Sent: Wednesday
://www.google.com/search?q=site%3Ahttp%3A%2F%2Flists.digium.com%
2Fpipermail%2Fasterisk-users%2F+zrtp
http://bugs.digium.com/view.php?id=10024
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asterisk
Linux app) to send/receive the fax images via
BroadVoice. Or maybe there's some other Internet fax gateway I can use
either my fax/ATA or Asterisk, so I can use my cablemodem for this fax
send/receive work.
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prohibited. Please reply to the sender that you have received
the message in error, then delete it.
Hell, I wasn't even allowed to tell you that they're not allowed to tell
you.
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Tzafrir Cohen
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regards
Gonzalo
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, they have a
chance to
get through to me regardless of the screening. Teleslime doesn't, and
they've paid for the call anyway.
--
Godwin Stewart - Horwich IT services
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Regards,
Dean Collins
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Thanks
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to put my hands on an UCM6.1 box very soon
to try that out and eventually grab the xml profiles.
As soon as I get the info I'll surely post it on this ML and on
voip-info too.
Alberto.
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for testing queue-agents scenarios but i'm sure you can
adapt.
Atis
Alex Balashov
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to avoid or that changes the security property of a call.
As I said to Phil: A PBX is designed to be a man-in-the-middle
attack.
There's certainly room for discussion, brainstorming and wild ideas
here.
/O
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looking for replacement sound files for the default Allison,
as I
feel
she is kind of breathy. I have heard other sound files on other
asterisk
sounds, done by her, and they sound fine... are there two
recorded
versions of the prompts floating around?
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On Thu, 2008-01-24 at 13:51 -0500, Steve Prior wrote:
Matthew Rubenstein wrote:
Is anyone else interested in creating new voices for Festival (the
voice synth bundled with Asterisk) that might not be as good as
Allison's recordings, but are better than the current Festival voices
just want to say yes or no for the sake of the poll, fell
free to
respond to me off-list. However, also fell free to respond here if
you have
more verbose comments on the topic that you would like to share.
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Jabber calls as lightweight as, say, each SIP call? If
not, is there a way to increase the capacity of Asterisk to carry about
as many Jabber calls as it can carry SIP calls?
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services button. i read the asterisk wiki and it mention
there's a CMXML_App_Guide.pdf file but there's nowhere
can i find a link for it. does anybody know where can
i find it?
regards.
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-BRIstuffed-0.3.0-PRE-1q
SCCP firmware
Load File: TERM70.7-0-1-0s
App Load ID: Jar70.2-9-0-117.sbn
JVM Load ID: CVM70.2-0-0-112.sbn
OS Load ID: cnu70.2-7-4-134.sbn
Boot Load ID: 7970_64060118.bin
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] On Behalf Of Matthew
Rubenstein
Sent: Friday, December 21, 2007 10:16 AM
To: Asterisk -Users
Subject: [asterisk-users] 7970 CTLFile.tlv?
I've got a Cisco 7970 that's not completing its network
registration to
Asterisk. The Registering message stays on the screen (with the moving
file. It's optional, and will be skipped if it cannot
be found. Your problem is elsewhere. I've found that the 7970s are very
finicky. I've never had luck with the SEPMAC.cnf.xml - only
XmlDefault.cnf.xml (case may vary - check your tftp logs)
Matthew Rubenstein wrote:
I've got
-home
.
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wondered how much it could be loaded, so I tested it with pbx-test:
I could place up to 15 simultaneous SIP calls before it got no more
responsive.
All in all a good, stable and cheap solution for home and home-office
environments.
My 2 cents,
Giuseppe
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responsibility.
Glenn
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On Mon, 2007-10-01 at 11:44 -0500, Jason Parker wrote:
Matthew Rubenstein wrote:
I just got SIP firmware images from Cisco for installation on
7970G.
The way I understand it, that $15 doesn't actually even give you the
right to
use the SIP firmware. It only gives you the right
to solve it if so.
Regards
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Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded in some other app/let. Preferably GPL. Thanks.
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/
Your server seems very slow, often timing out.
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DirectoryEntry
NameEmployee B/Name
Telephone7002/Telephone
/DirectoryEntry
/CiscoIPPhoneDirectory
Check also Open 79XX XML Directory :
http://web.csma.biz/apps/xml_xmldir.php
hope that help
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in, say, an LDAP server? All development using Asterisk instead of
CallManager services, of course.
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for Asterisk to keep 2 separate calls, or pairs
of legs in a conference call, synced exactly enough (including traveling
over the Net between the same 2 IP#s) for them to arrive as a stereo
pair at the endpoint?
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by asterisk. polycom has now HD codec
On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Are there any speakerphones or other conferencing HW phones that play
the audio in stereo? Either their own speakers, or jacks for an amp with
room speakers? Is there any way for Asterisk to deliver
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Matthew Rubenstein wrote:
I tried asking in another thread this week, but I'm not sure
people saw
the actual subject of the question. Does anyone know where to find
documentation of xPL
, but where is some expertise for using it without Trixbox?
