I suggest you go the channel bank route.
On Wed, 18 Aug 2004 10:16:01 +0200
Miroslav Nachev [EMAIL PROTECTED] wrote:
Hi,
We have a case where we need of 16 x FXS, 12 x FXO and
1 x E1. To
do this using Digium products I need of 8 PCI slots. This
is not
possible to be done in one
Hi
I know of a product called a Parlay which does this, but
its expensive. Someone on the list said that asterisk could
do this with a quad T1 card.
I think that would be very nifty if asterisk could transfer
the isdn calls based on CLID or DNIS before the call is
actually answered.
If you get
On Wed, 4 Aug 2004 13:34:02 +0200 (CEST)
Peter Svensson [EMAIL PROTECTED] wrote:
On Tue, 3 Aug 2004 [EMAIL PROTECTED] wrote:
There is a device called a parlay made by a crowd
called
voxtream which will route the ISDN calls based on the
DID
and/or the callerid, before the call is
Hi
From what I have heard, Asterisk does not allow for iLBC to
take advantage of the lost packet concealment.
I understand this has something to do with the jitter
processing.
If lost packet concealment doesnt work with ilbc, I can
assume the same applies to other codecs who claim to have
this
On Tue, 3 Aug 2004 05:47:59 -0400
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 03 August 2004 02:58, [EMAIL PROTECTED]
wrote:
From what I have heard, Asterisk does not allow for
iLBC to
take advantage of the lost packet concealment.
I understand this has something to do with the
Hi
There is a device called a parlay made by a crowd called
voxtream which will route the ISDN calls based on the DID
and/or the callerid, before the call is answered.
It would be nice if this feature could be done in Asterisk
as well, but at this point in time, it first answers the
call.
Rodopi is a radius system.
Just build your own using freeradius.
Are you using Cisco that you need radius?
Cheers
Clive
On Fri, 30 Jul 2004 11:44:14 -0700
Darren Bentley [EMAIL PROTECTED] wrote:
Hello,
Has anyone used Asterisk in conjunction with a billing
system like
Rodopi? Is the
Hi
I am trying to read a number back using the command
SayNumber. This worked fine in older versions of
asterisk, but now I am trying CVS head and I get this
error:
Jul 27 16:10:53 WARNING[507921]: file.c:1004
ast_waitstream_full: Wait failed (Interrupted system call)
The line in code is:
Hi
Out of interest, (this may be not possible) but I think it
would be an excellent idea to modify firmware to handle the
IAX2 protocol. Especially since its a linux based phone.
Thoughts?
Regards
Clive
On Mon, 19 Jul 2004 21:54:59 +
Joshua Colp [EMAIL PROTECTED] wrote:
Hello
Hi
Just create a new context, and use ex girlfreind logic.
cheers
Clive
On Wed, 21 Jul 2004 14:58:17 +0200
GIBERT Frédéric [EMAIL PROTECTED] wrote:
Hello,
Can someone explain me how to do caller based routing.
Here is my example.
I have an asterisk between a PBX and the PSTN. The
hi
I had the saem trouble, so I just took my x100p card out
and the problem went away:)
I know its not the ultimate solution, but I decided to use
an ATA with my analgue phone instead.
I would suggest trying to put the analogue lines as channel
7 and the isdn lines as channels 1-6
Good luck
Hi
I got the billion card for hfc-s to work, but not with the
rh9 kernel, I downloaded a new kernel 2.4.26
The trick then is to make sure you have the symbolic links
correct, then it compiles and works like a dream!
hope this helps.
Regards
Clive
On Wed, 7 Jul 2004 13:07:16 +0200
Thomas
Sorry, I forgot to mention, you need to use bristuff 0.0.2
thats the zaphfc driver
cheers
Clive
n Wed, 7 Jul 2004 15:41:49 +0500
Junaid Saeed Uppal [EMAIL PROTECTED] wrote:
Hello There,
I am trying to get Asterisk to work with Billion ISDN
Adaptor, But i
couldnt get isdn4linux to work. I
Hi
does anyone know if its possible to run Bristuff together
with a tdm card in the same computer.
