Let's fork Digium's GUI.
Zeeshan Zakaria wrote:
Why don't they say FreePBX. After all trixbox is all about FreePBX. If
they remove FreePBX from Trixbox, nothing is left in it. A half
working HUD, and another small little things don't make any major
difference after all.
So are the
Hello All,
While looking for solution to solve my Callback DTMF problem, I came
across LumenVox Speech Recognition software.
Has anyone tried out? Need some feedback before I purchase it... Please
help...
Cheers,
Nitesh
___
--Bandwidth and
Dean Collins is probably the list expert on this.
Thanks,
Steve Totaro
Nitesh Divecha wrote:
Hello All,
While looking for solution to solve my Callback DTMF problem, I came
across LumenVox Speech Recognition software.
Has anyone tried out? Need some feedback before I purchase it... Please
Hi Steve, no I'm no expert at all I do however (or did) have an
interest in building a far more comprehensive solution for an ASP
solution combining other solutions that would have helped the asterisk
community however could never get it off the ground.
Nitesh to answer your original
Dean,
Don't be so modest, you have done your homework... ;-)
Question: Can you elaborate on this a little more, however there are
limitations but for 90% of applications will work great.
Thanks,
Steve
Dean Collins wrote:
Hi Steve, no I'm no expert at all I do however (or did) have an
Thanks Dean and Steve,
I am planning to use for my IVR notification application which is built
using PHPAGI and A2Billing (Callback, Calling Card).
I saw the $50.00 Starter kit does it provide some functionality?
Cheers,
Nitesh
Dean Collins wrote:
Hi Steve, no I'm no expert at all I do
Linux 2.6.20, asterisk 1.2.23, mISDN 1_1_5, Digium B410P BRI card.
When calls come in via ISDN the destination phone does ring but the
caller hears no ringing tone, once the SIP phone is answered everything
works as expected. Calls from SIP phone to SIP phone internally do let
the caller hear a
Hi Nitesh - yep great place to start.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nitesh Divecha
Sent: Saturday, 11
Dermot Bradley wrote:
Linux 2.6.20, asterisk 1.2.23, mISDN 1_1_5, Digium B410P BRI card.
When calls come in via ISDN the destination phone does ring but the
caller hears no ringing tone, once the SIP phone is answered everything
works as expected. Calls from SIP phone to SIP phone internally
Hi - I guess it's not possible to use Asterisk BLF function for
queues... Can someone confirm that? I'm looking for that type of
function with calling queues. I have Grandstream gp2000 phones.
thanks
Todd
___
--Bandwidth and Colocation
Thanks Dean... will update you on the progress...
Cheers,
Nitesh
Dean Collins wrote:
Hi Nitesh - yep great place to start.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL
Bill,
I am not aware of any commercial Asterisk-compatible cards that support
North American BRIs right out of the box. The best I have been able to come
up with was a card sold on eBay, where the seller promises to supply a patch
that needs to be applied to Asterisk (based on BRIstuff) so that
Steve Totaro wrote:
Do you have the r option in the line that dials the extension coming
from the PSTN?
The dial command is visible in the 1st log line I sent and includes a
r option:
Aug 10 17:43:44 VERBOSE[1810] logger.c: -- Executing
Dial(mISDN/1-1, SIP/1000|15|tr) in new stack
Dermot Bradley wrote:
Steve Totaro wrote:
Do you have the r option in the line that dials the extension coming
from the PSTN?
The dial command is visible in the 1st log line I sent and includes a
r option:
Aug 10 17:43:44 VERBOSE[1810] logger.c: -- Executing
Dial(mISDN/1-1,
Basically I tried to pitch the concept of an ASP based speech
recognition SIP service to Tellme back in 2005 as a JV in that they
provided the software and ports and I (eg investors) provided the
servers and bandwidth.
http://www.voip-info.org/wiki/view/tellme
In 2006 they came out with a JV
Yes, I have followed your history with this idea/project with great
interest. I am well aware of what went down.
What I want to know is what specifically you meant by, however there
are limitations but for 90% of applications will work great. What
limitations? What 10% of applications will
Steve Totaro wrote:
OK...
Have you tried removing the r switch?
Ok, removing it now gives a ring tone. The logs show:
Aug 11 18:30:30 VERBOSE[13406] logger.c: -- Executing
Dial(mISDN/1-1,SIP/1000|15|t) in new stack
Aug 11 18:30:30 DEBUG[13406] chan_sip.c: Setting NAT on RTP to 524288
Aug
Dermot Bradley wrote:
Steve Totaro wrote:
OK...
Have you tried removing the r switch?
