hell i just got a quote today for D240JCT-1T1 for $4500ish
Yup. That's the one with the fancy DSP on-board that supports echo
cancellation (continuous speech processing) and is generally used for
speech recognition apps.
By the way -- has anyone in the * community looked at adding ASR to the
Linux benefits from the fact that it runs on very common hardware and you
get commodity prices for most parts of an Asterisk system, and the
software is free.
Quite honestly the X100P cards cost peanuts for what is a effectively a
specialist item. If they cost $10 it wouldn't be financially
On Thu, 2003-10-23 at 13:32, Ethan wrote:
what their costs are or what makes them successful. Armchair
businessmen are a dime a dozen; it doesn't help that everytime you
post to the list, you advocate products which will undercut Digium's
source of revenue.
Isn't this what Linux is
I followed the script (closely) that you referenced (including the export),
and a ps ax shows:
21007 pts/2S 0:00 /bin/sh /usr/src/festival/bin/festival_server
21017 pts/2S 0:00 festival --server ./festival_server.scm
21039 pts/2S 0:00 sleep 60
Then I start *, and make a
Does anyone know how to read? The X100P **IS** a winmodem:
http://www.kobian.com/products.php?productid=180
Buy one here for $15.50:
http://www.accupc.com/itemDetail.jsp?pid=fmint56vs/w
I am using one of these right now along with a real X100P without any
issues. They are IDENTICAL, FCC ID's and
Don't put the text in quotes. i.e. Festival(Would you like to play a
game?)
On Thu, 2003-10-23 at 18:07, Rich Adamson wrote:
I followed the script (closely) that you referenced (including the export),
and a ps ax shows:
21007 pts/2S 0:00 /bin/sh /usr/src/festival/bin/festival_server
On Thu, Oct 23, 2003 at 07:54:17PM +0100, Muhammad Nasim wrote:
Hi. Can anyone help me with the following:
[globals]
OFFICEHOURS
OFFICEHOURS=100 ; default to incoming-officehours context
[internal]
exten = *80,2,SetGlobalVar(OFFICEHOURS=100)
On Thu, 2003-10-23 at 16:48, Steven M. Sokol wrote:
hell i just got a quote today for D240JCT-1T1 for $4500ish
Yup. That's the one with the fancy DSP on-board that supports echo
cancellation (continuous speech processing) and is generally used for
speech recognition apps.
By the way --
Greetings everyone.
Anyone knows about a GUI that can spit out various config files without me
having to know every syntax? Please let me know.
Thanks.
Ricky
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Greetings everyone.
Did anyone try using 2 NIC cards on the machine? For some reason, asterisk
can not identify which IP should be used. In the config files (IAX.conf,
sip.conf etc), there is a way to bind the IP address but if the machine is
hooked to a DHCP server (such as cable modem), then
I think that was one reason for the EAGI extension. There is a agi
script to pass audio to Sphinx for speech recognition.
CMU's Sphinx if you carfully design the application to limit
the number of words that could be spoken at any one time, works.
This means re-loading grammers in real
Gerald == Gerald Henriksen [EMAIL PROTECTED] writes:
Gerald On Thu, 02 Oct 2003 11:26:56 -0700, Jan Rychter
Gerald [EMAIL PROTECTED] wrote:
Having worked with GPL software quite a bit, also in the commercial
world, and having gotten some legal advice, I believe that the
anti-patent
If you bind to 0.0.0.0 in a *.conf file, it will
bind to ALL interfaces.
I'm sure there may be a reason NOT to use 0.0.0.0 but
I can't think of it right now.
I'm currently scanning www.pricewatch,com looking for parts
to build a new firewall box so I'm right behind you
--- [EMAIL PROTECTED]
Okay, changed it to look like:
exten = 555,1,Festival(Testing one two three.)
restarted * from scratch, dialed 555 and see:
CLI shows:
-- Executing Festival(SIP/3000-c1e7, Testing one two three.) in new stac
k
== Parsing '/etc/asterisk/festival.conf': Found
WARNING[1209269552]: File
I'm desperate trying to understand the SIP native
bridge. The Asterisk server get the client to bridge together, and everything is
allright. But when a client hangs up, the second stay conntected forever. With
any soft of hard we did get the same issue.
Of course, disconnection works
:) imo it is called Budge Tone .. "budge" from "move"
What will you guys think what the name "HandyTone"imply, which could
suggest convenience ?
The BT looks better as the majority of all hotel room phone i've ever
seen in the US.
