Date: Mon, 21 Feb 2005 00:20:39 -0800 (PST)
[zone:-], [EMAIL PROTECTED]
mentioned in msg: Re: [Asterisk-Users] Asterisk
H323 support that ...
with Openh323 - v1.12.2 and pwlib - v1.5.2 I use
asterisk-oh323 v.0.6.3b and it works fine
What version of Asterisk are you running? And on
Hi!
I have a runing asterisk box and i want to dial to a analog. pbx using a
4FXS Welltech. Let's say that my pbx have no. 700.
If i want to dial to a person in that direction i have to dial pbx
prefix (ex. 700), wait for pbx to ansear with hello message and after
that to dial internal
I've been unable to resolve *.asterisk.org and www.digium.com for the
last several hours... as far as I can tell it's not limited to my
location.
Needless to say... lists.digium.com does resolve..
Jared
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[EMAIL PROTECTED] wrote:
I'm having problems with international calling via Global
Crossing. I'm told I need to send a true ani versus a sudo ani.
What is the difference and how can I configure asterisk to do this.
Global Crossing is denying calls with sudo anis.
I'm wondering if they
On Tue, 22 Feb 2005, Ginel Tudorache wrote:
I have a runing asterisk box and i want to dial to a analog. pbx using a
4FXS Welltech. Let's say that my pbx have no. 700.
If i want to dial to a person in that direction i have to dial pbx
prefix (ex. 700), wait for pbx to ansear with hello
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
On Tue, 22 Feb 2005 09:07:43 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
I registered at a few sip server in different countries
Now I want to route outgoing calls for that country threw that sip
server and
When adding the details in AMP for when caller dials 3, I have
referenced it using 'custom-myapp,s,1', and if I go to
'extensions_additional.conf' I see the following line under the rest of
menu item info that was created :
exten = 3,1,Goto(custom-myapp,s,1) ;
and in the
On Mon, 2005-02-21 at 14:56 -0500, Jared Watkins wrote:
I've been unable to resolve *.asterisk.org and www.digium.com for the
last several hours... as far as I can tell it's not limited to my
location.
Needless to say... lists.digium.com does resolve..
Must be your DNS setup.
Maybe you
Hi,
I have the following redirection scenario:
- incoming call on oh323 - incoming DNIS is passed to an AGI script - the script searches in a table and returns the new destination - * dials out on h323 using the new destination
The redirection itself it's working but nothing is written in the
What phone are you using when calling? Does it have silence supression
on? Try disablig it... It could be a timing issue.
Julian.
On Mon, 21 Feb 2005 12:27:54 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
Brian:
Found the MOH random answer on the wiki, you were right... All the basic
stuff
Title: Segfault when using res_config_odbc on x86_64
I'm trying to move our asterisk setup from an i686 server to an x86_64 (Dual AMD Opteron) server.
Everything has been manually compiled: MySQL 4.1.10, MyODBC 3.51.11, unixODBC 2.2.10 (because I couldn't find any usable RPMs). And
Yes, one of the supermicro motherboards will work fine with both TE10P
and TE05P (not sure of the specifics 6014P-8R) just make sure you
look out for the slots parameters as follows:
for TE10P the board must have a 32-bit PCI -slot with 66Mhz speed
bus and importantly uses 3volts.
(if
I have 2 Zyxel Prestige and I m happy with them. In the beginning Its not
very easy to use, but when you get used to It, Its nice and easy. The batery
lasts long.
He isnt so good behind NATs.
Joao
- Original Message -
From: Kurt Fankhauser [EMAIL PROTECTED]
To: 'Asterisk Users Mailing
We have sucessfully got a NAT traversal box going using SER and
mediaproxy. What it does is make the natted UA's apear to be
registered at the IP address of the NAT traversal box and then rewrites
the packets accordingly.
[EMAIL PROTECTED] wrote:
Hi,
We're deploying a small VoIP solution for a
Dragos Ungureanu schrieb:
...
The redirection itself it's working but nothing is written in the CDR
Hi,
include
amaFlags=billing
and maybe
accountCode=AN_APPROPRIATE_ACCOUNTNAME
into your oh323.conf!
Roger.
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This noise problems occur between a SIP ATA connected to a wireless analog
phone and a user that was called using a ZAP (x100p) interface. There is no
firewall in betwwen except for a local 2wire router with NAT, but since We
(the ata and asterisk server) are on the same network behind the Nat
Im using an analog phone connected thru a grandstream handytone 286 ata
No clues if it has silence suppresion though but seems the hickups have
become minimal now... Weird since I didn't change anything but if they show
up again Ill check the silence suppresion config.
