Hi, here it is...
[EMAIL PROTECTED] asterisk]# cat cdr_mysql.conf
;
; Note - if the database server is hosted on the same machine as the
; asterisk server, you can achieve a local Unix socket connection by
; setting hostname=localhost
;
; port and sock are both optional parameters. If
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 02 February 2004 16:53, Senad Jordanovic wrote:
As I've been unable to get app_transfer to work, could someone
explain how it is supposed to work?
Currently I have two Asterisk boxes. A call comes in via zaptel to
ast1. ast1
dials
Help me
i'm managing a call center with asterisk, GS 102 and diva server 4 bri.
i have big problem with big choppy sound, In the direction External
user --- Agent
Please aid me!!!
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Several people have requested more information on my cluster setup, I'll try to put
something together today but things are very busy here at the moment ... but keep an
eye for a mail today ...
-Original Message-
From: David Luyens [mailto:[EMAIL PROTECTED]
Sent: 03 February 2004 07:39
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 02 February 2004 16:53, Senad Jordanovic wrote:
As I've been unable to get app_transfer to work, could someone
explain how it is supposed to work? Currently I have two Asterisk
boxes. A call comes in via zaptel
Hi all,
I playing around with some C7912 IP phones (SIP FW).
They work nice with asterisk, but I found the following issue:
o When I configure the voicemail number (8500) to access VM
I can push the messages button on the phone to access my VM
o The phone can setup DND and call
On Tuesday 03 February 2004 11:36, Swen Veckes wrote:
Hi all,
I playing around with some C7912 IP phones (SIP FW).
They work nice with asterisk, but I found the following issue:
o When I configure the voicemail number (8500) to access VM
I can push the messages button on the phone to
Hello all,
As I'm sure is pretty common, I have some extensions that dial mobile numbers
after a local timeout. I would like to prompt the caller to record their
name after the local timeout and have the recipient be able to hear the name
prior to accepting the call.
Recording the message
Hi
When I call a phone with CAPI if the phone available I hear ringing ok
but if the phone is busy I don't hear anything at all.
Also, when I call a mobile phone and it is turned off I don't hear the
operator voice answer me telling me that the request phone is turned off
or unavailable.
Any
show application dial from asterisk cli:
snip
't' -- allow the called user transfer the calling user
'T' -- to allow the calling user to transfer the call.
'r' -- indicate ringing to the calling party, pass no audio until
answered.
'm' -- provide hold music to the calling
I wish 'A(x)' was available with AgentCallBackLogin!! :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matteo
Brancaleoni
Sent: Tuesday, February 03, 2004 6:48 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Playing announcement to called user
Having exactly the same problem with 7905. In addition it doesn't
seem you can
disable (at least on my sip fw release) the
redirect-to-vm-on-busy feature.
Yes, that's right, only the value for no answer can be changed (set to
high value == disable).
Whenever the phone has the VM number
Does anyone know if this feature is actually implemented? I just tried
it with a Dial statement of mine and it doesn't play any file. Doesn't
report any errors, and I'm sure the file exists.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew B
Dear Matthew
yes it work great A(playmex) where playmex is gsm file in sound dir.. i have
made some simple hack to app_dial.c to have a new option B(playmex) with it i
can play a mex to the caller when the call is connected i use it to play a
dtmf code...
Thanks in advance
Dimitri
PS:
go with early b3
matteo.
Il mar, 2004-02-03 alle 12:44, Matteo Rancilio ha scritto:
Hi
When I call a phone with CAPI if the phone available I hear ringing ok
but if the phone is busy I don't hear anything at all.
Also, when I call a mobile phone and it is turned off I don't hear the
Hi,
I'm trying to get my X100P to Dial the following sequence to gain access to
speed dial numbers on my Norstar PBX that the X100 is connected to...
[FLASH] [*] [0] [22] (where 22 is the speed dial number)
But so far I've had no luck, with the following extension:-
exten =
Ciao Matteo,
I tried with these ones but nothing change much:
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+CAPI+readmediff=2
My extensions are
[outgoing]
exten = 0,1,Goto(outgoing-isdn,s,1)
[outgoing-isdn]
exten = s,1,NoOp()
exten = _X.,1,Dial(CAPI/mynumber:b${EXTEN}|30)
I am new to the list and I apologize for being late to the party. I have a
couple of Cisco phones that I cannot get to register with *, any advise
would be appreciated.
Thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi all.
I'm evaluating a mediatrix 2fxs 1102.
seems great (it has also supervised transfer, that's
very needed in office environments and works well).
the only I thing I cannot make work is the auth
to my asterisk server.
If I don't set a password into the mediatrix and
*, I can call out, but
Hi
i'm managing a call center with asterisk, GS 102 and diva server 4 bri.
i have big problem with big choppy sound, In the direction External
user --- Agent
after a quick phone call with Cristian, we
managed to find out 2 things :
* hypertreading was enabled and that caused irq errors
*
Dan Iordanescu wrote:
[snip]
1. How do you switch the card from ISDN PRI TE to NT? This means from
being configured as User Equipment (the PABX) to be Network Equipment
(the Exchange).
This is done in the /etc/asterisk/zapata.conf file.
2. How do you configure the card for E1 or T1?
This
Hi,
that's my sip.conf entries for my Cisco 7060 Phones:
[general]
port = 5060 ; Port to bind to
bindaddr = my IP of * Server ; Address to bind to
context = intern; Default for incoming calls
disallow=all; Disallow all codecs
allow=alaw
I'm evaluating a mediatrix 2fxs 1102.
seems great (it has also supervised transfer, that's
very needed in office environments and works well).
the only I thing I cannot make work is the auth
to my asterisk server.
If I don't set a password into the mediatrix and
*, I can call out, but
At 10:59 PM +1000 2/3/04, Christopher Lee wrote:
Hi,
I'm trying to get my X100P to Dial the following sequence to gain access to
speed dial numbers on my Norstar PBX that the X100 is connected to...
[FLASH] [*] [0] [22] (where 22 is the speed dial number)
But so far I've had no luck, with the
John Bittner wrote:
Hello.
I have been working on getting my asterisk box to connect to a lucent
definity PBX using a T100p. I connected it to a t1 port on the lucent
Let me start by saying I have not worked on a lucent definity. Having said
that, I'll tell you my thoughts, and maybe they're
Hi All,
I have been busy with this problem for a while now, but I can't find any
solution. First I thought this was a problem with the phones, but all my
phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried
all firmware versions I could find for the phones.
First, my
Asterisk is still saying it accepts G729. That is prolly the problem.
Try updating to the latest CVS of Asterisk or to Asterisk version 0.7.1.
If there any reason you are allowing both ulaw AND alaw.
On Tue, 2004-02-03 at 08:48, Geert Nijpels wrote:
Hi All,
I have been busy with this
I have a Dial Statement and at the end ,m,A(transfer) but when the
extension picks up it doesn't play anything
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of reseaux
Sent: Tuesday, February 03, 2004 7:14 AM
To: [EMAIL PROTECTED]
Subject: Re:
I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank T1 card suggestions,
please. I've also just completed an eval of the Mediatrix 1204 which
does not
Eric Wieling wrote:
Asterisk is still saying it accepts G729. That is prolly the problem.
Try updating to the latest CVS of Asterisk or to Asterisk version 0.7.1.
If there any reason you are allowing both ulaw AND alaw.
Sorry forgot to mention it. I'm already at latest CVS, but I have this
When I use option t and m together in the same dial
statement the music on hold doesnt appear to work.
Is this a normal operation?
This strikes me as something that should be really very simple to do, but I
can't figure it out.
Is there a way of logging all SIP debuging info to a file somewhere?
It would help me greatly!
Cheers,
Steve
--
Steve Foy| http://www.unite.net
UNITE Solutions | Tel: 028 9077 7338
On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
Hi,
I'm trying to get my X100P to Dial the following sequence to gain access to
speed dial numbers on my Norstar PBX that the X100 is connected to...
[FLASH] [*] [0] [22] (where 22 is the speed dial number)
But so far I've had no luck,
Steve Foy wrote:
This strikes me as something that should be really very simple to do, but I
can't figure it out.
Is there a way of logging all SIP debuging info to a file somewhere?
It would help me greatly!
