On Saturday 17 January 2004 00:31, Chris Albertson wrote:
Software RAID vs. Hardward RAID???
Welcome to the 80s.
There IS no Hardward RAID it's all software the difference is
only where the software lives, in ROM on the controler card in
the RAID box or in a Linux driver.
Actually, hardware
On Friday, 16 January, 2004 12:27, Steven Critchfield wrote:
On Fri, 2004-01-16 at 06:47, Andrew Kohlsmith wrote:
If you value your data, don't use software raid. If you value
performance don't use software raid. If you value uptime/stability
don't use any raid on IDE.
That's pure
Paul Crick wrote:
If you want some classic office phone ringers:
http://www.leonine.com/~lion/phones.php
These are Merlin rings. For some, myself included, they're
a bit nostalgic.
But cool with it! Now I can make my phone sound like those ones on 24 ;-)
Not being too familiar with the Merlin
Philipp von Klitzing wrote:
Hi there,
whenever I use a macro to dial out I see only s recorded in the dst
field of the CDR. Is there anyway to get around that problem except for
not using a macro?
Example:
)
Try to match every extension before dialing out instead, using s is a bad thing for
Hi there,
After a lot of valuable insights from the list, incoming and outgoing calls finally work through OpenLine4! Thanks for all the input!
Now I have 2 minor issues:
Sometimes Voicetronix dials too quickly before an actual dial tone is obtained from the phone company. E.g.
I solve it for h323 in follow way:
1. Exclude all codecs except g723.1 from h323.conf:
disallow=ULAW
allow=g723.1
2. Add format_g723 module (http://www.agk.nnov.ru/format_g723.c.gz)
into project
3. convert all wav and gsm sound into g723 format (use lbccodec from
g723_1 demo
Saturday, January 17, 2004, 12:49:26 AM, Eric Wieling wrote:
EW You can purchase the G.723.1 reference code from the ITU, then you'll
EW need to make it work with Asterisk
I made codec_g723 with this code, but for compression of PCM file 12
sec long requires 37 sec :) (2x600MHz server)
So my
Hi! Thanks for your suggestions.
How easy is it to use the A and B number of the incoming calls in
certain scripts (e.g. are functions present to read them and use them
for further operation)
I have not worked with AGI scripts before and probably need to have a
look at this. Is the AGI capable
Hello..
I have a customer who wants to connect 2 PBX's over
IP..
The setup should look like this:
[PBX] -- SS7 -- [Asterisk]
--IAX -- [Asterisk]
-- SS7 -- [PBX]
Since there are no SS7 cards , I was thinking at a
way of carrying the E1 data as bulk...Can I do that ? How ?
Is possible a
But now i'm stumbling on another problem.. Asterisk seems to want
to send the SIP udp packets directly to the SIP clients.
In the case of a SIP user/client behind a NAT, this obviously doesn't
work.
Have you tried reinvite=no in your [ser] section of sip.conf?
P
Hi!
I thought I was making progress on my dialplan when I realized that the
class features that are available for zap channels aren't available for
SIP channels.
http://www.voip-info.org/wiki-CLASS
3. Anyone willing to share some of their cool features that they've
come up with ??? I'd
Dear
sorry for my next post on the same object but i need to know if is possible
to hack libPRI to have the function that KAPEJOD made in CAPI for the early
B3 function.
I'm the only in the list have this type of problem with Free Message? :-)
Thanks in advance
Dimitri
Alexandru Coseru wrote:
I have a customer who wants to connect 2 PBX's over IP..
The setup should look like this:
[PBX] -- SS7 -- [Asterisk] -- IAX-- [Asterisk] -- SS7
-- [PBX]
If you succeed in doing this, i.e., turning asterisk into a softswitch, you'll
be quite famous, at
I see a lot of chatter in the archives about intercom and paging, but
has anyone addressed zone paging? Each of the 50 telephones in a large
clinic would be members of one or more paging zones. Someone could then
page Dr. X in zone #1. Would this be possible with analog phones? SIP?
Hi again,
I've finally got Voicetronix OpenLine4 working so am a happy man ... thanks
again to all those who helped! Just a few outstanding questions of curiosity
:
1. I have finally got my setup to work by allowing ONLY g711alaw and nothing
else. Why should enabling a few extra codecs cause
I just started getting the following
notice message and was wondering what it meant.