Thx!
-Original
Hi guys,
which typo of access point you are preffere? Is there any that support
roaming between areas without interruption of existing SIP call?
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Hi all... I have a slight problem with getting speex running
I Downloaded Speex sources (v. 1.0.4 stable version) and did make; make
install sucessfully
Then I re-maked the asterisk sources and clearly saw a speex.so module
being built (so the makefile for sure detects that there is a speex lib
what does the parameter -vvvgrcmeanand are there any others as well?KindestMuhammad Muzzamil Luqman
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I have the following configuration and am obviously missing something
small that is causing * not to work as expected.
I have the following defined in sip.conf
[ccme-in]
type=peer
host=10.0.9.1
context=devel_in
disallow=all
allow=alaw
nat=no
canreinvite=yes
qualify=yes
and [devel_in] is defined
Asterisk 1.0.3, Copyright (C) 2000-2004, Digium.
Usage: asterisk [OPTIONS]
Valid Options:
-V Display version number and exit
-C configfile Use an alternate configuration file
-G group Run as a group other than the caller
-U user Run as a user other than the caller
On Tue, 22 Feb 2005 15:41:59 +0500, Muhammad Muzzamil Luqman
[EMAIL PROTECTED] wrote:
what does the parameter -vvvgrcmeanand are there any others as well?
You'll find this and very many other useful pieces of info in the
Wiki, which also is very well indexed by Google.
asterisk -r attempts to connect to a running asterisk process (rather
than starting another one)
-v means be verbose (the more v's the more verbose)
-c provide a control console for asterisk
-g remove resource limit on core size - a debugging thing maybe?
To find all this out for yourself and
Hi,
There seem to be some codec incompatibility.
On *, you define alaw and you set ulaw on the Cisco.
Set both to same or add the other codec on (at least) one side.
Try if that solve it
Ex:
Add allow ulaw on * after the allow alaw
And / or
Add codec g711alaw on Cisco above the codec g711ulaw
hello
i was using CVS Head version for realtime mysql it was
working well. now i want to use odbc connection for
realtime database it is not working i am using it with
stable release. i have checked everything my conf is
ok odbc connection is working. any one working with it
res_conf_odbc.conf
Extraction from:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime
RealTime requires CVS-HEAD. If you attempt the following with Asterisk 1.0.3
or earlier, it will not work.
Ariel Pablo Klein
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi
Is it, or could it be possible to gateway from ISDN videophones to IP
videophoning with asterisk using libpri/zaptel etc?
roy
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To
hallo,
does sombody know how to get the callerid from iax.conf ( callerid=name
1234) via the manager interface?
Action: IAXpeers
gives only the Name/Username but not the call number?
any ideas how to do this?
thanks,
alex
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Thanks Roger, amaFlags solved my problem
Dragos
On Tue, 2005-02-22 at 12:10, Roger Schreiter wrote:
Dragos Ungureanu schrieb:
...
The redirection itself it's working but nothing is written in the CDR
Hi,
include
amaFlags=billing
and maybe
google for inalp isdn sip gateway
On Tue, 22 Feb 2005 12:23:39 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
Hi
Is it, or could it be possible to gateway from ISDN videophones to IP
videophoning with asterisk using libpri/zaptel etc?
roy
I wish to initate calls from a web interface, by clicking on a link and then
connecting to the automatic outgoing call by picking up an analogue phone.
I've got one fxs and one fxo and I wish to automate the call using a call
file (which I can do now). How can I pick up a handset and connect to
Hi list,
I would like to use the * VMS application with a GSM network, I know
that * support Unavailable and Busy Redirecting Reason in the
extensions.conf but, what's about the No Reply ??
Then, I know that doing a debug on the PRI, I see the redirecting reason...
My question is; is there
This same topic comes up about every month or two, and the exact same
words are used over and over again. The last run at this was on the -dev
list about one/two months ago and shouldn't be hard to find.
If memory serves anywhere near correct (which is a stretch), lots of
folks agreed
while using iax and a soft phone, the sound of breathing comes through
so clearly that it has started bothering me. Earlier I was amazed at
the quality, but now feel it is irritating. Wondering if there is a
way to cut it down. I am in the process of exploring using iax for a
call center, but
Hallo All
I been looking at all list and i know this been
discussed millions of times
for incomming calls
I need to send the dialled number to the pbx so that
the correct extension
can be rung on the pbx side
current setup:
pbx ext-- fxs asterisk -- asterisk fxo ---co
line(pbx).