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,
concise, *tested* guides and instructions for Asterisk/xPL home
automation somewhere other than just a needle in the
http://www.google.com/search?q=xpl+%22home+automation%22+asterisk
haystack? Or maybe there's a better interface than xPL.
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to become reliable. This outage is a serious warning for
future dependence on those connected services. If the media can't even
report it, how can we expect anyone to do anything to fix or mitigate
it?
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prefers, then find the one they each have in common so the
fewest legs need Asterisk to transcode to their odd man out codec.
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; charset=US-ASCII
On Sat, 04 Aug 2007 19:52:21 -0400
Matthew Rubenstein [EMAIL PROTECTED] wrote:
I currently have an AGI that calls the Festival text2wave app
to write
a wav file that my dialplan plays into a call with the Background()
command. But the voice sounds terrible: like
or some other Asterisk command
to take the WAV data from a pipe to a running text2wav process, rather
than writing a file with text2wave and then reading it (and then
deleting it) in the dialplan/AGI?
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to manage. And there's probably
lots of performance optimization - not to mention deployment
optimization.
Stanis?aw Pitucha
Gradwell Dot Com
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-
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On Sat, 2007-07-07 at 08:39 -0500,
[EMAIL PROTECTED] wrote:
Date: Fri, 06 Jul 2007 12:02:53 -0600
From: Stephen Bosch [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office.
To: Asterisk Users Mailing List - Non-Commercial Discussion
/patog/week31/OG/html/1309-1/US07085364-20060801.html
. Certainly one to watch, as he's watching us and Asterisk.
On Sat, 2007-07-07 at 08:32 -0700, Tom Lynn wrote:
On the other hand, the guy could just be using his work e-mail for
personal interests.
On 7/7/07, Matthew Rubenstein [EMAIL
than 100Mb, even if just while
switching
streams?
Video of 100Mb/s? ;-) HDTV doesn't consume more than 20Mb/s, Gige is
an overkill for IP Phone. Though it is used for switching, I assume it
is a 1 in 100 use.
Thanks,
~Vamsi
On 6/13/07, Matthew Rubenstein [EMAIL PROTECTED] wrote
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with those features exist already, or do I have to write
it?
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Regards,
Dean Collins
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/Cognation
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help you find a place to buy one:
http://www.abptech.com/aboutus/find_reseller.php
Shanon
ABP Technology
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multipart/alternative-- next part
--
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that the click does not navigate away from the current
page.
It requires an Asterisk Manager connection.
See http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script for
more
details.
Kind Regards,
Richard Hamnett
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did work on converting G.729 to G.711 to disk storage in real time
and
that took about 3% of a Xeon CPU for full duplex.
Memory wise each convert call might have used 640KB in buffers and
trash, but not much.
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the cost of other pass
through codecs.
I did work on converting G.729 to G.711 to disk storage in real time
and
that took about 3% of a Xeon CPU for full duplex.
Memory wise each convert call might have used 640KB in buffers and
trash, but not much.
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://lists.digium.com/mailman/listinfo/asterisk-dev
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-
javascript/DHTML
friendly and lightweight.
Is that applet available unbundled from the rest of your software and
service package? At a flat (ie not per-instance) price?
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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it for a
low-load persistent store like a HD, where a HD would be overkill in
every way.
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)
1.514.312.7030
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Rubenstein
Sent: Sunday, April 01, 2007 9:08 AM
To: Asterisk-Users
Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off
Topic: Open Source USB Softphone)
Here's a flipside of this subject: what
of Asterisk's vulnerability to
these? Any mitigations?
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displayed in the same day's view list, as if the
layers were all events in a single calendar.
And is there a way to get the OX Web interface to do this? Or a place
in the source code that can be recoded to do it? Thanks.
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Sorry, I sent that message to the wrong list. Tho if you know the
answer, please don't let that stop you from emailing it to me :).
On Thu, 2007-02-15 at 08:21 -0700,
[EMAIL PROTECTED] wrote:
Date: Thu, 15 Feb 2007 08:54:43 -0500
From: Matthew Rubenstein [EMAIL PROTECTED]
Subject
transcoding
work
being done there?
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a
major cost fraction. And of course running out of capacity by surprise
is a crippling blow.
On Sat, 2007-02-10 at 15:57 -0500, Andres wrote:
Matthew Rubenstein wrote:
Are there 45 G.729 instances for the 45 ZAP legs in addition to 45
G.729 instances for the 45 SIP legs? Or do the ZAP
DB from the install.
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Is there an Asterisk command, app, AGI (or other) that can be called
with a phone# (or list) that will lookup somewhere definitive and report
whether the phone# is registered to a mobile phone or not? How about
other data, like its home city/district etc?
