People working on this have found that context influences the
pronounciation of words. I think the root cause of this is
that the human vocal tract cannot re-shape itself for different
sounds instantly and must move from the previous sound to the next
sound, we hear the movement. If it does
Also almost forgot. They sell the demo voices on their site for 29.99.
Linux and windows versions. Since I believe what they use is based off
festival, perhaps the voices could be made to plug into the existing
festival plugin for asterisk?
I have been working with app_festival for about a
I'm trying to Make a Goto inside a agi to another context/priority
I used SET CONTEXT callh323, SET PRIORITY 1, SET EXTENSION s
Apparently the SET EXTENSION is still assuming the value defined
initially(), what is not defined in the new Context.
Anyone has any turnaround for this?
Isamar
I must say this is basically correct
BUT
Remember that festival is actually based phonetically. remember
that and modify your text accordingly and you might be surprised at the
results.
yes the standard voices do suck !
On Tue, 15 Jul 2003 23:04:24 -0700 (PDT), Chris Albertson wrote:
Lubomir Christov wrote:
yes
put something like this in your extension.conf
it will route all calls started with 0 (it will send the numbers without
0) to phoneserve accounts
exten = _0.,1,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,)
exten = _0.,2,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,)
Lubo
yes, just tray it :)
Sergey S. Stasyuk wrote:
Lubomir Christov wrote:
yes
put something like this in your extension.conf
it will route all calls started with 0 (it will send the numbers without
0) to phoneserve accounts
exten = _0.,1,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,)
exten =
Thanks Matteo,
Now I have a backtrace if that will help. I am not a programmer and this
really means nothing to me. I can only tell you that I have a g723.1 encoded
file (conf-onlyperson.g723) in /var/lib/asterisk/sounds/ when this happens.
#0 0x08058291 in ast_write (chan=0x8111718,
On Wed, 16 Jul 2003 [EMAIL PROTECTED] wrote:
Has anybody tried Cisco 7960G? Or 7940?
sure, using them all the time here (the Skinny version, which requires
Cisco CallManager which in turn connects to asterisk via H.323).
The hardware (same as for the SIP version, in fact you can convert
Anyone wanting to go ahead and use VoiceMail2 will probably need this new version of
the addmailbox utility..
I have updated the addmailbox utility to create the correct directory structure
required by VM2 and copy the required files to the correct locations.. I just called
it addmailbox2..
I
Hi All,
I'm looking at getting some Cisco VoIP hardware to play with in
combination with a Asterisk server.
I've heard that there is beta software available to do SIP on the 7905G.
So, I'm thinking of either getting a 7905G or a ATA186.
My dillema is, which one to buy?
I can get both for
Hi,
I've been playing with Voicemail and Voicemail2 a bit for my users, and
there are a few things I'm wondering about:
- We can specify parameters to the mailbox (s, b or u) to select which
prompts to play. However, if we specify 'b' or 'u' it plays that
(customisable) message, but it also
hi,
Can anybody pls tell me, how to increase the time gap between 2 digits when you
transfer a call.
ie, the operator answers the call, and presses hash key to transfer, and then enters
the extension
number, some times, it timeouts too quickly before the operator enters the whole
extension
Hi,
Use s before u or b together with Voicemail2 app.
The default message will be skipped.
BR,
Dan
P.S. I think it works with Voicemail too
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 16, 2003 2:56 PM
Subject:
Are there any web based frontends for asterisk, for mananging voice mail etc
and asterisk in general?
Kind Regards,
Chris Bond
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As far as I know the Sip support for the 7905 has not been generally
released so comments you've seen on this list refer to test versions of the
code.
You can set up a call between two phones on an ATA186 through asterisk.
Iain
--On Wednesday, July 16, 2003 9:28 pm +1000 Steven Honson
Hey Florian,
On Wed, Jul 16, 2003 at 13:56:45 +0200, Florian Overkamp wrote:
Hi,
I've been playing with Voicemail and Voicemail2 a bit for my users, and
there are a few things I'm wondering about:
- We can specify parameters to the mailbox (s, b or u) to select which
prompts to play.
Thanks for your reply. I tried changing the constant DEFAULT_CIDRINGS to
2. However I still don't get callerid information. Is this what you were
referring to?
