Hi,
I want to develop a IVR application that repond to speech input from the
caller in asterisk.
For example, imagine a caller who wants to speak with Ram Kumar. On a
traditional IVR/auto attendant, the caller may be entering “76484” to spell
“Kumar” and the system may respond with: “Press 1
Things that look simple r quite complex to build :-)
Indian Accent ASR on proper names is herculean task.
No speech recognition known to mankind as of date can handle so many
dialects being spoken in India, so in short what you want is nice to have,
but nearly impossible to develop.
Better try
ok,how can i develop with short vocab like sales,support,etc.
I have read many article but I'm not able to pick the right point, how can
i develop or configure speech reorganization with asterisk.
Is there any article or link please share and guide me.
Regards
Akhilesh
On Thu, Jul 5, 2012 at
Greetings,
I know this is not a Elastix mailing list, but could anybody please tell
where I can download Elastix 2.3.0.1 (the latest version) ?
There is only version 2.3.0 (April 2012) on Elastix website, not the
2.3.0.1 (May 2012), but the changelog information are there.
Thanks in advance.
4 jul 2012 kl. 13:32 skrev Elliot Murdock:
Hello,
I am trying to get clarity with the sip.conf timer configuration. The
current configuration states:
;--- SIP timers
; These timers are used primarily in
dear
please Help. I am continously getting this message after sip set debug
on. and not getting clear voice from both side.
--- Transmitting (NAT) to 122.163.193.94:1893 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.106:5060
Hi,
*CSeq: 245 OPTIONS *
*
*
This is just SIP keep-alive. It has nothing to do with any Call-media
degradation. If you are not getting clear voice check the codecs, network
latency/delay/loss/jitter parameters.
BR
Sammy
On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava alok...@gmail.com wrote:
previously i was using for codec
allow=all
after that i changed
disallow=all
allow=silk24
and i also change softph x-lite from jitsi(because of codec)
now voice was coming fine from both side.
But when i came to home from office not getting voice from both side.
Threr is Airtel Broadband at my
Put disallow=all below all of the allow=
--
_
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Hi,
Now and then, I'm facing environments in which it could be helpful to
integrate building intercom systems with Asterisk.
Those intercom systems are made of :
- a main panel, showing company names and equiped with a speaker, a
microphone and an optional video cam
- a doorstrike
- several
Le 05/07/2012 15:21, Olivier a écrit :
[...]
Would you say it's possible to install several intercom phones on the
same line, both ringing at the same time (but only one of them being
able to answer) ?
If positive, is it possible to connect one intercom line to asterisk
and let an asterisk user
Hi,
I'm curious about the availability of Multi Function Printers with the
following feature :
- user feeds paper sheets in
- user dials a phone number (0123456, for instance) then a hits single button
- the result is that the paper sheets are scanned into a file which is
emailed to a given
El 05/07/12 02:19, Satria Anamarta escribió:
Greetings,
I know this is not a Elastix mailing list, but could anybody please tell where
I can download Elastix 2.3.0.1 (the latest version) ?
There is only version 2.3.0 (April 2012) on Elastix website, not the 2.3.0.1
(May 2012), but the
T.37
http://en.wikipedia.org/wiki/T.37_(ITU-T_recommendation)
There were some scanners manufactured with this in mind, however I
cant remember who made them.
On Thu, Jul 5, 2012 at 10:15 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I'm curious about the availability of Multi Function Printers with
I searched a bit more,
http://www.muratec.com/catalog/F320_config.html#email
The above model supports t.37 but no sure if you can have it function
such that any number entered will actually be send to a gateway.
On Thu, Jul 5, 2012 at 10:20 AM, C F shma...@gmail.com wrote:
T.37
Il 05/07/2012 15.21, Olivier ha scritto:
Now and then, I'm facing environments in which it could be helpful to
integrate building intercom systems with Asterisk.
Is there a standardized protocol available to run voice, video and
command on 2-wires and most probably used by intercom systems
Hi Shitian,
here's my sip.conf, but unfortunately I cannot make some other tests
with Asterisk 1.8 since the PBX is in production now with Asterisk
1.4.26.2 which seems to work very fine.
Thank you
G
NOTE: tried to change nat and canreinvite parameters but with no success.