I get an error when trying to start asterisk in chan_zap.c
My zaptel.conf looks like this:
loadzone=us
defaultzone=us
fxsks=1
fxoks=2
fxoks=3
span=1,1,3,ccs,ami
bchan=4-5
dchan=6
Thanks
Clive
The most economical way is just multiple asterisk boxes,
even though it may use more space.
On Thu, 3 Jun 2004 20:13:03 -0600
brian k. west [EMAIL PROTECTED] wrote:
Go spec some hardware dsp chips and boards that can do
100 channels... I
think you will fall out of your chair.
bkw
-
I have done a quick search and there are some nice looking
dsp-pci cards out there. (Dunno abt prices). It may take
some coding to get them working with Asterisk , and one
would not require a super-power quad xeon processor if it
had a huge dsp card.
May be an interesting way to scale asterisk
Hi all
I am attempting to install bristuff, and have not had much
success.
I have my kernel sources installed (RH9), and am following
the instructions step by step.
Things seems to fall off the rails when I try make make
clean all in the zaphfc directory, which is part of the
install.sh script.
My advice is just sell them.
no-one I know is bothered with Icasa approval, as long as
it works, its fine.
That card has FCC approval, as far as I know.
ALles van die beste!
Regards
Clive
On Fri, 30 Apr 2004 15:17:13 +0100
WipeOut [EMAIL PROTECTED] wrote:
Altus Snyman wrote:
Good day
Hi
I am also having jitter trouble on IAX2, and I can vouch
that the jitter buffer is busted.
On Wed, 07 Apr 2004 09:56:01 -0400
Steve Kann [EMAIL PROTECTED] wrote:
Andrew Kohlsmith wrote:
Are there open problems/issues with iax2 and jitter
(quality)?
Just upgraded to
Jason, hi
Don't waste your time on old technology.
This was done on the old komodo phone , sold by net2phone
as a yap jack and there are some ipphones (VIC phone) comes
to mind with an analogue modem built in.
We even have adsl here in parts of Africa, (not that its
got any bandwidth
Hi
I havent been able to get the jitter buffer to work even
with correct typing.
If you have any luck, please let me know how it performs
for you.
Thanks and Regards
Clive
On Thu, 12 Feb 2004 19:56:27 + (GMT)
Michael T Farnworth [EMAIL PROTECTED] wrote:
I had noticed that the
Steve hi
Yup, adsl, seems to be getting slower by the day.
Maybe we can configure * to change the iax to port 21 udp ?
Regards
Clive
On Thu, 5 Feb 2004 13:21:08 +0200 (SAST)
Stephen Davies [EMAIL PROTECTED] wrote:
On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote:
Hi
I wonder if
Basically voip is only legal if used between branch offices
of a company that are connected using leased lines.
Archaic.. yes, stupid... yes, but thats the law here..:(
Our telco is strangling the country so they can line their
pockets.
On Thu, 05 Feb 2004 11:57:57 +
Chris Lee [EMAIL
Steve, I still would love to know how to improve the jitter
settings:)
I still have managed a conversation, but its not great at
all with the sound breaking up.
Some sort of jitter control will definitly help.
Thanks
Clive
On Thu, 05 Feb 2004 14:19:07 +0200
[EMAIL PROTECTED] wrote:
Steve hi
Hi
I wonder if anyone has a fix or any advice for the IAX2
jitter buffer.
My internet connection here in South Africa has an
international ping time of 550ms +- 50 ms. According to the
scientific approach I would like to add a 100ms jitter
buffer. (nevermind the latency)!
I have tried playing
I wonder which voice codec they use, they say one can use a
28k modem using their service which rules out ilbc.
On Mon, 1 Dec 2003 17:34:41 -0500
Chris HARIGA [EMAIL PROTECTED] wrote:
Hi,
VoiceGlo is comercial version of Asterisk? :)))
loo
Take a loock on
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