Thanks,
Hafeez
On Wed, 16 Jul 2003 15:33:47 +1000, Gary [EMAIL PROTECTED] said:
On Tue, 15 Jul 2003 17:44:56 -0800, hafeez bana wrote:
At 15:41 2003-07-15 -1000, Matthew John Darnell wrote:
Why hasn't someone found 50 people who sound alike, put them in sound
studios and record the 10,000 most commonly used words. You would all
differnent forms of the 1,000 most words, i.e. leading, trailing, question
etc.
You can synthesize the
Is the hassle in running it or setting it up?
This gets back to my interest in a CD to boot and install a basic system on a hard
drive.
Something like a 2 line 4 station version and then a single T1, 4 station, 2 line.
This is why there is a users list and a developers list.
As a user, I just
Hi,
Me too, but look at a similar thread started by me in the message
archive..:-)
BR,
Dan
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 16, 2003 4:45 PM
Subject: Re: [Asterisk-Users] voicemail instructions
Hi,
At 15:16
I'm in the process of setting up a test/demonstration system to show that
VoIP is realistic and applicable for our needs. We put a 7905 and 7960 on
a request for quote that went out the other day (to people like CDW
Microwarehouse). All of the vendors returned thier quotes without
including the
I asked [EMAIL PROTECTED] the other day. They wrote back:
US list retail price of BudgeTone SIP phones:
Model 101 $75/ea (available now)
Model 102 $85/ea (available now)
US list retail price of HandyTone VoIP analog telephone adaptor:
$75/ea (available in late July 2003)
Please
On Wed, 2003-07-16 at 09:10, jltaylor wrote:
Is the hassle in running it or setting it up?
This gets back to my interest in a CD to boot and install a basic system on a hard
drive.
Something like a 2 line 4 station version and then a single T1, 4 station, 2 line.
This is why there is a
On Wed, 2003-07-16 at 15:44, Marian Danisek wrote:
hello,
i found in list archives some notes about grandstream sip voip phones.
Does anybody succesfuly tested those phones with asterisk ? Mark ?
They seem to work with asterisk. I don't yet have a couple myself but on
irc there are people
In cdr table or in /var/log/asterisk/cdr-csv/Master.csv
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
[EMAIL PROTECTED]
Enviado el: miércoles, 16 de julio de 2003 23:54
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Dial SessionTime
Hi
I have been testing a couple of them for about 2 weeks now..
They are very good for the price..
The only issue that I still have is that the phone does not seem to be able to pickup
the time correctly from an NTP server that is not on the local network so the display
always shows 1900-XX-XX
Hi.
I'm experiencing a issue (not big, but important)
I have an asterisk installation with a buch of sip
phones analog ones.
I have 2 1 sip phone that's outside in the world,
and is nat'ed. I'm using g.729 with it.
I wanna use g.729 only for the remote phone, and ulaw
for the local ones, since
From: hafeez bana [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Wed, 16 Jul 2003 05:17:59 -0800
Subject: Re: [Asterisk-Users] X100P in Australia
Reply-To: [EMAIL PROTECTED]
Thanks for your reply. I tried changing the constant DEFAULT_CIDRINGS to
2. However I still don't get callerid
I have been using the ATA-186 with good success (with
exception of the fact that you have to recycle it from time to time). The one
thing I havent been able to do is to figure out how to make use of
parking, transfers, etc. My cordless doesnt seem to pass DTMF, so I
havent had any success
fixed in CVS thanks!
On Wed, 16 Jul 2003, Florian Overkamp wrote:
Hi,
At 15:16 16-7-2003 +0300, you wrote:
Use s before u or b together with Voicemail2 app.
The default message will be skipped.
Hmm, I was put off by the At most one of 's', 'u', or 'b' may be
specified. in the help
Thanks for your enthuastic response.
There's this Linux project out there for 802.11 at:
www.station-server.com
They have figured out how to make this type of distribution package work.
Don't get me wrong, Asterisk seems to have just about everything from a feature
standpoint. The open source
On Wed, 2003-07-16 at 11:47, jltaylor wrote:
Thanks for your enthuastic response.
There's this Linux project out there for 802.11 at:
www.station-server.com
They have figured out how to make this type of distribution package
work.