I dont know if it's possible using asterisk. You can use the command
'script -a
Or you could modify the logger and have all SIP messages set at a different
log level and have them go to a file (/var/log/messages/sip) for example.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Geert Nijpels
Sent: Tuesday, February 03, 2004 11:38
Hello,
I use 2.4.18-14 and as soon as I did CVS after Jan 10th, 2004
everything went wrong in terms of compiling zaptel.
No matter what I get compiling errors related to different header files
from linux kernel source tree.
Which kernel version you guys used when you tested the latest zaptel
When I use option t and m together in the same dial statement the
music on hold doesn't appear to work.
Is this a normal operation?
1) Please don't post with HTML. Read the archives for several
lengthy flamewars over this topic. Comments as to how I suck because
I don't like HTML will be
Hi.
any hint?
I've never played with the 1104, however others have reported that it
does register correctly when properly configured (and with * properly
matching).
In order for anyone to offer any suggestions, however, you'll have to
pass along the config info for both * and the 1104.
I upgraded to 0.7.1 from a cvs version from a few weeks before 0.7.1 was
relesed.
now I am having troubles with my dialing plan and voice mail.
As part of the upgrade I re-built the machine so there was a blank slate
however after installing 0.7.1 I had no mail box creation script and
could
Steven Critchfield wrote:
On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
Hi,
I'm trying to get my X100P to Dial the following sequence to gain
access to speed dial numbers on my Norstar PBX that the X100 is
connected to...
[snip]
The problem with your example is that a flash must be
On Tuesday 03 February 2004 01:44, Tomica Crnek wrote:
[global]
hostname=localhost
dbname=asteriskcdrdb
password=**
user=asteriskcdruser
;port=3306
;sock=/tmp/mysql.sock
sock=/var/lib/mysql/mysql.sock
Okay, and so does this work?
bash$ echo select max(calldate) from cdr; | \
mysql
Title: RE: [Asterisk-Users] Still looking for small fxo sip gateway
You might want to take a look on the Wiki pages for VoIP, in particular:
http://www.voip-info.org/wiki-VoIP+Gateways
Offhand at our site we're trying to set up something similar (although a little larger, 10 FXO lines,
Neo: What are you trying to tell me? That I can dodge bullets?
Morpheus: No, Neo. I'm trying to tell you that when you're ready,
you won't have to.
There have been over 500 downloads of the RedHat Asterisk RPMS
since they were released 2 weeks ago, and I have received many comments
to
This strikes me as something that should be really very simple to do, but I
can't figure it out.
Is there a way of logging all SIP debuging info to a file somewhere?
It would help me greatly!
Might take a look at /etc/asterisk/logger.conf file to see if that's what
you're looking for.
Is there a way to make the Cisco 7960 SIP phone dial out automatically
without having to press the dial button, once the numbers that you have
entered match a specific pattern? This feature is present when the phone
is working with a Cisco CallManager. For example, if all of my internal
Title: RE: [Asterisk-Users] Still looking for small fxo sip gateway
The wonder is none of the FXO devices works fine except
asterisk X100P.
I'm not sure what is the stupidity present in that analog
technology.
Kannaiyan
- Original Message -
From:
Kostur,
Andre
To:
I have created a pdf document about my experience in integrating a Nortel
Norstar MICS with *. This is not a cookbook, but it does describe the
process I followed and gave a copy of the relevant configuration files.
If anybody is interested, please feel free to download a copy at
Take a look at dialplan.xml on your tftp server.
DIALTEMPLATE
TEMPLATE MATCH=0 Timeout=1 User=IP/ !-- Local operator--
TEMPLATE MATCH=8,011* Timeout=6 User=IP/ !-- International
calls--
TEMPLATE MATCH=8,1.. Timeout=0 User=IP/ !-- Long Distance --
I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank T1 card suggestions,
please. I've also just completed an eval of the Mediatrix 1204 which
does not
Hello all,
Saturday night, after a couple of shots of bourbon, I realized
that I had an old PC sitting in the garage that I could use as an Asterisk
gateway if I just blew the dust off it and reloaded it with a modern Linux
distribution. In my characteristically impulsive manner, I
On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
Steven Critchfield wrote:
On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
Hi,
I'm trying to get my X100P to Dial the following sequence to gain
access to speed dial numbers on my Norstar PBX that the X100 is
connected to...
