Hi * users,
Is that possible to make voicemail audio file (that is attached to forwarding email) as MP3 file, rather than WAV?
TIA
Kuni
--
Kuniyoshi Murata
English-Japanese Interpreter Macintosh Webcast Specialist
[WebSite
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Macintosh Webcast Specialisthttp://www.macwebcaster.com
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Does anyone know how to redirect or pipe the processing of voicemail
sound file from
Asterisk to another application for what I want to do as described below?
Any input is welcome.
TIA
Kuni
-- Forwarded message --
From: Kuniyoshi Murata [EMAIL PROTECTED]
Date: Oct 22, 2005 8:54
to redirect or pipe the voicemail file processing from
Asterisk to another application.
Is there any reference site or any information?
Thank in advance.
Kuni
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Macintosh Webcast
separately disable guidance for
each key functions
Any input is welcome.
TIA
Kuni
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Macintosh Webcast Specialisthttp://www.macwebcaster.com
recorded of voicemail system is low.
Is there any parameters to fix the recording level of voicemail?
TIA
Kuni
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Macintosh Webcast Specialisthttp://www.macwebcaster.com
Andy Kuo writes:
Hi,
Maybe you can record the sound file vm-five.gsm as five hour in
Japanese, instead of just five.
AK
I don't think you can do that.
Because that vm-five.gsm can be used as message number also (e.g. message FIVE)
--
Kuniyoshi Murata.iChat
files? I mean,
asterisk-sounds-1.2.0-beta1.tar.gz, libpri-1.2.0-beta1.tar.gz and
zaptel-1.2.0-beta1.tar.gz
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Macintosh Webcast Specialisthttp://www.macwebcaster.com
Linux Box's default audio device. But, the default audio
device is unavailable.
Now, I think I want to disable Asterisk's access to console audio device
based on the logic above. How can I do that?
Thanks for any input
Kuni
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
Hi,
As I read http://www.areski.net/asterisk-meetme/about.php?s=0, meetme2
seems attractive to me. My question here is...
Can meetme2 and existing meetme can coexist and can be used whichever I want
when I want to have a conference?
Thanks for your input
Kuni
I want to have a single meetme conference room that interconnects H.323
video phone clients and sip/iax audio phone clients.
I have already set up for meetme to be shared by sip/iax audio phones and I
have just now installed open h323 stuff.
Regarding to this, I have several
Hi,
I'm using Asterisk-1.0.0 on Fedora Core 1
Date: Mon, 21 Feb 2005 14:22:13 +0900 [zone:Tokyo],
[EMAIL PROTECTED] mentioned in msg: Re: [Asterisk-Users] Asterisk
H323 support that ...
For channel asterisk-oh323-v0.6.5
need
Hi,
Is there any future possibility that Asterisk will be compatible with connection to 3G video mobile phone such as Nokia 7600, Nokia 6630 and many ohters in Japan, Europe and HongKong?
If this become possible, H.323 video clients and 3G mobile phone will be able to share video
Hi,
I'm thinking of setting up Asterisk for H.323 video phone clients.
Now, what is the difference between native H.323 that come with Asterisk and "Open H.323 for Asterisk" ?
TIA
Kuni
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
Hi,
Is MacOSX version yet to come?
Date: Fri, 28 Jan 2005 21:11:48 -0600 [zone:Chicago/Mexico City],
[EMAIL PROTECTED] mentioned in msg: [Asterisk-Users] iaxComm
version 1.0 released that ...
iaxComm is an Open Source softphone for the Asterisk PBX.
iaxComm
Hi,
I have following setup already.
PSTN call via zap channel is working, Xlite via sip channel is working, and conference call is working.
And here is what I want to do.
A. My friends are making conference call in a conference room and all clients are Xlite.
B.
Hi,
I know the following is mostly the issue of SER and I already posted the
same content to SER User list. Just for more input, I posted it to this
list. Sorry for the cross post for some people.
I've set up SER for UA to UA call.
I'm thinking of setting up SER to relay
Hi,
I'm now setting up a VoIP conference room using Asterisk.
All the clients are SIP phone (to be exact, Xlite), number of clients that should be registered are around 50 and concurrent users are maybe 15 clients at most.
So, basically I think I can handle the situation
Hi,
Is there anyone who created logwatch (which is an admin tool included in RetHat/Fedora) module for analizing Asterisk logs?
If there is, I would like to use it.
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter
Hi,
Does anyone know if it's possible to make Asterisk's Caller ID function to
be compatible with Japan's "Number Display" system?
TIA
Kuni
--
Kuniyoshi Murata.iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:[EMAIL PROTECTED]
Hi,
Date: Thu, 11 Nov 2004 08:42:16 +0200 (SAST) [zone:-], [EMAIL PROTECTED]
mentioned in msg: Re: [Asterisk-Users] quasi-skype channel for Asterisk?
that ...
On Wed, 10 Nov 2004, Kuniyoshi Murata wrote:
http://www.pcphoneline.com/skype
If I have a spare PC
http://www.pcphoneline.com/skype
If I have a spare PC-AT running Windows 2000/XP and use their devices to
convert skype's input and output to conventional phone jack, I guess I can
connect that to Asterisk and skype can be one of the channels.
Is my understanding right?
Hi,
Steve Kennedy writes:
On Wed, Nov 10, 2004 at 06:10:21AM +0900, Kuniyoshi Murata wrote:
http://www.pcphoneline.com/skype
If I have a spare PC-AT running Windows 2000/XP and use their devices to
convert skype's input and output to conventional phone jack, I
Hi,
I'm thinking of introducing Asterisk on Linux for IP PBX.
Now I'm using ISP that has VoIP service and I have VoIP terminal box for
that ISP and a SIP account for SIP server of the ISP.
Now, what I would like to do is the following.
A. Setup IP PBX on Linux by
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