And there is nothing stopping you from getting a knoppix CD
Mark,
Any news on the enhanced queue app progress? Just wondering.
Jim Friedeck
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--- Moshe Yudkowsky [EMAIL PROTECTED] wrote:
SNIP
The real trick is to get the correct posidy. Here's three sentences
with
the same words but each with different prosidy:
I said 'yes.'
I said yes?
_I_ said '_yes_'???!!
Both formative and concatenative systems add prosidy. Adding
Hi,
I'd like to have a SIP phone at home and at the office and have them both
ring when my extension is dialed. Right now I used the same config for the
phones (Cisco 7960's). So they both register with the same login pw. This
doesn't seem to work quiet right, where only the last phone to
I'd like to have a SIP phone at home and at the office and have them
both
ring when my extension is dialed. Right now I used the same config for
the
phones (Cisco 7960's). So they both register with the same login pw.
This
doesn't seem to work quiet right, where only the last phone to
On Wed, 2003-07-16 at 12:20, Justin Eckhouse wrote:
Hi,
I'd like to have a SIP phone at home and at the office and have them both
ring when my extension is dialed. Right now I used the same config for the
phones (Cisco 7960's). So they both register with the same login pw. This
doesn't
At 10:11 2003-07-16 -0700, Chris Albertson wrote:
SNIP
if you want a synthetic voice to sound
natural you will have to tell the software the _intent_ of the words
not just the words. You would need a markup language for that
emph I /emph said quotequestionword yes /quote/questionword
The W3C has
Moshe Yudkowsky wrote:
At 10:11 2003-07-16 -0700, Chris Albertson wrote:
SNIP
if you want a synthetic voice to sound
natural you will have to tell the software the _intent_ of the words
not just the words. You would need a markup language for that
emph I /emph said quotequestionword yes
I'm trying to get my newly flashed 7960's to play nice with Asterisk, but
I'm having some problems. I can get my 7960 to register with the proxy,
and if I dial my own extension, according to my dial plan asterisk should
transfer me to voice mail. Asterisk thinks its playing me voice mail
prompts,
Hi,
I'm running asterisk in the following setup
Phone - WX100USB - * - Internet - * - WX100P - PSTN
The two Asterisks talk to each other via IAX2 and use GSM for voice.
This seems to work quite well except for occasional pauses in voice
transmission. These seem to occur in _one_ direction only
I'm working greatly with 40+ Grandstream phones. Audio quality is good
enough for production environment, the cost is really low and the
configuration is *Really* easy.
But a little answer to Wipeout is:
The only issue that I still have is that the phone does not seem to be
able to pickup the
Hey Jan,
On Wed, Jul 16, 2003 at 11:45:13 -0700, Jan Rychter wrote:
Hi,
I'm running asterisk in the following setup
Phone - WX100USB - * - Internet - * - WX100P - PSTN
The two Asterisks talk to each other via IAX2 and use GSM for voice.
This seems to work quite well except for
turn off jitterbuffer in both servers.
aka
jitterbuffer=no in iax.conf
jitterbuffer, unfortunately, is buggy and don't work
as expected.
Il mer, 2003-07-16 alle 20:45, Jan Rychter ha scritto:
Hi,
I'm running asterisk in the following setup
Phone - WX100USB - * - Internet - * - WX100P -
Hello,
I'm looking at getting some Cisco 7910G+SW phones through a local
liquidator.
Is anyone currently using these with asterisk?
Any special limitations with this one as compared to the 7940 and 7960's?
Will they work with SIP or are they stuck in Cisco-land with Call-Manager
only?
On Wednesday 16 July 2003 03:52 pm, Greg Renouf wrote:
Grandstream can improve the quality of their 'user interface' (many others
have already accomplished this goal,) I can see very few situations where
the $10-20 cost saving will make the quality sacrifice worthwhile.
What other phones are
Have you tried to mantain the default ntp server on your phone? (the *.gov
one)
I normally use internal ntp servers but in a particular context i've used
that ntp server and it worked perfectly.
I have tried many public NTP servers and all have the same result..
Could be a Firewall
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 16, 2003 10:19 PM
Subject: Re: [Asterisk-Users] grandstream sip phone
I have tried many public NTP servers and all have the same result..
