Displaying 20 results from an estimated 9000 matches similar to: "how to improve sound file quality?"
2005 May 13
3
Audio quality
I'm a new Asterisk user. I've managed to set it up to do everything I
want except sound good. Currently, Asterisk sounds considerably worse
than my cell phone. I know VOIP can be _better_ than my cell phone,
because I've heard Skype do it. (Using 32k iLBC, I believe.)
I did an experiment with audio quality:
1) I made a recording which was pretty good. I used an iSight
2005 Jul 17
2
DNS SRV
I have added in my zone file;
_sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com.
As I understand it should mean that any sip connection to
<anyname>@elmit.com should go to the udp port 5060 at the host
vpb.elmit.com.
In Asterisk's extensions.conf I have in the context [default]
exten => ronald,1,Dial(${PHONE_615},60,tr)
exten => ronald,2,Voicemail,u615@office
exten =>
2004 Sep 18
9
No sound
Hello,
I have just set up an asterisk box (Debian unstable) and I would like
to test it with a H.323 application (gnomemeeting). When I call the
demo voice menu, I can't hear any sound. asterisk says that the
soundfile is played:
-- Executing BackGround("H323/ip$212.9.189.172:30005/29597", "demo-congrats") in new stack
-- Playing 'demo-congrats' (language
2005 Jul 28
12
Can you caculate with me?
before I accuse somebody to "overbill" I would like you to calculate
with me:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for the answers: 0.7269 or 0.2929 ???
bye
Ronald Wiplinger
2005 May 26
5
SIP Soft Video phone for Asterisk usage
I am looking for a SIP Soft Video phone, which I can use with Asterisk.
If you have one installed (regardless if free or purchased) please tell
me which one, the settings in Asterisk and your experience with it.
bye
Ronald
2007 Feb 14
5
Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the
bandwidth to different users. Each user should get e.g., 512kB/s plus
256kB/s dedicated for VoIP.
What kind of device can I use for that ? (managing switch ??? which one?)
bye
Ronald Wiplinger
2005 Jan 12
5
Grandstream Bugetone 101 & mwi
I tried to use message waiting indicator, by "Subscribe for MWI" in the
web menu of the phone.
However, it does not light up / flash, even if a voice mail is waiting.
Where is the switch to turn it to?
bye
Ronald
2005 Mar 06
3
SJphone on PDA registering with Asterisk???
I try to setup SJphone on my PDA, but it does not register with Asterisk.
I have setup a sip account on asterisk, ...
Can anybody give me a hint?
bye
Ronald
2002 Jan 10
2
-b flag at low sample rates?
As the subject implies, my question is: is it possible to use the -b (or
-M) flag at non-44K sample rates?
I'm working with an application that is trying to optimize for very small
audio filesize. I found that downsampling to 11K and then using q0 gives
high compression, but won't seem to drop below 64kbps or so. It seems like
the combination of downsampling, then reducing to 30kps
2006 Apr 26
6
I am looking for a webphone on MY SITE
I am looking for a way of not to install a softphone, preferable as a
link on a web site to a webphone on MY SITE !!!
Has anybody an idea for that? AJAX?
bye
Ronald Wiplinger
2005 May 17
4
Is SKYPE a threat or should we do something (together)
Skype is very succesfsfull and get more and more users, ... we can
ignore them, accept them or do something,...
My suggestion is that we try to do something, ...
If we would peer to each other, than we get soon also a great amount of
users together, and than our service becomes more valuable, ...
Let's discuss advantages and disadvantages!
bye
Ronald
--
Ronald Wiplinger (CEO of
2005 Jun 07
4
I want to move the MySQL server out to another machine
I tried to add the databases from the localhost to the database server
and changed the every /etc/asterisk/*.conf from host=localhost to
host=192.168.10.10
(my dababase server)
When I restart asterisk, I do not get any errors, but after a phone call
I see:
Jun 7 18:11:56 ERROR[7877]: cdr_addon_mysql.c:400 my_load_module:
Failed to connect to mysql database cdr on 192.168.10.10
Or if I try
2011 Mar 24
5
Sox and bad quality when converting to 8 kHz
Hi list,
I have an 44100 Hz file with human voice, stereo with 16Bit.
When convertig this to 8 kHz, mono I loose a lot of quality and have
some ground noise. I tried several sox options but without success.
Can somebody help....
best regards Thomas
2005 Jan 18
3
Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore)
I bought three plus two Grandstream BudgeTone 101 phones.
The shipping cost more than the phone itself from Pulver store.
The first shipping had one phone defect. Nothing on the display. (Can
happen!)
The second shipment had one phone with a defect display, but it still
worked.
The second phone's handset was defect too (microphone did not work).
Changing the handset from this one to the
2005 Jun 14
3
How to setup a test number to know my extension number
I would like to setup a test number, that speaks back my phone number.
How can I set this up?
bye
Ronald
2006 Apr 02
1
Who is on a call?
I would like to know which extension number is engaged in a call.
show channels shows me:
*CLI> show channels
Channel Location State
Application(Data)
SIP/asterisk.elmit.com-0 690@default:2 Up
Echo()
SIP/8807-066 690@newcontext Up Echo()
2 active channels
2 active calls
but it is not
2005 Feb 23
1
Sound files quality and volume
I just noticed that quality of .gsm files for using with asterisk is not
that good.. is there any way to make then sound better? asterisks sample
voices sound way better than theones recorded using applications like
wavepad or with asterisk like unavailable messages... any tips? Do you know
the command line for sox to adjust the volumen levels or gsm files (make the
louder)?
Also, do they have
2011 Sep 01
1
No buffer space available - loses network connectivity
Hi,
I have a centos 5.6 xen vps which loses network connectivity once in a
while with following error.
=========================================
-bash-3.2# ping 8.8.8.8
PING 8.8.8.8 (8.8.8.8) 56(84) bytes of data.
ping: sendmsg: No buffer space available
ping: sendmsg: No buffer space available
ping: sendmsg: No buffer space available
ping: sendmsg: No buffer space available
2006 Apr 10
2
Outbound calls through Broadvoice
Hi all, a noob here, I am trying to get outbound calls through asterisk
working with Broadvoice.
I have consulted the following two online tutorials:
http://www.broadvoice.com/support_install_asterisk.html
http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice
in an effort to make outbound calls.
My current settings are as follows:
sip.conf
register =>
2004 Nov 27
3
How to test if PCI 2.2?
Is there a way to test if the motherboard is ready for a Digium card
(PCI 2.2) ?
I would like to know from a remote computer, where I have (root) access,
if this computer is ready for a TDM22B.
bye
Ronald