Is there a way to make the Cisco 7960 SIP phone dial out automatically
without having to press the dial button, once the numbers that you have
entered match a specific pattern? This feature is present when the phone
is working with a Cisco CallManager. For example, if all of my internal
Steve == Steve Foy [EMAIL PROTECTED] writes:
Steve Is there a way of logging all SIP debuging info to a file
Steve somewhere?
Use tethereal or tcpdump to log sip (and/or rtp/rtcp) packets to a
pcap file, then use ethereal (presumably on a different box) to view
them.
-JimC
Greg Boehnlein wrote:
Is this the smallest Asterisk server ever? :)
WHY??? just kidding. That's pretty cool. Maybe if you kicked it up to 64
MB, you could create a 4-port sip fxo device and stop all of these posts
about not being able to find one...
This could be good news for the embedded
Hi David,
I've been working with an * setup with a VoiceTronix 4-port FXO card and
I had similar problems with detecting dialtone. I had to chan_vpb.c to
use a VPB driver call progress dial command, and to change the dialtone
dectection (I'm in North America) -- please see patch against version
I tried with these ones but nothing change much:
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+CAPI+readmediff=2
My extensions are
[outgoing]
exten = 0,1,Goto(outgoing-isdn,s,1)
[outgoing-isdn]
exten = s,1,NoOp()
exten = _X.,1,Dial(CAPI/mynumber:b${EXTEN}|30)
Any ideas?
If asterisk'll compile against uclibc, it'll go on the toaster. Most
toasters (and coffee grinders such) don't have enough flash memory for
a full glibc...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Gomillion
Sent: Tuesday, February 03, 2004
Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox.
I was thinking of using that as an IAX-sip translator for offices with NAT.
CPU MPC855T (PowerPC Dual-CPU)
Memory 32MB RAM / 4MB Flash (TS100)
Interfaces1 Ethernet 10/100BT on RJ45
1 RS232 Console on RJ45
RS232 Serial
On Tuesday 03 February 2004 13:10, Swen Veckes wrote:
In addition it doesn't
seem you can
disable (at least on my sip fw release) the
redirect-to-vm-on-busy feature.
Actually I think of a soultion like changeing the VM app to be canceld when
dialing
* and one can enter the vm and pin
Now, here's the real question: can you install it on a toaster?
It builds and runs on NetBSD, minus the hardware part (for the
moment)...so yeah.
Asterisk on NetBSD/Vax. Hrm.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Tuesday 03 February 2004 08:57, Matthew B Marlowe wrote:
I have a Dial Statement and at the end ,m,A(transfer) but when the
extension picks up it doesn't play anything
Well, that would be why it doesn't work. Please recheck the help
document. You will find that you cannot separate options
Smallest Asterisk server? No. That old Gateway box must
be about 2 cubic feet. 1.5 ft^3 at a minimum. I've got one
that is about 0.2 ft^3 a factor of maybe 10 smaller.
I've installed a working Asterisk server on an older Toshiba
notebok PC. The Notebook has a 144Mhz Pentium, 80MB RAM
and a
Steven Critchfield wrote:
On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
Steven Critchfield wrote:
On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
Hi,
I'm trying to get my X100P to Dial the following sequence to gain
access to speed dial numbers on my Norstar PBX that the X100 is
I'm still getting a sementation fault with mpg123. I have tried different
parameters creating mp3s the last from cd audio ...
lame -m s --resample 8000 -q 0 -a --cbr -b 32
and several versions of mpg123. I have always created 8000 hz outputs. I've
got other * boxes that don't use moh that have
This thread got me thinking of other servers that would run asterisk. The
obvious question comes up if Xebian (the xbox version of Debian) would run
as a SIP only server? Asterisk on an XBox would be a small box! Cheap too.
Tom Schaefer
___
Hi All,
We are compiling an Asterisk interoperability list.
If you have connected Asterisk to either a PBX or another voice/Voip
device (gateway, gatekeeper, etc ...) please drop me an email. I will
compile it and make it available to the list and on the wiki.
Please make sure to send
On Tue, 2004-02-03 at 11:33, David Gomillion wrote:
Steven Critchfield wrote:
On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
Steven Critchfield wrote:
On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
Hi,
I'm trying to get my X100P to Dial the following sequence to gain
I read a report of Asterisk running on a Microsoft X-Box.