Wait.
I have tried many public ntp too. It
In iax.conf, do you have jitterbuffer set,
try jitterbuffer=no, or try some values from 1-5, i use 3
-Original Message-
From: Jan Rychter [EMAIL PROTECTED]
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Date: July 16, 2003 11:45 AM
Subject: [Asterisk-Users] IAX pauses
Hi,
I'm running
Other things:
The phone constantly says Ethernet Disconnected. Even though it
tftp's
configs and registers with the proxy.
Something is wrong with either the phone hardware itself, the network
port on the hub or switch it is attached to, the ethernet cable it is
connected with, or the 10/100
This phone was working with the same cable/switch when used with a Skinny
configuration. But I'll try your suggestion out just to make sure when I
get home tonight...
-Steve
On Wed, 16 Jul 2003, John Laur wrote:
Other things:
The phone constantly says Ethernet Disconnected. Even though it
Matteo == Brancaleoni Matteo [EMAIL PROTECTED] writes:
Matteo turn off jitterbuffer in both servers. aka jitterbuffer=no in
Matteo iax.conf
Matteo jitterbuffer, unfortunately, is buggy and don't work as
Matteo expected.
Interesting -- this has indeed helped and the quality is better, too!
Dlink has the dhp-90 (currently in limited release like Grandstream) for
$60-70. It doesn;t have a digital display- but it works flawlessly.
There are a few others- you just need to look around...
-GSR
- Original Message -
From: marrandy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Il mer, 2003-07-16 alle 23:04, Jan Rychter ha scritto:
Matteo == Brancaleoni Matteo [EMAIL PROTECTED] writes:
Matteo turn off jitterbuffer in both servers. aka jitterbuffer=no in
Matteo iax.conf
Matteo jitterbuffer, unfortunately, is buggy and don't work as
Matteo expected.
Hi All,
I got a doubt about something I want to do with asterisk. I have this
office (site a) with only a Panasonic analog PBX and another office
(site b) with an Asterisk Box with an ADIT 600 . I want to interconnect
both via IAX. Is it possible to put a new asterisk box in site a
It sounds like you want to use the IAX to provide dialtone to the
Panasonic PBX? You'd use FXS cards in the asterisk box to provide signal
into a CO port on the Panasonic.
On Wed, 16 Jul 2003, Iván Aponte wrote:
Hi All,
I got a doubt about something I want to do with asterisk. I have this
Hi,
I'm trying to interconnect sip and h323 endpoints using asterisk
and asterisk crashes with segmentation fault whenever h323
connection needs to be established. It registers with gatekeeper ok though.
Here are the symptoms.
If the call initiated by SIP device, asterisk replies to it Trying
- Original Message -
From: Anthony Wood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 03, 2003 6:04 PM
Subject: Re: [Asterisk-Users] Voice Modem + Soundcard Driver
On Tue, Jun 03, 2003 at 05:47:54PM +1000, Mathew Frank wrote:
Woody wrote:
The problem with using
Title: Back-to-back connected boards load test
Hi,
Two asterisk boards in two asterisk boxes are connected back-to-back with crossover cables.
T400P (box1, pri_net) creates a flow of calls, TE410P (box2, pri_cpe) takes them. Cables are OK.
Boxes are powerful. SMP is in place. Interrupts are
I want to setup 2 asterisk boxes
in 2 different statesto make long distances calls.. I know I need to
get a TDM400P 1 port FXS and a X100P
I think that I can use IAX to connect them over the internet
I want to be able to pick up the phone in state one, and have it pick up
the line
Ok so no more ethernet disconnected errors. However, I still am getting
the same voice mail errors. (BTW, don't plug the cable into the wrong port
on the back... sigh)...
The phone starts the call and then immediately disconnects. I've included
the * console, and tethereal dumps.
Any Ideas?
Figured it out...
sip.cnf:
change:
allow=all
to:
allow=ulaw
ta da!
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Hi,
I have been trying to workout how to use the call pickup.
So Far, I have the following in zapata.conf
[channels]
signalling = fxo_ks
context = local
pickupgroup=1
callgroup=1
channel = 1-3
When I dial *8# all I hear is busy tone.
What have I missed?
thanks
Jay.
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