Displaying 20 results from an estimated 10822 matches for "voiced".
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voice
2002 Jul 12
2
Lattice help (again?)
...elconstr",groups=voicing)
Some information on the data set:
> sapply(sampledata,levels)
> $votms
> NULL
>
> $age
> NULL
>
> $Cplace
> [1] "bilabial" "dental" "velar"
>
> $tokentype
> [1] "CC" "voiced C" "voiceless C"
>
> $voicing
> [1] "voiced" "voiceless"
>
(The entire data set follows below..)
Please, help me with this. The solution is probably an easy one..
/Fredrik
> sampledata
> votms age Cplace tokentype voic...
2006 Apr 21
1
How to select Ceptral's Voice in Asterisk's Swift application??
Hi,
I'm using Cepstral as a TTS Engine for Asterisk with Swift application.
It works fine when I have just 1 voice installed. Now I have 2 voices in
the same language installed but I can't seem to find the way to select
which voice to use in Swift's application in Asterisk. Does anyone know??
Thank you,
Pim
2004 Aug 15
0
how can i config a Cisco IAD 2430 config as a sip client
Hello,
I have a cisco ATA 188 registering both of its lines
to * I can place calls between then an to kphone an
MSN messenger (both registering with * too), a few
days ago a friend lend me a Cisco IAD 2430 and I was
willing to do the same thing with it, since it has 24
ports I was willing to to use 24 analog phones with it
however something really weird happens I can place
calls from my ata,
2006 Jun 12
0
ICLID or CNAM calling name and number through a cisco isdn gateway
All,
I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is....
One pri terminating into a Cisco 2431 router
Sip messages from the Cisco get sent to a asterisk server
linksys ata's a each remote end.
I can receive the calling name if the call originates
2004 Dec 17
2
OT: "Integrated Access T1" voice problems -is this possible?
> Mark Farver wrote:
>
> > On Fri, 2004-12-17 at 16:26 -0600, Kristian Kielhofner wrote:
> >
> >> We are getting pricing and one provider is telling us
> that they have
> >>quality issues with the "Integrated Access" product. From
> what they
> >>say it sounds like you can have audio dropouts on the voice
> channels
>
2006 May 25
2
Volume configuration on Polycom Soundpoint 501phone
Could not find your post for 4 months ago.
-------------- Original message --------------
From: "Anton Krall" <akrall-lists@intruder.com.mx>
> Yes, check a post that I made about 4 months ago, I posted the cofig for
> setting the speaker, handset and ring volumes ..
>
> |-----Original Message-----
> |From: asterisk-users-bounces@lists.digium.com
>
2005 Jul 27
0
Polycom gain settings
Hi All,
I have some Polycom IP300's and I'm interested in increasing the max volume
for the headset (not handset), I'm wondering if anyone has experience
adjusting these values:
<gains
voice.gain.rx.analog.handset="0" voice.gain.rx.analog.headset="0"
voice.gain.rx.analog.chassis="3" voice.gain.rx.analog.chassis.obs="-12"
2019 Apr 08
2
pjsip endoint woes
On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:
> On 4/5/19 10:36 AM, sean darcy wrote:
> > I'm trying to set up pjsip to work with an obi202 and google voice. But
> > I can't configure the endpoint.
> >
> > pjsip:
> >
> > [obi202-auth](!)
> > type = auth
> > auth_type = userpass
> > password = <mypass>
> >
>
2003 Dec 03
2
Cisco IAD with MGCP
I repost a message I put a week ago:
I have a Cisco IAD 2431 which has MGCP protocol; I cannot make
it to work againts Asterisk; at least there is some MGCP conversation
between them but when I offhook a phone attached to IAD I get no tone at
all.
As anybody managed to get working Asterisk against an MGCP Cisco
gateway ?
Which MGCP version should I use ?
