I would just like to follow-up on the ringback
problem I'm getting from *. As I've said in my previous post, I am not
hearing the "real ringback" from the Cisco gateway terminating my call. I
don't want to provide false ringback from * (r option of dial), because it'll
still give me
Hi,
I'm interested in participating on the embedded side. One of our RD
labs is working on a number of embedded server solutions, including
servers that are built around a 3 square PCB, linked to a 2 square
PCB with a compact flash interface. It's robust, and up to military
standards (but
Any one has documented how-tos for making voicetronix openline 4 to work
with Asterisk.
I have been contacting Australian Digium resellers and Digium cards are
not approved in Australia. So I suppose Australian users are interested
into putting Voicetronix in use.
Any expereience to share
On Tue, 13 Jan 2004, Areski wrote:
Sorry Chris, actually, I cannot help you regarding your problem!
But I would like to know how allow an user to change of conferences (go
to an other room) !?!
When a user presses # he exits the conference. Then you just direct hiim
to another.
THat's not bad 20 calls through a 800Mhz P3. I new 3Ghz P4
could likely handle 60 then. Not bad.
But don't beleive top. First off if acverages. Think for
a minute. We all kow a CPU can never by 20% in use it is either
in an idle loop (at 0%) or doing real work (100%) it can't be
in an
On Tue, 13 Jan 2004, Christopher Arnold wrote:
i have a setup with chatrooms, several MeetMe conferences wich users can
change inbetween. 10 users maximum in each room.
It seems like when i have more than 40-45 users on the system at the same
time asterisk drops abt 20 and continnues
Steven Critchfield wrote:
On Wed, 2004-01-14 at 13:26, Jorge Mendoza wrote:
Hi,
A customer has an old PBX, which accept only T1 (not PRI) trunks. The
local telco only provides Euro PRI. Could the following config works?:
[telco] -- E1 PRI -- [Asterisk] -- T1 -- [PBX]
Many thanks for your time
--- calvis [EMAIL PROTECTED] wrote:
I am real close to finalizing my hardware selection for my Asterisk
test
machine. I am going to use the following hardware:
Dell 400SC w\Red Hat 9.0
1 - 4 Port TDM40B Card (FXS)
3 - Wildcard X100P Cards (FXO)
It does not matter if the PC is a
On Thu, 15 Jan 2004, Peter Pauly wrote:
Are there any cheap SIP phones (like the Grandstream
for example) that support power over ethernet?
SNOM-105, SNOM-200, and all Cisco phones should support PoE.
What is necessary to support SIP phones in a
Cisco Call Manager environment?
easiest
http://www.intel.com/software/products/ipp/samples_table.htm#
Has anyone taken a look at the value of these sample libraries for use in
Asterisk.
cameron.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I have a Dell 400sc sever on order. It will be shipped on the 27th. It is a
2.4GHz P4 with a 533 MHz front side bus, a 40GB disk, 128MB of memory, sound
card, ethernet, and year of on-site next day maintenance.
It is $318 delivered after rebates. Yes, $318.
This is a real server, by the way,
I am
rather
curious as to why I seem to be using up all memory although I am not
running
any unnecessary processes, or should I actually disable all modules,
other
than really necessary ones to support VOIP?
Do you mean that Asterisk is using up all of your memory
or that all of your
# ifconfig xl0
xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500
address: 00:01:02:78:11:e8
media: Ethernet autoselect (10baseT)
status: active
inet 203.219.167.126 netmask 0xfffc broadcast
On Fri, 16 Jan 2004, [EMAIL PROTECTED] wrote:
This is ifconfig on openbsd box:
fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500
I think this output shows that the fxp0 interface is on simplex mode.
Yes its in simplex mode, but this parameter is NOT related to half/full
duplex
I did initially, but I was having problems (possibly just in thinking it through)
getting the provided h323 driver to either
a) register as a gateway with my gatekeeper - that just does not seem to be and option
(please correct me if I'm wrong!!!)
or
b) setup a 'variable' extension (yes,
At one point I had Asterisk running on a Fedora Core 1 based embedded
system using a Soekris embedded device. Once the OS is running, the only
hard part is finding a source of timing for the MOH and conference calling.
However, I think the new Soekris units have a timing source on them (USB).
I do not have any zaptel hardware on the Asterisk box, I could not have
meetme functioning. I did modify the Makefile in zaptel directory on
line 168 by including ztdummy as one of the modules to compile in.