What are
Don't know for sure, but it happens on the TDM card presumably due to
asterisk translating certain voices into dtmf. Busycount=6 corrects
the problem on it; presumably it would on a T1 as well.
Would enabling Busydetect really help if Asterisk thinks it detects an
it's working. Thank you!
Ginel
Peter Svensson wrote:
On Tue, 22 Feb 2005, Ginel Tudorache wrote:
I have a runing asterisk box and i want to dial to a analog. pbx using a
4FXS Welltech. Let's say that my pbx have no. 700.
If i want to dial to a person in that direction i have to dial pbx
prefix
Yes stop using asterisk for phone sex !!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hariharan
Gopalan
Sent: Tuesday, February 22, 2005 6:57 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Sound of breathing
while using iax and a
Would I be able to achieve this:
Outside call to Agent A.
Agent A puts call on hold.
Agent A Speaks to Agent B.
Agent A transfers call to Agent B.
In the cdr, I would like records like
callA
AB
callB
In other words, 3 cdr records. One for initial call to agent a, one for
agent a to
Whenever some call comes in i want it to be
automatically picked up and then it plays some message "Welcome to
xyz,Press 1 for sales and 2 for support" and then it takes it to the
particular extension of sales/support.
can i achieve this thing using
asterisk?
Kindest
Muhammad Muzzamil
I've noticed that too (its not just when having phone sex either! :-).
It depends on the phone being used (or is that abused)?
I have a budgetone that really picks it up and a generic IN1800 (or
something like that) that doesn't pick it up much at all.
And when using soft phones, it depends on
I guess it could/would depend on the quality of the codec your using,
which ones are you using? (*not* for phone secks!)
-Original Message-
From: Mark Benson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 22, 2005 6:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Yes
Application Background()
On Tue, 2005-02-22 at 14:35, Muhammad Muzzamil Luqman wrote:
Whenever some call comes in i want it to be automatically picked up
and then it plays some message Welcome to xyz, Press 1 for sales and
2 for support and then it takes it to the particular extension of
while using iax and a soft phone, the sound of breathing comes through
so clearly that it has started bothering me. Earlier I was amazed at
the quality, but now feel it is irritating. Wondering if there is a
way to cut it down. I am in the process of exploring using iax for a
call center,
Hi,
I have two asterisk machines, chomper and otao.
otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no
PSTN connections.
chomper is at my house, connected to cable modem behind NAT, but has a single
X100P PSTN connection.
I would like to establish two way calling between
read the default extensions.conf file. some
examples included
- Original Message -
From:
Muhammad Muzzamil
Luqman
To: asterisk-users@lists.digium.com
Sent: Tuesday, February 22, 2005 2:35
PM
Subject: [Asterisk-Users] does asterisk
support menus?
Sure you can. Look it up on the Wiki.
You may also want to look up AGI scripts, if you're looking to apply
more intelligence to your voice response system.
Mohit.
On Tue, 22 Feb 2005 17:35:59 +0500, Muhammad Muzzamil Luqman
[EMAIL PROTECTED] wrote:
Whenever some call comes in i want it to
HI all
What is the best to send a fax with a PRO.
I got it working on the receiving and e-mailing it.How do I send one
Thanks
Altus
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You sure can using Background cmd for saying stuff and then
program a dialplan using command and extensions.. quite easy
really.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Muhammad
Muzzamil LuqmanSent: Martes, 22 de Febrero de 2005 06:36
a.m.To:
All clients tryingtoconnect tothe asterix says that there is a problem
comunicating with the server. It looks like the clients don't see the
server. But from the log it looks ok. Maybe may config is wrong.
What is the simplest configuration to allow external clinets tocontect to
my server. For me
Try adding nat=yes to each extention in the sip.conf file.
Example
[3420515]
username=3420515
type=friend
secret=1234
qualify=no
port=5060
pickupgroup=
nat=yes
mailbox=
host=dynamic
dtmfmode=rfc2833
disallow=
context=from-internal
canreinvite=no
callgroup=
callerid=TEST
A little off-topic maybe, but it's still for the Adit used with Asterisk. ;)
I wonder where I can buy 50 pin Amphenol cables, with connector on one side,
and open cables on the other for mounting in our own patch panels.