Also I recently
2019 Apr 05
2
pjsip endoint woes
I'm trying to set up pjsip to work with an obi202 and google voice. But
I can't configure the endpoint.
pjsip:
[obi202-auth](!)
type = auth
auth_type = userpass
password = <mypass>
[obi202-aor](!)
type = aor
max_contacts = 2
; ===== endpoints ========
[gv-voice](obi202-endpoint)
auth = gv-voice
aors = gv-voice
identify_by=auth_username
;identify_by=username ; I also tried
2004 Dec 17
2
OT: "Integrated Access T1" voice problems - is this possible?
Hello,
I am currently pricing out various T1 and PRI options for a client of
mine. We need voice and data - we want T's. Whether it be two seperate
T's, two superate fractional T's, or one combined fractional T, we need
it done.
We are getting pricing and one provider is telling us that they have
quality issues with the "Integrated Access" product. From what they
2003 Nov 09
1
chan_capi & Eicon Diva problem
Hello,
I have an issue getting the chan_capi module to load in asterisk cvs
from today. Plain 2.4.20 kernel with melware patches for the Eicon Diva
Server Bri card.
I load the modules with: modprobe -v divas divacapi
I load the firmware with: divactrl load -c 1 -f ETSI -vd6
Output in /var/log/messages is:
Nov 9 19:26:26 voice kernel: Eicon DIVA - DIDD table
(http://www.melware.net)
Nov 9
2006 Apr 10
1
"chan_iax2.c: Ooh, voice format changed to ..."
Can someone explain me this message:
"chan_iax2.c: Ooh, voice format changed to ..."
Where can I find a list of numeric codes used to identify voice format?
Then, sometime I get an infinite loop of messages like these:
DEBUG[15015] chan_iax2.c: Ooh, voice format changed to 1
WARNING[15015] channel.c: Unable to find a codec translation path from g723
to alaw
DEBUG[15015]
2008 Jun 20
1
Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to
2014 Feb 14
2
Want Queues to ignore mobile operators voice mails and continue ringing...?
Hi All,
Lets say I want to setup a queue that will handle inbound calls to
dynamically added agents that are all mobile numbers. Now when I do this
setup it works, it loads the agents dynamically and if the mobile phone
is on and have reception it works. But when the phone is for arguments
sake off or dont have reception it goes to voice mail for that mobile
phone.
I don't want this to
2008 Feb 05
4
Cannot hear voice through SIP Phone from one side
I have a asterisk server. Two SIP Soft XLites are connected to the server. I am able to make
calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice.
They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice.
The microphone and speakers are
2004 Sep 03
7
Dropping incompatible voice frame
Hi: i have a problem.
Mi extensions.conf:
exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN})
exten => _N.,2,SetAccount(${customer})
exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1})
exten => _N.,4,ResponseTimeout(5)
exten => _N.,5,Background(ifyou)
exten => _N.,6,Background(silence/1)
exten => _N.,7,Background(ifyou)
exten => _N.,8,Background(silence/5)
exten
2017 Apr 27
3
SIP and Voice on different nets
?I have connection with two networks (by VoIP provider setup)
1 - 10.10.10.0/24 = SIP
2 - 10.10.11.0/24 = Voice
How to tell Asterisk send / receive voice traffic not on SIP network. When
I look into dumps, I see Asterisk trying to use SIP net for voice
Unfortunately, I _need_ to use two networks instead of one?
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2007 Mar 18
1
Choppy sound with chan_capi + Fritz Card USB
Hi everybody,
I have a problem which I cannot eliminate on my own. Has anybody any idea
for the following:
I am using the asterisk-version from Debian-Testing (1.2.13) with the
latest chan_capi (also tried an older version).
When using the Capi-Channel, everything works fine except from the sound
it sounds extremely choppy and is unusable :-(
When e.g. capisuite is used for fax, everything
2014 Sep 18
1
Voice-Recognition / ASR / with barge in
Hi there,
I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine
:-) But I am wondering if there is a solution/application which will
enable me to implement voice recognition while playing a voice file
(barge in). So that the caller hears a voice file and can interrupt it
with his voice.
Currently (on our platform) the caller has to wait for the end of the
voicefie. Then we play