Jan 16 15:56:11 NOTICE[240654]: File
rtp.c, Line 263 (process_rfc3389): RFC3389
support incomplete. Turn off on client if
possible
Todd Wallace
Hi,
As I have said before I am not that hot on building kernels and what
effects of various options would have on Asterisk..
I am thinking of using a distro called Trustix to build my new Asterisk
server.. Its super small and is built for security so everything is
disabled by default which is
Sorry for the fragmented messages from me - one last thing I forgot to ask
in my last post.
When incoming calls come to us, our PSTN line is picked up almost
immediately - and then asterisk will proceed to dial the SIP extensions.
During this time the caller hears dead slience - obviously not
there isn't already in the wiki... this is really a FAQ!
btw, that means that your device is using silence suppression.
since * doesn't support that, it issue the NOTICE below.
that's not harmful, but if you're annoyed by those msgs, just
turn off silence suppression in your device.
Jan 16
I see a lot of chatter in the archives about intercom and paging, but
has anyone addressed zone paging? Each of the 50 telephones in a large
clinic would be members of one or more paging zones. Someone could then
page Dr. X in zone #1. Would this be possible with analog phones? SIP?
Use music on hold, and tell the dial app to use it..
Luciano
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Terence
Parker
Enviado el: Sábado 17 de Enero del 2004 11:38
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Playing background message
Sorry for
hi
The caller will hear a recorded message, followed by music. What I want is
the caller to hear this WHILE the SIP phones are ringing - but using the
'Background' option in extensions.conf seems to make it so that my SIP
phones won't be dialled until AFTER the music clip is finished - i.e.
When incoming calls come to us, our PSTN line is picked up almost
immediately - and then asterisk will proceed to dial the SIP extensions.
During this time the caller hears dead slience - obviously not very good as
some would think the line just went dead and hang up. I have toyed with the
Here's how Mark wants it done:
In the channel_pvt structure, we have a pointer to a new structure:
struct ast_common_features {
char fwd[AST_MAX_EXTENSION];
char fwd_off[AST_MAX_EXTENSION];
.
.
.
};
This allows the codes to be redefined per
That's about the scale we're planning per server. We just purchased a Dell
PowerEdge 1600SC for the job (able to go dual Xeon, but we're starting with
one 2.8GHz, will add a second and recompile if we need 2 procs).
The nice thing about this box is that it seems to have 2 of each kind of PCI
Is there a way to tell which device it is coming from?
We have Grandstream phones and it is intereconnected with a Nextone MSW
using SIP.
Todd
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni
Matteo
Sent: Saturday, January 17, 2004 8:41 AM
Terence Parker wrote:
Hi again,
I've finally got Voicetronix OpenLine4 working so am a happy man ... thanks
again to all those who helped! Just a few outstanding questions of curiosity
:
1. I have finally got my setup to work by allowing ONLY g711alaw and nothing
else. Why should enabling a few
T == T Chan [EMAIL PROTECTED] writes:
T Thanks alot for your explanation. Can you tell me if there is a way
T to confirm if I have the nptl in the boxes ?
grep for nptl in the installed pthread libs:
grep -i nptl /lib/libpthread.so.0 /usr/lib/libpthread.a
does it on my box.
-JimC
On Sat, 2004-01-17 at 01:33, [EMAIL PROTECTED] wrote:
Thanks guys, thought SER had to 'register' to be able to use
any Asterisk contexts.
But just defining a new entry in the sip.conf with just context ip worked!
But now i'm stumbling on another problem.. Asterisk seems to want
to send the
Rich Adamson wrote:
Unless I'm missing something here, from the CLI do a 'show application dial'
and checkout the r option, as in:
exten = 3015,1,Dial(SIP/3015,15,tr)
I agree, I'd rather have the caller hear ringing instead of MOH as
ringing gives the caller some feedback as to what is
You can get a DELL 400SC server for $400 or so. www.dell.com
Check out the refurbished units, although lately the new ones have been a
better deal.
Paul
Paul Mahler
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aram
Ter-Martirosyan
Sent: Friday,
There are a number of paging interfaces available which connect to a regular
phone line on one side
and to a paging amplifier on the other side.
Use one or more of these - connect them to FXS cards either Digiums 4xFXS
or a channel bank and make the extensions your paging zones in the dialplan.