Dave Weis wrote:
On Wed, 22 Oct 2003, Michael T
At 09:15 PM 10/23/2003, you wrote:
:) imo it is called Budge Tone ..
budge from move
What will you guys think what the name HandyToneimply, which
could suggest convenience ?
handi-tone (handicapped features) ??
The BT looks better as the majority
of all hotel room phone i've ever seen in the
Gus,
I think I might see the issue. In /etc/asterisk/festival_server.log, I see:
Load server start ./festival_server.scm
festival port=1314
wrapper Thu Oct 23 20:08:37 CDT 2003 : USING DEFAULT CONFIGURATION
wrapper Thu Oct 23 20:08:37 CDT 2003 : waiting
At 8:44 AM +0200 10/23/03, Kerker Staffan wrote:
Hi!
Does * do DNS-lookups when outgoing calls are placed to a different
SIP domain? Can I call from sip:[EMAIL PROTECTED] to
sip:[EMAIL PROTECTED]?
Can * work as a regular SIP proxy in that aspect?
Can * handle SIP URI:s that are complete SIP URI:s
No. When I put 0.0.0.0 in *.conf, I get all kinds of warnings stating
that iax.conf or sip.conf will not work because IP address can not be
identified. There should be an easy fix to this for someone who knows
asterisk (may be mark can take a look). I will be happy to share the
warnings messages
FYI. Haven't dug enough to be able to report any more, but
re-fetched CVS to verify that sometime in the last few days CVS
changes now break my GS phone.
It appears to be at the RTP level. It seems to set the call up just
fine, but no audio is passed back to the instrument.
I reverted, and
Dear Asterisk Users,
I have a question about last call return function, when I press *69 the system only speak out the last caller no. after that I will hear the disconnect tone. I just wondering is it normal or the function should dial out the last call no.? If the system will dial out the no.,
In attempting to make my asterisk server the latest and the greatest
tonight I attempted to upgrade my CVS. In the asterisk directory I ran
the make clean, executed the CVS update -d, and after all the files
completed ran make upgrade. My problem is that when I pull up the CLI the
cvs
[EMAIL PROTECTED] wrote:
In attempting to make my asterisk server the latest and the greatest
tonight I attempted to upgrade my CVS. In the asterisk directory I ran
the make clean, executed the CVS update -d, and after all the files
completed ran make upgrade. My problem is that when I pull
about being cheaper than the next guy. I'd even say that there are those
who would say it isn't cheaper with regards to total cost to set up, and
get started. The bump in the cost is usually due to the cost of
knowledge acquisition. It costs money or time to get this knowledge.
True. I think
Hiya,
here is a patch to fix that:
[EMAIL PROTECTED] channels]# cvs diff chan_vpb.c
Index: chan_vpb.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_vpb.c,v
retrieving revision 1.9
diff -r1.9 chan_vpb.c
100,102c100,102
static
On Wed, 2003-10-22 at 07:44, Andrew Kohlsmith wrote:
Can you _please_ trim the quoted text? There's absolutely no reason to
quote the entire post you're replying to, signature lines and all... +2
points for bottom-posting though. :-)
No, -10 points for bottom-posting but not trimming. If
Hello,
I am new to Asterisk, as of today. Installed on a RH9 box, with no
problems. Built with 'make samples' as to get an understanding of how
things work. Currently, I am utilizing SJPhone as a SIP client - not
interested in shelling out cash for the IP phones, until I know I have a
hold on
On Tue, 2003-10-21 at 17:13, John Brown (CV) wrote:
Can you provide more specific information. Saying Its Broke Jim
doesn't provide enough content :)
True that. :) My biggest complaint was how they used to sometimes take
over the server's MAC address, confusing the crap out of my switch. We
My 'sip.conf' file reads:
[general]
port=5060
bindaddr=0.0.0.0
context=default
[sjphone]
username=name
secret=password
host=dynamic
defaultip=192.168.1.120
username= can go... the part in [] is the username or is on all my
I am trying to get an initial setup up and going
which I assume is a very common question here. My basic
questionsare the following:
Can I get Asterisk up and going without voice cards
using it with SoftPhones internally as a proof of concept. (just calling
extensions and leaving voice
when I put a station on hold I receive this message
res_musiconhold.c, Line 280 (monmp3thread): Read 372 bytes of audio
while expecting 1600
Regards Mick
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when I dial out from my Cisco phone I get this error
File channel.c, Line 2258 (ast_channel_bridge): Didn't get a frame from
channel: SIP/210.9.49.216-c26e
Regards Mick
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