[general]
Il 05/07/2012 16.43, gincantalupo ha scritto:
here's my sip.conf, but unfortunately I cannot make some other tests
with Asterisk 1.8 since the PBX is in production now with Asterisk
1.4.26.2 which seems to work very fine.
I'm using the same provider on many sites without special issues.
My
Hi All,
It's small issue but making a big problem for my application. I have CentOS
release 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41
because Flite work in this version.
problem is that when I make changes on .call file to make it future call
file with *touch *command then it
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones.
These releases are available for immediate download at
On 07/04/2012 01:47 PM, sathiish kumar wrote:
Thanks for the response.. I did change it in the [general] settings.My
setup is something like I have a remote conference (not meetme) which
will send reinvite to redirect the RTP flow to a different server to
load balance.There are three clients who
I am new. Here is the code that I am playing with on CentOS 6.x
When I dial the number that corresponds w/ my SIP account I get a
recording: reached a non-working number
I built Asterisk a few times last year and am now back working on a similar
project. In my view, there is
- Original Message -
I am new. Here is the code that I am playing with on CentOS 6.x
register = 5552530146:funnytiger...@sip3.voipvoip.com
[outgoing]
username=5552530146
type=peer
qualify=yes
secret=funnytiger123
nat=auto
insecure=very
host=69.90.209.57
fromuser=5552530146
Hi,
I changed these codes to not coincide with actual account info.
Thanks
On Thu, Jul 5, 2012 at 5:48 PM, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
I am new. Here is the code that I am playing with on CentOS 6.x
register =
On Wed, 2012-07-04 at 10:15 +0530, Chandrakant Solanki wrote:
So, is http://sourceforge.net/projects/aterisk-amr/files/ same patch
also works in 1.8.13.0??
On Wed, Jul 4, 2012 at 3:18 AM, Hans Witvliet aster...@a-domani.nl
wrote:
On Tue, 2012-07-03 at 17:13 +0530, Chandrakant
Hi All;
If I set a context other than the default context, then I do not see a
generation for a configuration in the extensions_additional.conf for this
context, but always the generation for the configuration is for the default
context (from-internal).
Normally, I have to put some Phones in
Dears;
In FreePBX, when I select voicemail for the extension, and if the caller sent
for the voicemail, and he leaved (or did not leave) a voice message, and did
not press #, so the channel will stay open and this is not good specially if
the call was coming from outside via the analoge lines
Asterisk, and by extension FreePBX, automatically end the voicemail recording
when the caller hangs up. You have some OTHER issue. Perhaps Asterisk is not
detecting the hangup?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hey,
If you want to have all the dialplan features for your extensions and still
need to implement some outbound calling restrictions then you need to look
for some modules in freePBX. i've used that module exactly for this purpose
and it works..can't remember its name.
Just google it or lookup
The module is custom contexts - its a third party option in the module admin
But you can write contexts in the extensions_custom.conf if you want to
I wouldn't use freepbx to generate your code - its quite complex code for a
roll your own system, but very useful if you learn its gui and options
On Thu, Jul 5, 2012 at 5:20 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
You can get modules to do what you're looking for, but if you really want
to make a custom context but still have all the available features of the
default context, you can add the following at the end of your
umm Warren, yes including from-internal is the way of getting all the
features,,,but in my experience the calls going out using the dialplan
script we manually enter in our custome context don't get inserted into the
FreePBX CDR and recording stuff !!
On Fri, Jul 6, 2012 at 10:01 AM, Warren
On Fri, Jul 6, 2012 at 12:11 AM, SamyGo govoi...@gmail.com wrote:
umm Warren, yes including from-internal is the way of getting all the
features,,,but in my experience the calls going out using the dialplan
script we manually enter in our custome context don't get inserted into the
FreePBX
Hi,
Did you get anything working on it !! See the permission for the user
running asterisk process and see if that user can touch files like that.
Regards,
Sammy
On Thu, Jul 5, 2012 at 10:47 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
It's small issue but making a big problem for my
Thanks Gohar,
I found the issue was copy file to outbound folder not moving. that's why
after making future time asterisk start reading file.
On Fri, Jul 6, 2012 at 11:16 AM, SamyGo govoi...@gmail.com wrote:
Hi,
Did you get anything working on it !! See the permission for the user
running
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