Hello
Does anybody has a good ideea for combining 2 audio
frames (from 2 different channels) in order to send them to an 3rd channel
? I can't send them both cause it seems that i'm overflowing the 3rd
channels and it drops the 2nd frame (kind of logic, though)..
I have two channels ,
Hi Jan,
in the sccp_registration i would then handle the registration for the
7920 how the callmanager is behaving.
I've just gotten one step further with my 7920:
Got it successfully registered to asterisk. Still doesn't actually work, but
definitely a step in the right direction.
The problem
Quoting Brian West [EMAIL PROTECTED]:
Works fine here.. got two of em.
bkw
Hmpf! I donno whats wrong then, both phones do the same thing.
So you can keep the headset in the cradle, hit the 'speaker' button, dial a call
and it doesn't disconect ?
I wonder, are you using an xml dial plan
I am new to asterisk and am wanting to know if it can do some things:
in a large/ distributed environment users move about either office to
office or branch to branch can they log in and have their virtual
extension routed to the one they are on?
naturaly this implies the question: if branch
Chris Lee wrote:
I am new to asterisk and am wanting to know if it can do some things:
in a large/ distributed environment users move about either office to
office or branch to branch can they log in and have their virtual
extension routed to the one they are on?
naturaly this implies the
Title: Asterisk/X100 - Sipura Configuration
I have a Sipura behind my Asterisk Box. I have a X100 card in the box. Calls are coming in ok. When I try to configure my extensions for dialing out, I can dial
9X and then get a fast busy.
I followed an example from moxilla on the Spirua
I am experiencing a problem that from list archive it appears others are
running into. When I dial from Cisco 7960 via the * to Free World
Dialup
destinations that supports G.729 the call fails. The major error from
the debug log is
Jan 15 00:11:14 NOTICE[22545]: channel.c:1481
On Fri, 2004-01-16 at 16:55, Robert L Mathews wrote:
At 1/16/04 7:25 AM, Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
That's pure bullshit -- I use software RAID *specifically* because I value
my data. I don't want to buy two hardaware RAID controllers to have one
sit on the shelf just in
Hi!
in a large/ distributed environment users move about either office to
office or branch to branch can they log in and have their virtual
extension routed to the one they are on?
http://bugs.digium.com/bug_view_page.php?bug_id=102
.. currently its not somthing that is
/20040117.newsounds.tar
All of the sounds in that tarball are also in the main ../sounds/
directory in individual .gsm format.
I've created a directory called ../sounds/AIF which contains the
high-quality versions of the new sounds that I'm clipping apart, for
those of you who want to process them
I did find something interesting. If you set reinvite=yes then * can
setup the RTP stream so that it avoids the media proxy in the * box
completely. I haven't tested to see if it changes anything.
- Dustin -
[EMAIL PROTECTED] wrote:
I am experiencing a problem that from list archive it
Dustin Goodwin wrote:
I did find something interesting. If you set reinvite=yes then * can
setup the RTP stream so that it avoids the media proxy in the * box
completely. I haven't tested to see if it changes anything.
Can we please kill reinvite - it does not exist in the SIP channel as an
I got 1.0.4.38 from SIPphone's server at 130.94.123.253 but last time
I
tried it was offering 1.0.4.35. For me 1.0.4.38 cleared all my
problems
but from SIPphone's email it had hosed some phones, such that they
were
talking about replacement units.
I'm still on fscking 1.0.3.81,
Maybe someone will write a patch to print an error to the console if
reinvite= is found in the config file.?
On Sat, 2004-01-17 at 15:44, Olle E. Johansson wrote:
Dustin Goodwin wrote:
I did find something interesting. If you set reinvite=yes then * can
setup the RTP stream so that it
Can we please kill reinvite - it does not exist in the SIP channel as an
option for anything. Period.
There is an option called canreinvite that you can set to yes or no.
Setting reinvite to anything will not change anything at all.
Olle,
I thought I was the only that was loosing it with
On Saturday 17 January 2004 15:44, Olle E. Johansson wrote:
Dustin Goodwin wrote:
I did find something interesting. If you set reinvite=yes then *
can setup the RTP stream so that it avoids the media proxy in the *
box completely. I haven't tested to see if it changes anything.
