Hello all,
I was wondering if the DTMF were generated from the phone or from the
ATA? I have a cisco ATA 186.
Thanks
K.
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Hello,
Is there a particular order in which codec should be entered in the
oh323.conf file?
I believe that they are put in order of priority. But depending on which
codec is put before another, even if the caller does not support all of
them.
Let me clarify. I have a cisco ATA. When I have this
Hello,
I am using oh323 and i think there is a bug. When i enter any digits,
There is a white space following the digits.
E.G. when i enter 333 oh323 responds 3 3 3
Because of that the DTMF does not get recognized.
Has anybody encountered and solved this problem?
Any hints will be greatly
Hello,
I am using oh323 and i think there is a bug. When i enter any digits,
There is a white space following the digits.
E.G. when i enter 333 oh323 responds 3 3 3
Because of that the DTMF does not get recognized.
Has anybody encountered and solved this problem?
Any hints will be greatly
Hello,
I am setting up a prepaid card system.I need to be able to send from
some numbers DTMF inband, and from some other number DTMF out of band.
In my h323.conf file, when i set DTMF modeto rfc2833 or inband
separately it works. I don't know how to set up both mode, so i was
thinking about
what os are you running?
K.
- Original Message -
From: Rodolfo Grave [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion [EMAIL PROTECTED]
Sent: Monday, December 13, 2004 1:27 AM
Subject: Re: [Asterisk-Users] Re: Cant set H323 up
Hi
Now I
hello are you sure that you have loaded the module in the modules.conf
files?
load = chan_oh323.so
K.
- Original Message -
From: Rodolfo Grave [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, December 11, 2004 3:32 PM
Hello,
Is there an utility for asterisk for codec
conversion? I tried google but i haven' got anything.
I am trying to initiate a call with G711 codec to
asterisk and i would like asterisk to call a gateway with an g729 codec,
therefore making a codec conversion from g711 to g729. I know
hello all,
can both chan_h323 and asterisk oh323 be installed on the same machine?
tx
K.
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If you are using GnuGK, i think this should do,
in your h323.conf file, configure an asterisk endpoint as follow for
instance
[time] Username
type=h323
e164=99
context=test
K.
- Original Message -
From: Nahuel Alejandro Ramos [EMAIL PROTECTED]
Can't compile either
I have the oh323 v 0.6.4
I downloaded both openh323 Janus v4 and pwlib Janus v4 on inAccessNet
website
untared it, applied the patch. and compile both pwlib and open h323 (this
version compiles much much faster than the other version that seems VERY
weird to me
I believe i
i have had some problems with the H323 channel ...
Other party not anwsering SIP 2 H323 bridge.
the chan_oh323 solves the problem. Use
it.
(Even though it is quite complicated to install but
READ the README file)
Nahuel that should solve it!!
Kido
Hello all,
I found a LSI161SCREV2 Dialogic board in one of my
drawers, and i was wondering if by any luck, i could make some magic happen with
asterisk ... If asterisk does not support it, is there any PSTNto H323 or
PSTN to SIP gateway that support this dialogic card and that can be
Hi Nahuel,
in you h323.conf file, add the following line
gatekeeper = yougk.ipadress.here
then create an asterisk endpoints in your gk like this
[detgw]
type=h323
e164=100
context=context
Then if you h323 endpoint is registered and if you modify you
extensions.conf file like this it should work
hi all,
i need an asterisk board that can support up to 30
FXS ports. i found the following on digium: TDM400P. It only has 4 port, does it
mean thati have to buy 8 of those to have my 30 FXS line? Or is there any
channel bank solution that digium or other providers can provide us with in
I got some answer after googling but any advice
will be greatly appreciated..
Just in case someone else need some info. http://www.voip-info.org/tiki-print.php?page=Asterisk+hardware
- Original Message -
From:
kido noagbodji
To: Asterisk Users Mailing List -
Non
Hi,
But i just can't seem to
make it work using oh323/coded g729? Its like it does
not respond to DTMF signals? I have dig into many
mailing list and not any clear solutions. Could
What DTMF mode are you using? Do you have a g729 license installed on your
system? Remember that g729 only works
Hi all,
Is there any commercial g723 license for asterisk?
Where can it be purchased? Has somebody used it?
Thanks
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Hi Paul,
Are you using the h323 or the oh323 channel. Please, what is the status of
the bug that you are talking about?