Ok, removing it now gives a ring tone. The logs show:
Aug 11 18:30:30 VERBOSE[13406] logger.c: -- Executing
Dial(mISDN/1-1,SIP/1000|15|t) in new stack
Aug 11 18:30:30 DEBUG[13406]
On 8/10/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Aug 09, 2007 at 08:12:12PM -0500, David Bandel wrote:
[snip]
...
exten = s,n,Set(loop = $[${loop} + 1])
exten = s,n,Set(loop=$[${loop} + 1])
Thanx Tzafrir. It works now. I guess the documentation I read that
said that white
On Fri, Aug 10, 2007 at 01:08:18PM -0600, Anthony Francis wrote:
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
Really? http://www.pt.com/products/prod_segway_ntwksolution.html
You've vastly misread
On Fri, Aug 10, 2007 at 05:20:38PM -0400, Alex Balashov wrote:
It might be possible to glue something together with it and OpenSER and
a media gateway control protocol like H.248 and a few of these SS7-IP
appliances, but it would have all the ragtag qualities of Napoleon's army
routed in
On Sat, Aug 11, 2007 at 01:25:32AM -0400, Alex Balashov wrote:
If it ever happens, it will certainly take Asterisk's usefulness to a
whole, whole new level, at least as a good cheap media gateway.
I dunno; my instinct is usually not to use PCI media cards at all; I'd
much prefer all
Dean,
Are you aware of any better options for speech recognition? (though
I'm sure more expensive)
On 8/11/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Thanks Dean... will update you on the progress...
Cheers,
Nitesh
Dean Collins wrote:
Hi Nitesh - yep great place to start.
Nuance etc. and Steve to answer your questions - lumenvox just doesn't
have the engine or phonetic capabilities that some of the the larger
systems have.
Like I said before - I've been stunned considering how cheap it is how
good it is but. if you are looking for a less defined utterance
Hi,
I have debian etch 4.0 machine (2.6.18) with two TW-ISDN PCI (Hfc) cards. I
use bristuff-0.3.0-PRE-1y-e (asterisk-1.2.17,libpri-1.2.4,zaptel-1.2.16). I
also have patched zaphfc with zaphfc_0.4.0-test1_florz-13.diff.gz (I load
module: insmod /usr/src/bristuff-0.3.0-PRE-1y-e/zaphfc/zaphfc.ko
Dean,
Hmm.. I was hoping something that could be used with Asterisk on the
machine locally... Nuance doesn't seem to offer that.
On 8/11/07, Dean Collins [EMAIL PROTECTED] wrote:
Nuance etc. and Steve to answer your questions - lumenvox just doesn't
have the engine or phonetic capabilities
No you cant. This message is being dropped as well.
On 8/10/07, Trevor Peirce [EMAIL PROTECTED] wrote:
C F wrote:
This is the postmaster at the list and I am notifying you that your
message failed.
Over the past two days my new posts seem to have silently been dropped.
I wonder if I can
No they have a standalone solution - lol NLVR is a whole separate
server (or server farm) in most onsite installations.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED]
I don't see what difference removing the r option has made from an
Asterisk perspective - in both cases Asterisk tries to emulate a
ringtone but fails for some reason when r is present. According to the
the show application dial help having no r present for Dial should
NOT generate a ringing
C F wrote:
No you cant. This message is being dropped as well.
Shame. Seriously though I posted a new thread right after I posted that
reply. The reply showed up but the new thread still seems to be MIA. No
bounce or anything (and I have no filtering on this account). Weird...
Maybe I'll
Dean,
Can the LumenVox Speech Recognition engine understand numbers?
Sorry to ask stupid questions but kinda curious... as for my application
all I want is to the software to understand the numbers and provide me
the output.
Cheers,
Nitesh
Dean Collins wrote:
No they have a standalone
Nitesh,
They claim to support numbers on their website so I would say yes.
On 8/11/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Dean,
Can the LumenVox Speech Recognition engine understand numbers?
Sorry to ask stupid questions but kinda curious... as for my application
all I want is to the
Trevor G. Hammonds wrote:
Bill,
I am not aware of any commercial Asterisk-compatible cards that support
North American BRIs right out of the box. The best I have been able to come
up with was a card sold on eBay, where the seller promises to supply a patch
that needs to be applied to
OMG, someone thought that it's for real. Wow.
On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote:
C F wrote:
No you cant. This message is being dropped as well.
Shame. Seriously though I posted a new thread right after I posted that
reply. The reply showed up but the new thread still seems
C F wrote:
OMG, someone thought that it's for real. Wow.
I don't think so. Read the sentence carefully:
On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote:
C F wrote:
No you cant. This message is being dropped as well.
Shame. Seriously though I posted a new thread right after I posted that
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