The error message from the concole:
-- Executing MeetMe(SIP/1002-e9ca, 4700) in new
Sorry for the malformed mail. My responses are marked with '***' below.
jesse
==
Hi,
I am a newbie in Asterisk as well, intending to use it in a similar way as
you are, communicating with AS5300 as well as other gateways including
MAXTNT.
I have had similar, but yet different experiences
--- [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Referring to my previous post about degradation of voice quality when
having more than 2 connection.
The actual route is:
pc xlite - local asterisk box - iaxtel - local asterisk
I have tried out a different situation:
pc xlite - local
The hardware drivers do seem to be migrating to 2.6. On the
other hand ztdummy has seen no love in awhile. For our environment
it is a better choice (limited slots, 3.3v instead of 5v, etc).
I'm no kernel programmer, but I've been working on at least clearing
the compile errors and loading of
Questions... happen to use webvmail?
bkw
On Thu, 15 Jan 2004, Brian Capouch wrote:
I have a user, running CVS a/o 11/23/03, who has complained about
phantom messages showing up days or even weeks after she has deleted them.
So I asked her to let me know when it happened again, and she
I know nothing about telephony ip phone etc.. however i have a few $$
that i am willing to spend to learn Asterisk . and i am very very
curious, i believe in learning by doing( but with some hand holding) so
i am looking for equipment suggestion . can anyone suggest a set of
equipment i could get
Hi, all !
I have a fast question, I am running a few Asterisk systems, but I just
noticed one thing quite peculiar. After I started safe_asterisk, and when
I ran PS or TOP, I could see 1 PID safe_asterisk and almost 10 PIDs
asterisk -vvvg -c even when there was no call. However, for the other
I just moved my system over to a new server with * 0.7.1. The old machine was
using a cvs from August/Sep timeframe.
On the new machine I did an make samples but then ovewrote with tar files of the
production configs in the
/etc/asterisk
/var/spool/asterisk
/var/lib/asterisk
folders.
Now the
The freenum.org project wants to use your trunks! The freenum.org project is an ENUM
parallel tree, which has as an eventual goal the distribution of ENUM numbering in
nations or areas which due to political or other issues are not able to get secure,
inexpensive, or functional ENUM
This is ifconfig on openbsd box:
fxp0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500
I think this output shows that the fxp0 interface is on simplex mode.
The voice degradation I referred was by using xlite soft phone. I open 2
line similtaneously and dial to FWD and back to
On Thu, 2004-01-15 at 18:27, Cameron Palmer wrote:
http://www.intel.com/software/products/ipp/samples_table.htm#
Has anyone taken a look at the value of these sample libraries for use in
Asterisk.
They might be useful, but you still have to license the codecs from the
patent holder, the
Jesse Peterson wrote:
I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM.
CVS UPDATE! That code is hardcore old.
Jeremy McNamara
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Thu, 2004-01-15 at 19:18, [EMAIL PROTECTED] wrote:
I do not have any zaptel hardware on the Asterisk box, I could not have
meetme functioning. I did modify the Makefile in zaptel directory on
line 168 by including ztdummy as one of the modules to compile in.
try modprobe ztdummy
On Thu, 2004-01-15 at 17:44, Chris Albertson wrote:
--- calvis [EMAIL PROTECTED] wrote:
I am real close to finalizing my hardware selection for my Asterisk
test
machine. I am going to use the following hardware:
Dell 400SC w\Red Hat 9.0
1 - 4 Port TDM40B Card (FXS)
3 -
On Thu, 2004-01-15 at 18:27, Cameron Palmer wrote:
http://www.intel.com/software/products/ipp/samples_table.htm#
Has anyone taken a look at the value of these sample libraries for use in
Asterisk.
Do you understand the GPL? Have you looked up the cost of those
libraries. They aren't free,
Thanx for all your help. I have been doing some research on shady dial and
also have been contacted by a few consultants, so hopefully I can have this
box up and running in the next few weeks.
thanx again
chris
From: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject:
Hi,
I'm trying to bundle the powers of Asterisk and SER.
Asterisk for pabx functionalities and termination to landline/PSTN, and
SER as SIP Gateway/Proxy.
With my current configuration the SIP user just adds 0 as a prefix to a
number, and the call will go out to PSTN over Asterisk.
For this to
Title: RE: [Asterisk-Users] capacity testing
Hi
all, and Jesse
1. So,
you did get the experience of crashing all of a sudden with the "Disconnected
from Asterisk server" error message. I got both this and the segmentation error
when crashing. I am running the version of asterisk, libpri
At work, we just put in managed switches... one user had lots of
collisions, which is strange for a switched network... we set the
computer to full/100, and the switch to the same settings, and now it
doesnt have any more collisions...
