Hi all,
I found some message in the Digium list archives that discussed ISDN
BRITE support. There was also some discussion on 4:1 bitrate
conversion. I did some searching on the source code and didn't see any
reference to BRITE or bitrate conversion.
Does the current code
On Fri, 2006-04-14 at 08:19 -0500, Rich Adamson wrote:
I believe the TDM2400 has the capability of doing on-card fxo-fxs data
flows (without hitting the pci bus), but that function has not yet been
implemented. Its basically required to support faxes in an analog
environment. When it is
On Fri, 2006-04-14 at 13:49 -0300, Joshua Colp wrote:
Some people have problems, some people don't. There is no way you can be
prepared for every situation out there. We try our best.
I was looking at using a Dell server for running Asterisk and noticed
that Dell has started using
on their server class machines.
...Jeff
Aaron
On Fri, 14 Apr 2006, Jeff Gustafson wrote:
On Fri, 2006-04-14 at 13:49 -0300, Joshua Colp wrote:
Some people have problems, some people don't. There is no way you can be
prepared for every situation out
On Fri, 2006-04-14 at 15:10 -0500, Kevin P. Fleming wrote:
Jeff Gustafson wrote:
I was looking at using a Dell server for running Asterisk and noticed
that Dell has started using PCI-X on a lot of their new systems. Does
this newer bus standard help the situation with faxing
On Fri, 2006-04-14 at 15:35 -0500, Kevin P. Fleming wrote:
Jeff Gustafson wrote:
My fault. I meant to say PCI-e, which is a newer bus that Dell is
shipping on their server class machines.
Right. That is not supported by any Digium products yet, but it still
won't help the FAXing
On Mon, 2004-05-24 at 19:33, Paul Mahler wrote:
I have had good experiences with Adit. Their customer service and
documentation are excellent.
Sounds good. So they have chassis that can handle = 100 analog
phones? I looked at something called the Adit 600. But I wasn't sure I
could
On Tue, 2004-05-25 at 14:06, Steven Critchfield wrote:
Most devices you will be using will be in some multiple of 24 as that is
the number of channels in a T1. An Adit 600 will allow you up to 48
channels as it is capable of handling 2 T1s on the back and 6 x 8port
cards. So your 100 phones is
: 2000.103
cache size : 256 KB
Also, is yours the true Digium card or is yours a
One real, 3 clones. Customers get real cards, my poor testing lab gets
clones.
...Jeff
Thanks!
Jeremy
-Original Message-
From: Jeff Gustafson
Caller*ID used to work as some point, but I can't seem to get it going
these days. The card is a x101p. I've tried going up and down the
rxgain scale. Can the txgain effect it at all? When I plug in a phone
into the line with a splitter it can decode caller id with no problems.
. If
that turns out to be the case, I may be forced to go ahead and get an
actual Digium card sooner than I anticipated in order to prove the
theory.
Regards,
Jeremy
-Original Message-
From: Jeff Gustafson [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 08, 2004 12:06 AM
To: [EMAIL
I just wanted to ask about using Adtran boxes to support analog lines
into an Asterisk box. Right now x101p's are just too sensitive to RF
noise inside the PC. Going with an external chassis looks like a good,
albeit expensive option. It looks like I can use the Digium T1 card
into an
On Mon, 2004-04-05 at 17:29, Jonathan Biggs wrote:
Had a similar problem
What kind of phone are you using on that line?
My case, the line was sharing a Palm modem on it.
Electrical noise from the power strip was causing the
problem.
I'm having a similar problem. The difference
Voicetronix is actively working to improve the vpb driver in Asterisk.
Right now the code they sent me is for the Asterisk CVS code. Right now
the Asterisk CVS code seems to be totally hosed for incoming audio over
an analog line even with a x101p. I'm hoping that things will clean up
Hi all,
I'm having a terrible time with a voicetronix Openswitch/8 board. I
don't mind having to patch up the driver a bit, but I don't understand
why there is such a *drastic* echo. For example, if I call my cell
phone from a Cisco SIP phone, I heard my self on the cell phone and on
the
I'm trying to setup an 8 line PBX with asterisk. I'm not having good
luck with the Voicetronix Openswitch board. I was wondering if I went
with a channel bank like an Adtran 750 if I would have echo problems.
This is the main problem that my boss complains about. I've tested the
x101p,
Sorry to put out 3 messages in one day with similar questions, but this
echo stuff is driving me crazy. The Openswitch card gave me a 3 second
echo today. That's really long! It seems it's only that bad when I
dial into Asterisk to an extension versus dialing out. I've adjusted
gain
On Mon, 2004-03-01 at 22:09, Andreas Anderson wrote:
Hi,
does anyone has an LDAP based directory for the Cisco 7900 Series? I found
some directorys based on
sql, but an ldap directory would allow syncronisation with evolution,
mozilla, multisync (Palm, OPIE, Mobile phones etc...)
