I think dealer cost is about $1800.
plop sound of jaw hitting floor. Why not set up a T1 interconnect with
an asterisk box? I'm sure the $500 T100P and the cost of a T1 port is
smaller than the Dialogic card, the licenses for the dialogic driver,
and this device you mention. Not to
This is an idea from FreeSWAN, which was implemented in the recently released version
1.0.
Basically the idea is that FreeSWAN sites automatically encrypt traffic between them
when possible, without having to set up the link ahead of time.
How this works is:
The sites publish some info in DNS.
yeah, i got one, its called google :)
site:lists.digium.com searchphrase
put ^ in the google box, and voila
- wasim
On Sun, 8 Jun 2003, Matthew John Darnell wrote:
Does anyone have an application that will parse the archives so you can
search them? I was going to search the archives
Try Google with this as the query:
[phrase to search for] site:lists.digium.com
/* Tielman Koekemoer
Unix and Network Administrator at Vista University
Tel: 012-352 4093
Cel: 083-445 0019
*/
-Original Message-
From: [EMAIL
After scouring the list archive and not finding the answer I decided to
post to the list. I'm sure the answer is glaringly obvious but please
bear with me.
Using Asterisk, I'm tasked with setting up a SOHO with 5 analogue (FXS?)
lines and a number of soft-phones for internal extensions. I'm
Correction - My reference to analogue (FXS?) - Should be FXO
/* Tielman Koekemoer
Unix and Network Administrator at Vista University
Tel: 012-352 4093
Cel: 083-445 0019
*/
_
Content and Virus scanned
by Inflex
On Monday 09 June 2003 02:47, Matthew John Darnell wrote:
BTW, I think it has been covered here before that the D41 is a half
duplex card and wouldn't be good for conferencing.
--
Steven Critchfield [EMAIL PROTECTED]
Does anyone have an application that will parse the archives so you can
Hi all,
Finally I found why MP3Player was not working for me.
In the CVS of two weeks ago the path to mpg123 was hardcoded to
/usr/bin/mpg123.
I installed the latest pre0.59s because previous releases were not working
for me because of my fast Pentium IV 1,7Ghz processor. This release but
probably
Hi list,
I have an E400P using only one span with 4 channels, using EM immediate
signaling.
/etc/zaptel.conf
span=1,1,1,cas,hdb3,yellow
em=1-4
loadzone = us
defaultzone=us
-
/etc/asterisk/zapata.conf
-
Are you sure that you compiled zaptel for __SMP__ ?
Edit your zaptel/Makefile.
0: 75283844 75241320 75286285 75247088IO-APIC-edge timer
1: 1 0 1 1IO-APIC-edge keyboard
2: 0 0 0 0 XT-PIC
On Tue, 10 Jun 2003 09:37:22 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Try in /etc/zaptel.conf to add this line:
alaw=1-4
sine by default EM is used in US and the ulaw codec is being used.
Martin
thanks for your reply, but it still doesn't work
Eduardo
Did you do ztcfg after you added that line ?
Martin
On Tue, 10 Jun 2003, Eduardo Goncalves wrote:
On Tue, 10 Jun 2003 09:37:22 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Try in /etc/zaptel.conf to add this line:
alaw=1-4
sine by default EM is used in US and the ulaw codec is
Hi,
Has anyone done anything with the Linux advanced routing stuff to give
SIP and IAX traffic priority?
What I have in mind is a high-pri queue for voip traffic, all the rest
in another queue that gives way to the VOIP stuff.
Thanks,
Steve
___
On Tue, 10 Jun 2003 10:08:02 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Did you do ztcfg after you added that line ?
Martin
yeap :~
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From the same machine running asterisk or from
a linux router ?
Linux kernel by default prioritizes traffic if
the packet has some TOS bits set. so a standard
linux router should do a basic traffic shaping.
Of course, more complex rules could be made...
but if the *outside* world don't do
H, I to appear to have an odd mix of interrupts. It seems that the second CPU
doesn't do much
at all on my dual Xeon...
CPU0 CPU1
0: 40652580 0IO-APIC-edge timer
1:926 0IO-APIC-edge keyboard
2: 0 0 XT-PIC
Hi all,
i have read in the * whitepaper the following:
s: The start extension. A call which does not have digits associated with
it (for
example, a loopstart analog line) begins at the s extension.
I think this means the s extension will be execute when the phone is picked
up.
But when i pick
I'm back on the paging again - still can't get it working as I wish.
I have listed two attempts below where I run into basically the same problem
[pagejon] - I drop a file in the spool directory that starts that context,
I must use a device though that the outbound call is placed by. The phones
My dual-proc Xeon boxes didn't share IRQs across CPUs until I installed
the kernel-utils RPM and made sure the irqbalance service was
running... Just a word to the wise!
Jared Smith
On Tue, 2003-06-10 at 09:52, [EMAIL PROTECTED] wrote:
H, I to appear to have an odd mix of interrupts. It
To execute the s extension automatically when you pick up the phone,
you need to put that channel in immediate mode. (I'd tell you how to do
it, but I can't remember the syntax off the top of my head.)