Can we
I've just noticed that since swapping from the direct mysql cdr driver to
cdr_odbc, the call duration (and anything else that's an integer) isn't
logged - anyone else seen this and know the reason. The cdr_odbc driver
gives no error messages and records any string based fields correctly.
Check your tables. I logged everything as integer.
set verbose 10 and make a call and watch it.. then do reload and watch the
output. It will unload and reload and you can check to make sure your
accually connetcing to the database.
bkw
On Sat, 17 Jan 2004, Iain Stevenson wrote:
I've just
Their is no need to telnet with perl you can just shove a notify packet
down the cisco and let it reboot on its own. Really easy.
bkw
On Sat, 17 Jan 2004, Steven Critchfield wrote:
On Fri, 2004-01-16 at 15:26, B. J. Bomar wrote:
Yes, I was wanting to do it via a script, but telneting in will
The soundfiles I submitted earlier today have been cleaned up, and
added to the Digium CVS server in a more formal manner. Also, some
of the really bad formatting in my .txt description file has been
rectified. All of the sounds on my website are now on the Digium
site, and I will be
I see a lot of chatter in the archives about intercom and paging, but
has anyone addressed zone paging? Each of the 50 telephones in a large
clinic would be members of one or more paging zones. Someone could then
page Dr. X in zone #1. Would this be possible with analog phones? SIP?
Thanks for the replies
I've decided to simply add 'm' to the dialplan for now, but i'll investigate
call queues later - this sounds like the ideal setup for me though.
For the meantime though, music on hold works fine!
Thanks again.
Terence
I agree, I'd rather have the caller hear
Come monday I will see if I can get the PRI line working if we have an
extra 767 circuit pack. I promise that if/when we get this working I
will definitely write up a detailed explanation of the steps involved.
Right now we have a partial setup but a fully integrated box seems
within reach...
Here's a simple small expect script ...
I call it phreboot, usage: phreboot IP
$ phreboot 10.99.1.1
-- cut here --
#!/usr/bin/expect -f
set timeout -1
spawn $env(SHELL)
match_max 1
send -- telnet [lrange $argv 0 0]\r
expect -exact word :
send -- cisco\r
expect -exact Phone
send --
--On Saturday, January 17, 2004 8:49 PM -0500 John Todd [EMAIL PROTECTED]
wrote:
SNIP
Ideas welcome for more text; I may have another timeslot with Allison
early next week in which there will be some leftover room for additional
words. Short phrases and meaningful sets of words for existing
If what u mean by CLASS is Class of Service, ie: the ability to
allow/denny access to users to/from resources like public network based on
the number they dial, this can by nicely achieved by using a powerful tool
that * calls context.
Playing with contexts you can define several different
I just found another thread showing where the files are hiding. Thank you
for recording the first two below already - greatly appreciated. Please
note that the third is a new one, however.
For exact intonation of the below, Allison can dial 805/692-2323 and then
x234. You'll hear the first
Hi,
Just wondering if anyone else in Australia is using the X100P to
connect to the PSTN, and what configs they have for it?
Im finding at present when I make a call I get a fair bit of
echo of myself speaking, and also the person on the other end cant hear
me very well (perhaps need
At 7:25 PM -0800 1/17/04, Ken Alker wrote:
--On Saturday, January 17, 2004 8:49 PM -0500 John Todd
[EMAIL PROTECTED] wrote:
SNIP
Ideas welcome for more text; I may have another timeslot with Allison
early next week in which there will be some leftover room for additional
words. Short phrases
Matthew Branton wrote:
Come monday I will see if I can get the PRI line working if we have an
extra 767 circuit pack. I promise that if/when we get this working I
will definitely write up a detailed explanation of the steps involved.
Right now we have a partial setup but a fully integrated box
Hello all,
I have a new Carrier Access Adit 600 and am having a problem dialing out
and receiving incoming calls.
After the initial ring, the calling party will here nothing. There is a
silence, but the SIP phones that is supposed to ring will ring. The
calling party will not hear the sound of
You can easily have an incoming call ring multiple extensions. You could
also send the incoming call to an alternate extension.
Paul Mahler
mail:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Saturday,
On Saturday 17 January 2004 23:16, Brent Franks wrote:
I have a new Carrier Access Adit 600 and am having a problem dialing
out and receiving incoming calls.
After the initial ring, the calling party will here nothing. There
is a silence, but the SIP phones that is supposed to ring will
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