Greez
I have a x101p and I can't seem to get ztmonitor to work on it. I've
tried it on 2 different machines. One with a SBLive! card and the other
with a AMD-768 [Opus] Audio (rev 03) chip. Neither machine give me a
graph in ztmonitor 1 -v mode. If I run ztmonitor without the -v I
get:
Ah! I just checked out the latest ztmonitor out of cvs and it works
just fine.
...Jeff
On Mon, 2004-02-23 at 12:51, Jeff Gustafson wrote:
I have a x101p and I can't seem to get ztmonitor to work on it. I've
tried it on 2 different machines. One
It turned out that a Gb/E card was too close to the modem causing
activity on the card to bleed over to the modem.
...Jeff
On Thu, 2004-02-12 at 16:11, Jeff Gustafson wrote:
I'm experiencing periodic beeps or screeching when I'm on a call via
I'm experiencing periodic beeps or screeching when I'm on a call via
the x101p card to/from PSTN. Echo cancellation seems to be working
fine. The beeps seem to happen with echo cancellation on or off. Is
there a setting I can tweak for this?
The problem does not occur if I'm
I tried out today's release of Asterisk. I am now able to consistently
decode inband DTMF over SIP. The problem is that Asterisk only decodes
the first code. So extensions consisting over more than one digit do
not work. So, for example, in the demo extensions, extension 2 and 3
work
=alaw
Still, only the first digit is decoded. Anything past the first code
are not decoded.
...Jeff
On Wed, 2004-02-04 at 17:38, Jeff Gustafson wrote:
I tried out today's release of Asterisk. I am now able to consistently
decode inband DTMF over SIP
On Fri, 2004-01-23 at 00:33, Martin Bene wrote:
Hi Siggi/Jan,
The Error Verifying Config Info Message doesn't have anything to do with
the real problem. I also get that message, possibly because I don't keep a
device specific config file (SEP000D65707B78.cnf.xml) or
DISTINCTIVERINGLIST.XML on
Hello all,
I've been trying to get a simple PBX up and running with
asterisk. I decided to sign up with Voiceglo so I could have a PSTN
gateway. The problem is that I can't seem to get Asterisk to handle
dtmf decoding reliably. I tried inband and the rfc decoding. inband
tried to work
On Wed, 2004-01-21 at 06:04, Jan Czmok wrote:
Jeff Gustafson ([EMAIL PROTECTED]) wrote:
Hi again,
I found chan_skinny and that seems to work pretty good. the SCCP one
filled out all the buttons really nice, but skinny seems to be
working.
How do I fill out the second line
On Wed, 2004-01-21 at 06:04, Jan Czmok wrote:
Define a second section, however, you also might want to take a peek at
chan_sccp. We are currently reworking the complete chan_sccp to support
all functions known by the Skinny Protocol. Yes, EVEN the 7920 is
working with it now :-)
://10.0.0.112:8086/ciscodirectory?accent=true/directo
ryURL
idleURL/idleURL
informationURL/informationURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURL/servicesURL
/device
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
On Wed, 2004-01-21 at 13:17, Jan Czmok wrote:
Jeff Gustafson ([EMAIL PROTECTED]) wrote:
On Wed, 2004-01-21 at 06:04, Jan Czmok wrote:
Define a second section, however, you also might want to take a peek at
chan_sccp. We are currently reworking the complete chan_sccp to support
all
On Wed, 2004-01-21 at 14:05, Jan Czmok wrote:
Kewl, I was apparently trying to use older chan_sccp code which didn't
work.
Okay... just tried your new code. The phones keep resetting:
Error Verifying Config Info
then
Registering
you should use the CURRENT code,
Maybe it's not the new chan_sccp code that's the problem. When I put
in the SEP000785532D5F.cnf.xml on the tftp server, the phone never gets
to a usable screen. Instead it just tries to tftp files over and over.
Th one file, P00305000300.bin, I don't have. As far as I know I can't
get
Hi all,
I've been playing around with a cisco 7940 phone. It seems to like
talking to Asterisk with the chan_sccp plugin. The only problem is that
it tries to call out to a SEPX.cnf.xml file to verify it's
configuration. I've found docs for SEP*.cnf files, but not .xml ones.
Hi again,
I found chan_skinny and that seems to work pretty good. the SCCP one
filled out all the buttons really nice, but skinny seems to be
working.
How do I fill out the second line button on the phone with skinny.conf?
Thanks much!
...Jeff
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