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On Mon, 2007-02-05 at 23:05 -0500, Steve Prior wrote:
Matthew Rubenstein wrote:
The real advantage in choosing an AGI (or CGI or ...) platform/language
is *reusing* the existing code that already runs on that platform, with
Well of course you should pick whatever AGI
/UI?
Steve
Cheers,
Kate
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, #licenses = #calls. But if your app
decodes both G.729 legs into ulaw (or other working format) data that is
then mixed or otherwise processed, then the two simul codecs for the two
legs need two licenses.
-a
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.)
Hope to have more info posted this week.
-MC
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Thanks anyway for trying to help.
On Thu, 2007-02-01 at 22:59 -0500, Asterisk wrote:
On Thu, 2007-02-01 at 08:47 -0500, Matthew Rubenstein wrote:
The point is that the SIP carrier side gets the abort *before the SIP
carrier can complete the connection*. That doesn't take 45s
, Asterisk wrote:
Yeah, your waittime parameter in your call file is set to 45 seconds.
db
On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote:
I used the FreePBX on Debian HowTo at
http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
to initiate calls to my
command
'Command'
Jan 30 23:50:44 DEBUG[6268] manager.c: Manager received command
'Command'
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royalties).
Will H.264 become the favorite high-quality Asterisk codec, or will it
perhaps force G.729 to become free, or negligibly cheaper?
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? Any ideas how efficient is the Asterisk/x86 code
compared to the maximum in the algorithm, that either SW optimization or
porting to a more efficient processor (or both) could produce?
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install libpq-dev
and compile again
I hope this be helpfull ;p
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workaround for this scenario?
It shouldn't be a problem if you're only doing IP takeover and have
bound the licenses to each server separately. If you're sharing the
storage, then that could pose a problem.
Leo
DatVoiz Singapore Pte Ltd
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, performance, missing features) would make the
Digium (or other $) license worth paying for?
On Mon, 2007-01-08 at 14:40 +, Thomas Kenyon wrote:
Matthew Rubenstein wrote:
I connect to a PSTN carrier over SIP which requires me to connect with
a g729 codec. I'm using them for just robocalling
Matthew Rubenstein wrote:
I connect to a PSTN carrier over SIP which requires me to connect with
a g729 codec. I'm using them for just robocalling: Asterisk server
originates calls which play a prerecorded file. Can I pre-encode those
stored files in g729 so they don't need to be encoded
or criminal acts.
Matthew Rubenstein wrote:
As far as I know, the g729 patent requires buying a license to operate
any implementation of it, whether Digium's, Intel's, or any other.
Digium is set up to collect royalties (perhaps at a favorable rate) as
part of their license from
on the CPU
Keep in mind I have never used any Ver. of G 729
So tell me what you think.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Matthew Rubenstein wrote:
All of which hassle and expense can be avoided by buying a
license
widely more quickly, and we can get
behind that. If you don't like it, you can still roll your own, just
don't call it IKR when answering the call, and callers will be free to
use your klugey, nonstandard UI, and hate it :).
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, which
prevents economies of scale for consumers and developers.
John
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-type equity
sale on a losing business, which a lot of people are saying. But the
competition will still drive generic minutes rates lower, especially
outside US48 where $0.01:min is rare, even shocking.
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Gandhi
-Original Message-
From: Matthew Rubenstein [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 06, 2006 12:29 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] any possibility of Vonage Integration
On Wed, 2006-12-06 at 05:41 -0700,
[EMAIL PROTECTED
such thermostat available, and for that matter any other
Asterisk controllable home automation devices?
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I can pre-encode recorded audio files with a
G729 codec. So the server can send wakeup call messages to the SIP
carrier without running the codec at call time, just sending the
pre-encoded media to the SIP carrier.
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PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hi
Hi, and thanks for the help :).
On Fri, Dec 01, 2006 at 01:43:20AM -0500, Matthew Rubenstein wrote:
On Thu, 2006-11-30 at 17:56 -0700,
[EMAIL PROTECTED] wrote:
Message: 18
Date: Fri, 1 Dec 2006 00:56:10 +0200
From
]: Leaving directory `/home/roman/install/kernel/linux-2.6.19'
make: *** [linux26] Error 2
seems that commenting out typedef int bool; in xpp/xdefs.h on line
93 works
that out, but don't know if it's completely right thing to do
Roman
--
(C) Matthew Rubenstein
# make clean
rm -f torisatool
) Matthew Rubenstein
# make clean
rm -f torisatool makefw tor2fw.h radfw.h
rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so *.lo
make -C kernel-source-root-dir SUBDIRS=zaptel-1.2.11-source-root
) Matthew Rubenstein
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On Thu, Nov 30, 2006 at 02:44:03PM -0500, Matthew Rubenstein wrote:
I'm having problems installing ztdummy on my
CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP
only
to PSTN). I unpacked the kernel sources
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(C) Matthew Rubenstein
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