In Europe, or Sweden preferably.
It's said to be very common on telcos, but
What is the simplest configuration to allow external clinets tocontect to
my server. For me it was this entry in iax.conf
[client1]
type=peer
usernamename=client1
secret=test
context=sip
host=dynamic
allow=all
Just a side note : you can't connect mutiple IAX clients
simultaneously with
On Tue, 22 Feb 2005, Hariharan Gopalan wrote:
while using iax and a soft phone, the sound of breathing comes through
so clearly that it has started bothering me. Earlier I was amazed at
the quality, but now feel it is irritating. Wondering if there is a
way to cut it down. I am in the
FOP
http://www.asternic.org/
On Tue, 22 Feb 2005 22:40:32 +1100, PHP Mechanic
[EMAIL PROTECTED] wrote:
I wish to initate calls from a web interface, by clicking on a link and then
connecting to the automatic outgoing call by picking up an analogue phone.
I've got one fxs and one fxo and I
Digium had some internet outage on Feb 21st (someone on IRC said
this). Don't know if it is fixed yet.
On Tue, 22 Feb 2005 03:02:39 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Mon, 2005-02-21 at 14:56 -0500, Jared Watkins wrote:
I've been unable to resolve *.asterisk.org and
Yes, exactly (and there will be other settings as well, to identify
the
type of peer (network, trunk, endpoint) for other reasons).
cool, I really should read the lists more :)
That's coming too, but in a different way. Actually if your remote
peer
can send you Remote-Party-ID
On Tue, 22 Feb 2005, Daniel Nyström wrote:
A little off-topic maybe, but it's still for the Adit used with Asterisk. ;)
I wonder where I can buy 50 pin Amphenol cables, with connector on one side,
and open cables on the other for mounting in our own patch panels.
In Europe, or Sweden
Did you confirm you are not running graphics? (X, frame buffer, etc).
Did you confirm you have unmasked IDE interrupts (-u1 to haparm)?
Yes and Yes. ;-)
# hdparm -i /dev/hd[ac]
/dev/hda:
Model=SAMSUNG SP1203N, FwRev=TL100-24, SerialNo=0836J1FX419316
Config={ HardSect NotMFM HdSw15uSec Fixed
Alex G Robertson wrote:
Did you confirm you are not running graphics? (X, frame buffer, etc).
Did you confirm you have unmasked IDE interrupts (-u1 to haparm)?
Yes and Yes. ;-)
# hdparm -i /dev/hd[ac]
/dev/hda:
Model=SAMSUNG SP1203N, FwRev=TL100-24, SerialNo=0836J1FX419316
Config={ HardSect
Try hdparm -v /dev/hda
Here it is
# hdparm -v /dev/hd[ac]
/dev/hda:
multcount= 16 (on)
IO_support = 1 (32-bit)
unmaskirq= 1 (on)
using_dma= 1 (on)
keepsettings = 0 (off)
readonly = 0 (off)
readahead= 8 (on)
geometry = 14596/255/63, sectors =
Strike 1, you sent HTML email.
Strike 2, you obviously didn't google.
strike 3, well not yet..
http://www.google.com/search?q=Illegal+instruction+site%3Alist
s.digium.com
--
Steven Critchfield [EMAIL PROTECTED]
1) Muah, due to a recent (yesterday) forced upgrade to Outlook
Peter Svensson wrote:
On Tue, 22 Feb 2005, Daniel Nyström wrote:
A little off-topic maybe, but it's still for the Adit used with Asterisk. ;)
I wonder where I can buy 50 pin Amphenol cables, with connector on one side,
and open cables on the other for mounting in our own patch panels.
In Europe,
Dear list,
I have been using asterisk for some time now. However I have never
used it with any of the digium or compatable cards (Purely used for
SIP).
I understand that for using Meetme, I need to have a timing device,
which could either be hardware or zrdummy etc (I am not using any
right
Hi,
Can someone tell me if the timing device is needed for voicemail and
other applications too?.
i'm sure that searching on google and/or voip-info.org can lead
to an answer.
btw, the answer is no. only meetme and iax truking needs a timing
device.
--
Matteo Brancaleoni
System
Hi
Folks,
I've been
experiencing something very strange...
When I want to
listen to call between a SIP phone and a Zap Channel, I can listen a with a nice
audio quality.
When it comes to
record using the monitor command, I just have a wav file which is completely
noisy and I can't
Hi. I am trying to setup a pre-paid sip billing, but I can only find calling
card application.