DH
Rich Adamson wrote:
This is ifconfig on openbsd box:
Brian West wrote:
Questions... happen to use webvmail?
Nope. All access is via a station dialpad. . .
This has been happening to her ever since we installed. It is really
freaky, because the higher-number messages are messages that she thought
she had deleted, and in her telling, Then, days
IF I want to play sound files,
1.) what format should it be? (*.au or*.wav)
2.) where should it reside?
3.) what syntax should I follow? Is
exten=_.,102,Dial(SIP/[EMAIL PROTECTED],1,tHA(sound.au))
correct? I tried this and it doesn't work.
Thanks,
Dear all,
Is * capable to use qmail as a MTA?
If so, how can I set it?
I'm using asterisk v0.5, and TDM30B (FXS), Wildcard X100P(FXO), and
x-lite(SIP softphone).
In zapata.conf, I put already callwaiting=yes. My PSTN doesn't not support
the callwaiting feature, so I don't expect the FXO is call
On Thursday 15 January 2004 20:02, T. Chan wrote:
I have a fast question, I am running a few Asterisk systems, but I
just noticed one thing quite peculiar. After I started
safe_asterisk, and when I ran PS or TOP, I could see 1 PID
safe_asterisk and almost 10 PIDs asterisk -vvvg -c even when
On Thu, 2004-01-15 at 19:18, [EMAIL PROTECTED] wrote:
I do not have any zaptel hardware on the Asterisk box, I could not have
meetme functioning. I did modify the Makefile in zaptel directory on
line 168 by including ztdummy as one of the modules to compile in.
try modprobe ztdummy
This
Hello All,
I received my Adit 600 yesterday and I have an 8 port FXO CAC FXO card
installed. That is the only module in the CB. I have the config on the *
side correct, however I am not sure.
I had the system running great for about 2 hour, and then it seemed to be
having problems. Incoming
I know, but as I mentioned in the inital post, I haven't been able to get the last 2
cvs versions I've pulled to run stable enough to test.
I've seen a 0.7.0 version number mentioned. Is there newer, mostly stable version of
code I should try that just hasn't been officially released?
jesse
Same here. I can't recreate the problem. I think this is a windows media
player issue.
bkw
On Thu, 15 Jan 2004, Troy Settle wrote:
I can't reproduce this either, but I do have the gsm codec installed (though
WMP won't play a .gsm file).
I play the wav49 files in Winamp with no issue.
Create a new wav49 on your system and play it.
bkw
On Thu, 15 Jan 2004, Warwick Ward-Cox wrote:
I'm having the same problem.
Warwick
- Original Message -
From: Jim Flagg [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 5:39 PM
Subject: [Asterisk-Users]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, 16 January 2004 10:13
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicetronix Openline 4 + asterisk
Any one has documented how-tos for making voicetronix
Hi all,
I'm in the process of building a * box for home and ran across the
vmail.cgi script. It installs suid root in order to allow access to the
voice mail boxes. I've never been fond of suid root and was looking for a
better method.
I've patched my installation to make everything in the vm
Or use this http://www.cam.org/~noelbou/1-step.html
bkw
On Thu, 15 Jan 2004, Troy Settle wrote:
I can't reproduce this either, but I do have the gsm codec installed (though
WMP won't play a .gsm file).
I play the wav49 files in Winamp with no issue.
--
Troy Settle
Pulaski Networks
On Thu, 15 Jan 2004 07:14:00 -0500, Andrew Kohlsmith wrote:
Is * capable to use qmail as a MTA?
If so, how can I set it?
It shouldn't be an issue, as qmail has the standard 'sendmail' binary
included.
Regards,
Andrew
In My * box, it has a running and working qmail (with sendmail and
On 15 Jan 2004, kemal asad wrote:
can anyone suggest a set of equipment i could get to check and test the
cool functionalities of Asterisk. Computer , Phones, communication
cards.
Digium sell an Asterisk Developer's Kit - for about US$180 you get an
FXO card and the 1-port version of their
you can do that. But are u installing qmail and * on
same box. i wont
recommend that. i use qmail and *. qmail is strictly
for internet email. *
is on separate server not exposed to Internet. * box
also has sendmail. i hv
configured sendmail to use smart host (qmail server).
This way its safe and
[EMAIL PROTECTED] wrote:
you can do that. But are u installing qmail and * on
same box. i wont recommend that. i use qmail and *.
qmail is strictly for internet email. *
is on separate server not exposed to Internet. * box
also has sendmail. i hv configured sendmail to use
smart host
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