Thanks
- Original Message -
From: Paul Davidson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 22, 2004 3:11 PM
Subject: Re: [Asterisk-Users] Asterisk
Peter,
If you have the lastest CVS version of asterisk(1.0.11) , and the latest
version of asterisk-oh323(0.7.0), it won't work.
What version of asterisk are you running? what version of oh323 are you
trying to compile?
K.
Peter,
Peter Landy wrote:
New to Asterisk so I am sure this has
Hi Jorge,
The oh323 channel and h323 channel by NuFone are different.
As far as your problem, this looks like a codec problem i had. Try to look
that way.
K.
- Original Message -
From: Jorge Alayon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL
Hello,
I have a sattelite link (expensive link) and we are
very concern with our bandwidth usage. I would like to force asterisk to place
outgoing call(over the sattelite)to g729 codec no matter what codec comes
to asterisk.So asterisk will be making translation from the incoming codec
Hi all,
I wrote an AGI script in perli asked the
script to dial a number
$AGI-set_callerid($calleridnum);
if($left
0) {
$AGI-exec('Dial',"H323/$phonenumber");
}else{ $res =
mystreamfile("vm-goodbye");
$AGI-hangup(); }
The chan_h323 registers well to my gnugk. I call
Hello,
I just purchased 10 G729 licenses for my asterisk box from Digium I was able
to register the key. But when i start asterisk it fails with the error
message:
[codec_g729a.so]Nov 18 09:27:01 WARNING[135073792]: loader.c:248
ast_load_resource: Shared object libc.so.6 not found
Hi Hammoud,
It all depends on the codec that you are using.
Best case scenario is with G723 codec 6.3Kbps per channel * 20, around 126K
without the overhead. But you problably won't be able to use this codec unless
you are in passthru mode (license is pretty expensive).
Using g729 you will
Thanks for the input ...
Kido
- Original Message -
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, November 18, 2004 3:47 PM
Subject: RE: [Asterisk-Users] internet bandwidth
/SNIP/
Some
Hi,
Apparently, gcc is not installed in your system. Install it and continue
It is a C compiler ..
Checking for gcc... gcc
checking whether the C compiler (gcc ) works... no
configure: error: installation or configuration problem: C compiler cannot
create executables.
make: ***
Visit http://gcc.gnu.org/
- Original Message -
From: Iqbal Gandham [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, November 18, 2004 6:46 PM
Subject: Re: [Asterisk-Users] Can some bady help me ???
ur c compiler, do u have
Hello,
I just downloaded and installed the latest version
of asterisk under Fedora. (had it under FreeBSD but was having TOOO many
problems)
After my installation i noticed that the channel
H323 was not included ( I remember that i did not have to install it under
freeBSD) but I have seen
: [EMAIL PROTECTED]
Sent: Friday, November 19, 2004 2:02 AM
Subject: Re: [Asterisk-Users] Is H323 dying?
On Fri, 2004-11-19 at 01:48, kido noagbodji wrote:
Hello,
I just downloaded and installed the latest version of asterisk under
Fedora. (had it under FreeBSD but was having TOOO many
Hello,
I have seen a lots of references to IAX in the list
but i am not quite sure what it is. What does it mean?How does it work? Why
IAX?
I just installed asterisk and so far i am using it
for IVR and routing some H.323 call through a provider. How can i benefits from
IAX?
Thanks
Kido
the g729 codec, is there a way i can "test" it?
Many Thanks
Kido
- Original Message -
From:
kido noagbodji
To: [EMAIL PROTECTED]
Sent: Sunday, November 14, 2004 3:36
AM
Subject: [Asterisk-Users] Cisco ATA and
G729
Hi all,
I am new to a
to you and disallow/allow codecs in your configuration
until
you can find the source of your problem.
On Monday 15 November 2004 03:23 pm, kido noagbodji wrote:
Hello all,
* However, when i set my Cisco ATA to G711, i can't hear any sound
unless
I press at least two or three keys(any
Thanks
Kido
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, November 15, 2004 5:58 PM
Subject: Re: [Asterisk-Users] Cisco ATA and G729
Dinesh Nair wrote:
On 16/11/2004 00:08 kido
Hi all,
I am new to asterisk. I was able, but not without
pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs
softphone to work with the PBX.
Three remarks:
* On the SJphone, i use the GSM and the G711 (ulaw
and alaw) codec. In the h323.conf file i enabled those
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