Is there somewhere I can find one ?
Fredrik
_
MSN Messenger http://www.msn.no/computing/messenger Den raskeste veien
mellom deg og dine
On Feb 21, 2005, at 7:35 PM, Rudolf Ladyzhenskii wrote:
Hi, all
I am doing prrof of concept system. I will have two IP phones
connected to Asterisk box. Box itself will have 1 PSTN conenction and
one analog phone conenction. A basic minimal configuration.
At the moment I am planning to use an
On Tue, 22 Feb 2005, Michael Welter wrote:
Are you aware of the type 66 punch-down block with an AMP-50 connector?
Also the harmonica--an AMP-50 on one side and 12 RJ11 jacks on the
other (two pair/jack).
We punched the cable directly to the jacks since that was what we needed.
None of
Try this:
In the extensions_custom.conf file set:
[custom-myapp]
exten = s,1,SayDigits(1234)
exten = s,2,Hangup()
You're telling it to go to priority s,
(exten = 3,1,Goto(custom-myapp,s,1)
but in the custom-myapp context, you have priority 3
instead.
Hope that helps.
Maya
--- Chris
On Tue, 2005-02-22 at 10:48, [EMAIL PROTECTED] wrote:
When adding the details in AMP for when caller dials 3, I have
referenced it using 'custom-myapp,s,1', and if I go to
'extensions_additional.conf' I see the following line under the rest of
menu item info that was created :
exten =
Anybody know a good IAX provider for Canadian DIDs?
I currently use Xetricom for Toronto DIDs (C$7.50 each). I also know of
someone who can provide a Toronto DID with unlimited* GTA calling for
C$20.
Nabeel
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I'll let someone else speak to the missing .conf
files.
If you could post your extensions.conf and
extensions_additional.conf, it would be easier to help
you debug this. The suggestion from timebandit
_should_ have worked given your original post.
Personally, for learning my way around
Have you used them before?
Do they provide commercial grade service?
On Tue, 22 Feb 2005 10:08:57 -0500, Nabeel Jafferali
[EMAIL PROTECTED] wrote:
Anybody know a good IAX provider for Canadian DIDs?
I currently use Xetricom for Toronto DIDs (C$7.50 each). I also know of
someone who can
Just wanted to let you know you were just left a 2236:45 long
message (number 1) in mailbox 1000 from an unknown caller, on Monday,
February 21, 2005 at 05:38:14 AM so you might want to check it when you
get a chance. Thanks!
--Asterisk
Hmmm... Call came in
Hello All,
I am wondering if there is a way to make Polycom IP 500 SIP phone display
digits dialed after a call is connected. For example, when I call an IVR,
after the connection is complete, the digits dialed to enter account number
etc are not displayed on the phone. They are sent to the
Title: QSIG, Asterisk and Eicon DIVA
I am trying to get Asterisk to work with my old PBX, a Siemens Hicom 150E. In my Asterisk server I have a Eicon Diva BRI card(that supports QSIG) it is then connected to my PBX via S0/QSIG.
How do I configure CAPI to use QSIG? Is QSIG supported by
Title: Noob question on connection
Hello,
I just started with asterisk and I start to get it, but there is one thing that I don't seem to get:
If I put an FXS-card into my asterisk server, then I can phone to the server with a normal phone, but can that phone also be reached by de
Paul,
I saw this problem when my Sipura SPA3000 was not detecting the PSTN
line's the CPC Signal properly. I bumped the min CPC duration setting
(under PSTN line tab) from .2 to .5 and the problem went away. I have
never had this problem with X100p hardware.
-Matt
On Tue, 2005-02-22 at
Andy,
There's no such thing as a noob question. Everyone has to start somewhere.
FXS can only handle one call at once. You can get a second FXS phone,
use a protocol like BRI that can handle more than one call at once, or
use call waiting.
Two Asterisk servers can be connected over ethernet
I'm looking for a simple way to disable MusicOnHold in my environment.
I'm not really interested in having it and it causes too many problems
with hanging mpg123 processes and memory management errors. The problem
is, so many other modules seem to depend on it. I can't just cause a
noload
quote who=Nathan Alberti
I have the following defined in sip.conf
[ccme-in]
type=peer
host=10.0.9.1
context=devel_in
disallow=all
allow=alaw
nat=no
canreinvite=yes
qualify=yes
and [devel_in] is defined in extentions.conf
However when I try to call via the dial peer I have
My two proposals are:
1. You can contact webvoip.com they are billing guys who do what you
need.