Jared Smith
On Tue, 2003-06-10 at 09:57, Thomas Haeger wrote:
Hi all,
i have read in
That's a local phone.
if u what a local phone to exec 's' extensions,
put immediate=yes into zapata.conf .
Otherwise, you'll get a dialtone waiting for a exten input.
Matteo.
Il mar, 2003-06-10 alle 17:57, Thomas Haeger ha scritto:
Hi all,
i have read in the * whitepaper the following:
the channel has to be in immediate mode to work as you describe with s
otherwise nothing happens until you type some digits that match something
in the context the phone starts in.
At 05:57 PM 6/10/2003 +0200, you wrote:
Hi all,
i have read in the * whitepaper the following:
s: The start
Title: RE: [Asterisk-Users] Setting local IP address for the RTP port
Listening is not a problem. When we send RTP packets it's important
to make sure we use the specific interface. For example, one interface
is on internal subnet and the other one is on external. QoS etc.
Do you think
Maybe it's me, but it looks like you need to change to AGI instead of
extension logic for parts of this below.
Write a AGI app that does the authenticate, and record of message. If
the user hangs up the phone it is fine since the AGI still is running,
and can then submit a qcall to then start the
Hi,
trying to build the h323 channel driver that comes with asterisk works
fine, but only as long as I use openh323-1.11.7.
Unfortunately, that setup seems to have a bug which misguides one of the
audio streams. (So while * can hear me, the phone remains silent.)
I suppose that bug is fixed at
Ditto. I think vendor help/hints/suggestions/clarifications on this list are
extremely helpful and valuable. I hate spam as much as anybody but we need
to become evolved enough to know the difference.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jon
Hello,
I've been working with the chan_h323 myself, and I had several problems,
but finally got it working.
I had to do things in the following order:
(1) build and installed asterisk as root
(2)I built pwlib and openh323 into my home directory (not root) and
built them there as me, I
If you would have followed the build instructions laid out by the Open
H.323 folks you wouldn't have had to go thru all of that.
http://www.openh323.org/build.html
(Notice they NEVER tell you to make install ANYTHING, there is a reason
for that)
Jeremy McNamara
Kelly McDonald wrote:
On Fri, May 30, 2003 at 02:59:45PM +, Joe Antkowiak wrote:
how do I specify a different one for the voicemail script?
This is what the top of my asterisk.adsi looks like:
DESCRIPTION Asterisk PBX ; Name of vendor
VERSION 0x00; Version of stuff
FDN 0x85efd9da
SECURITY 0x78921d49
I currently have two fax machines on my system.
Both of them seem to send and receive very slowly. My end users
are complaining; saying it was faster before we moved to * (Straight
Analog Lines)
Any help would be great.
PS: I already have the d option on the Dial line.
Both fax machines are
I wish Mark would chime in on his thoughts.
I also am not opposed to someone answering direct questions about a
product they happen to sell even if it is in a public forum. We all gain
from it in that anyone searching google may see both asterisk and
Budgetone. Both may get a plus by this.
On
On 10 Jun 2003, Emanuele Pucciarelli wrote:
Il mar, 2003-06-10 alle 17:19, Stephen Davies ha scritto:
Has anyone done anything with the Linux advanced routing stuff to give
SIP and IAX traffic priority?
What I have in mind is a high-pri queue for voip traffic, all the rest
in
On Tue, Jun 10, 2003 at 04:30:50PM +0600, [EMAIL PROTECTED] wrote:
c) get a T400P + channel bank (expensive, but it does give you 24 ports)
Small typo. Just don't want newbies to be confused. The T400P
mentioned above should probably be a T100P which supports one T1 line
rather than four.
--
On Tue, Jun 10, 2003 at 11:21:37AM -0300, Eduardo Goncalves wrote:
Hi list,
This configuration works ok, I can dial on Zap/g1. But, when the
other side answer the call, I only hear a lot of noise instead
of the voice. Could anybody help me?
Is the noise loud and sounds
On Tue, Jun 10, 2003 at 07:08:45PM +0200, Sergio Serrano Revuelto wrote:
HI all,
we get a TDM10B to probe it. I find two problems:
-First, I hear a lot of noise in communication. I have tried do
dd if=/dev/zero of=/dev/null but it isn't work.
-Second, When I pickup
Is it possible for two PDA's to communicate like telephones via SIP channels
on a PC running Asterisk? If that is possible, does there exist any
applications that can be installed on a Zaurus 5600, which is a PDA with an
Xscale processor running on a Linux OS, that can essentially turn it into
Does anyone know of drivers/software that will allow
me to use the old Rhetorix 4108 T/R boards with
Asterisk?
Thanks!
Chip
__
Do you Yahoo!?
Yahoo! Calendar - Free online calendar with sync to Outlook(TM).
http://calendar.yahoo.com
On Tue, 2003-06-10 at 13:49, flickds wrote:
Is it possible for two PDA's to communicate like telephones via SIP channels
on a PC running Asterisk? If that is possible, does there exist any
applications that can be installed on a Zaurus 5600, which is a PDA with an
Xscale processor running
On Tue, 2003-06-10 at 13:59, Chip G wrote:
Does anyone know of drivers/software that will allow
me to use the old Rhetorix 4108 T/R boards with
Asterisk?
Not supported yet. First you need to see if you can get them to work in
linux. Then if you are still in need, there is sample drivers around
Thanks for the help. I was able to get my application to load with
Asterisk, now I just need to get it to work. After reading your comment, I
don't know that I fully understand what's going on as far as the channels
and extensions. Are you saying that the MWI is tied to the channel? If
that is
If there is a way possible, would someone tell me how I can
setup a dial by name feature under vmail2?
Thanks,
Kim Callis
Can I use a WILDCARD TDM400P to connect to
four Telco circuits aka FXO? Or will I need
four Wildcard X100P?
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On Tue, 2003-06-10 at 16:22, Johnny Witt wrote:
Hi Asterisk-Users
Ive been reading about the Asterisk project (all that I could get my
hands on J ). It sound to good to be true. But Ive got some questions
which I havent found a answer to anywhere :
1) Can I use Asterisk as a Call
On 10 Jun 2003, Emanuele Pucciarelli wrote:
That is not entirely correct. There is an output queue, and pfifo_fast
is the default (see the LARTC Howto, 9.2.1.1). But you are right when
you say you need something to slow down the data;the simplest choice
should be addingthe Token Bucket
Use tkcPhone, it's a SIP agent for the Zaurus.
http://www.thekompany.com/embedded/tkcphone/
flickds wrote:
Is it possible for two PDA's to communicate like telephones via SIP channels
on a PC running Asterisk? If that is possible, does there exist any
applications that can be installed on a
Please... stay away from the FXS to FXO converters The cheap white box
sold direct at $18 bucks and on the net for $70-$300 bucks work.. but do
you want to rely on a timer to disconnect the call after hangup?
-Greg
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To:
have a look on th zaurus site, there is a pay for one
On Tue, 10 Jun 2003 13:49:20 -0500, flickds wrote:
Is it possible for two PDA's to communicate like telephones via SIP channels
on a PC running Asterisk? If that is possible, does there exist any
applications that can be installed on a
On Tue, Jun 10, 2003 at 09:58:14PM +0200, Emanuele Pucciarelli wrote:
Il mar, 2003-06-10 alle 20:07, Stephen Davies ha scritto:
When the tos option is set correctly (to nodelay), the default
queueing in recent kernels already does that, because the pfifo_fast
queue is used (if I recall
On Tue, Jun 10, 2003 at 10:14:09AM -0600, Jared Smith wrote:
My dual-proc Xeon boxes didn't share IRQs across CPUs until I installed
the kernel-utils RPM and made sure the irqbalance service was
running... Just a word to the wise!
Yes, you need irqbalance and a kinda modern kernel in order to
On Tue, 10 Jun 2003, Jeremy McNamara wrote:
trying to build the h323 channel driver that comes with asterisk works
fine, but only as long as I use openh323-1.11.7.
Unfortunately, that setup seems to have a bug which misguides one of the
audio streams. (So while * can hear me, the phone
-- Forwarded message --
Date: Wed, 11 Jun 2003 01:10:16 +0200 (CEST)
From: Siggi Langauf [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
On Tue, 10 Jun 2003, Jeremy McNamara wrote:
If you would have followed the build
On Tue, 10 Jun 2003 [EMAIL PROTECTED] wrote:
H, I to appear to have an odd mix of interrupts. It seems that the second CPU
doesn't do much
at all on my dual Xeon...
You might have 'noapic' on your kernel command line... or your bios isnt
configured for MP 1.4 ...
-Dan
Aloha,
Does anyone have any screen shots of the Asterisk admin GUI?
I couldn't find any links in the archives or the asterisk web site.
-Matt
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Hi:
Any of you know a good web admin for asterisk???
Thanks
Alvaro Parres
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This mail sent through IMP: http://horde.org/imp/
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You should investigate TRIP (RFC 3129):
http://www.zvon.org/tmRFC/RFC3219/Output/
Find BSD-licensed proof-of-concept code at
http://www.vovida.org/downloads/trip/trip-1.0.0.tar.gz
If someone could incorporate this into Asterisk and extend the
functionality, that would be pretty nice. The
quote who=Scott Lambert
On Tue, Jun 10, 2003 at 02:10:56PM -0500, Steven Critchfield wrote:
On Tue, 2003-06-10 at 13:49, flickds wrote:
Is it possible for two PDA's to communicate like telephones via SIP
channels
on a PC running Asterisk? If that is possible, does there exist any
On Tue, Jun 10, 2003 at 09:59:03PM -0700, Robert Hajime Lanning wrote:
quote who=Scott Lambert
On Tue, Jun 10, 2003 at 02:10:56PM -0500, Steven Critchfield wrote:
On Tue, 2003-06-10 at 13:49, flickds wrote:
Is it possible for two PDA's to communicate like telephones via SIP
channels
We're doing a new * installation at a remote office soon, and I was just
curious what people's opinions were on hardware these days .. I've had
decent luck with T100Ps and Adtran, but I know times change ..
I'm looking to do roughly 15 handsets and 15 pstn, with some room to
grow. I had
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