3. You can wait for IBS to become integrated into Asterisk.
Your question is a little vague as to what you need.
The more chatty among us readers will surprise you with help if you can
give more
Hi all,
I'm playing with Asterisk and I've already configured all needed .conf
files.
It works quite well, but now I need your help to tune the system: when I
place a call from a softphone to the PSTN, I can't hear directly Telco's
tones and I can't use its services, e.g. a mobile's answering
you may not be aware of the asterisk-biz mailing list, which is probably
more appropriate for a discussion like this.
you'll find many VoIP termination vendors hang out there too.
Regards,
Scott Stingel
Mohit Muthanna wrote:
Have you used them before?
Do they provide commercial grade service?
Hi all
i am having odd problems.
nothing worng with the server starting up or anyhting like that.
asterisk server = P4 2.4 512M Ram 80gig/hd 100M/Lan Digium 4 port FXO Card
i am using X-Lite on macines on the same network all running on 100M/Lan
calling etc all works fine calling to PSTN works
Can you not just remove the sym link to the mpg123 process so asterisk
doesn't find it therefore no music on hold?
When I was trying to get music on-hold working I had to compile and sym
link the mp123 executable - when it wasn't present I had no music on hold...
Mark
MF Hulber wrote:
I'm
Here is the setup:
SIP/3044 - SetCallerID(5551212) - Call out PRI
The CDR shows a src of 5551212. That is a lie! The src of that call was not
5551212, the source was 3044! The translated source of that call was
5551212.
How can I get real source of this call and not some faky nonsense?
The
i have got a music file with extension mp3 and it
is not workign with background()
is there any way to convert the mp3 to gsm or any
other codec?
Kindest
Muhammad Muzzamil Luqman
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I use the following recipe for this in Linux...
mp3-decoder -w outfile.wav infile.mp3
normalize outfile.wav
sox outfile.wav -r 8000 outfile.gsm
Things sound pretty good like that. You can do it with sox at one shot
but I like the normalization so all of my recordings sound approximately
the
On Tue, 2005-02-22 at 21:55 +0500, Muhammad Muzzamil Luqman wrote:
i have got a music file with extension mp3 and it is not workign with
background()
is there any way to convert the mp3 to gsm or any other codec?
Yeah, we could send you to remedial linux classes where you would learn
how to
MF Hulber wrote:
I'm looking for a simple way to disable MusicOnHold in my environment.
I'm not really interested in having it and it causes too many problems
with hanging mpg123 processes and memory management errors. The problem
is, so many other modules seem to depend on it. I can't just
A standard scsi cable works great.
Just cut it in half.
Cheers,
Jon.
On Tuesday 22 February 2005 07:17 am, Daniel Nyström wrote:
A little off-topic maybe, but it's still for the Adit used with Asterisk.
;)
I wonder where I can buy 50 pin Amphenol cables, with connector on one
side, and
Take a look at this URL:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files
Ive used the following command
sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql
hope this helps
Grüsse / Best Regards
Mateo Meier
I have InAccess Networks' oh323 installed and partially working. I can call
the h.323 phone from asterisk using Dial(oh323/${IP_ADDRESS}). How do I
dial from the phone to an asterisk extension? It does not appear to me that
the phone actually registers (or attempts to register) with asterisk.
Hello Everyone,
I am looking into using Asterisk as our company PBX and voicemail system. I
am very familiar with Linux, but the VOIP stuff is new for me.
We are a non-proffit organization, so keeping things as cheap as possible is
very important. I am looking on some advice for best
Muhammad Muzzamil Luqman wrote:
i have got a music file with extension mp3 and it is not workign with
background()
First off, why are you trying to background an MP3? If you are
backgrounding music, then you should probably be using Music on Hold.
If you are using mp3's for audio prompts in
Hello,
I've just uploaded a patch to amportal project at sourceforge, to
support Zap Extensions...
http://sourceforge.net/tracker/index.php?func=detailaid=1146433group_id=121515atid=690574
I'd appreciate some feedback ;)
Greetings
Julian J. M.
On Sun, 20 Feb 2005 22:00:44 -0500, Robert
I will certainly look at this. However, it may not be till late this
weekend as I have a five page paper due Saturday and I have a lot of
work left on it.
This was the only thing that was keeping me from using the [EMAIL PROTECTED]
iso.
Look forward to being able to test this.
Robert
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