That's kind of a stunt as you could buy a decent PC for
the price of a Linux-capable XBox. Id's still like to
see Asterisk run on very low-end hardware
The Snom IP phone runs Linux inside? I assume as Linux
is GPL'd Snom will supply the
When using a dial statement of:
exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,m)
The call is placed with the music on hold and works fine but when I add
exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,t,m)
The music on hold will not work
If I use a statement of
When troubleshooting, I'll often
tcpdump -s 0 -w filename.cap -p host (ipaddressofphone)
To capture the entire contents of all packets from or to
ipaddressofphone non-promiscuously to filename.cap. Since
my workstation is Win*, I have to sz to move the capture
over to my desktop and then open
On Tue, 2004-02-03 at 12:01, Chris Albertson wrote:
I read a report of Asterisk running on a Microsoft X-Box.
That's kind of a stunt as you could buy a decent PC for
the price of a Linux-capable XBox. Id's still like to
see Asterisk run on very low-end hardware
The Snom IP phone runs Linux
Hi, all
When I use x100p card, my DSL modem can not connect
with ISP. Is my card bad or all x100p conflict with
DSL modem?
Best,
Michael
__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
--- [EMAIL PROTECTED] wrote:
This thread got me thinking of other servers that would run asterisk.
The
obvious question comes up if Xebian (the xbox version of Debian)
would run
as a SIP only server? Asterisk on an XBox would be a small box! Cheap
too.
It's been done. In fact by Mark hiom
Snom Does gives the souce and more:
http://www.snom.com/sources_en.php
- Original Message -
From: Chris Albertson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 4:01 PM
Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
I read a report of
On Tue, Feb 03, 2004 at 01:04:27PM -0500, Matthew B Marlowe wrote:
exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,t,m)
The music on hold will not work
I believe you do not want a comma between the t and the m.
-w
--
/~\ The ASCII Ribbon Campaign
\ /No HTML/RTF in email
X
You do not put a , between t,m or any of the end parameters.
See show application dial
On Tue, 2004-02-03 at 12:04, Matthew B Marlowe wrote:
When using a dial statement of:
exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,m)
The call is placed with the music on hold and works
On Tue, Feb 03, 2004 at 10:46:50AM -0700, [EMAIL PROTECTED] wrote:
This thread got me thinking of other servers that would run asterisk. The
obvious question comes up if Xebian (the xbox version of Debian) would run
as a SIP only server? Asterisk on an XBox would be a small box! Cheap too.
I
The Snom IP phone runs Linux inside? I assume as Linux
is GPL'd Snom will supply the source code? It would be
fun to install an Asterisk server in a phone.
Similarly, I know there was a stink about Linksys using linux
inside a router. I just picked up a USR 802.11g router that
would
Well I also though about this five minutes ago... I think the biggest
problem should be memory (we have 16 MB DRAM and 4 MB Flash).
Also, the question is if the plastic makes a box impression...
Christian
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
| I'm still getting a sementation fault with mpg123.
Isn't it time to get mg3 out of the equation?
Sox can convert just about anything to 16 bit signed mono pcm in
just about any container that support that. It looks like *'s
format_wav.c is for exactly that format, so for local files we
On Tue, 2004-02-03 at 19:14, Michael Zheng wrote:
Hi, all
When I use x100p card, my DSL modem can not connect
with ISP. Is my card bad or all x100p conflict with
DSL modem?
I have an X100p working in the same box as a Bewan PCI ADSL modem with
no problems. But adding a radio card caused no
Hi,
-Original Message-
This thread got me thinking of other servers that would run
asterisk. The obvious question comes up if Xebian (the xbox
version of Debian) would run as a SIP only server? Asterisk
on an XBox would be a small box! Cheap too.
The Xbox has USB ports, right ?
I'd love to have a non-mp3 music-on-hold option. Anybody put this as a
feature request yet?
MATT---
-Original Message-
From: James H. Cloos Jr. [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 1:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: sementation fault with
On Tuesday 03 February 2004 12:14, Michael Zheng wrote:
When I use x100p card, my DSL modem can not connect
with ISP. Is my card bad or all x100p conflict with
DSL modem?
Perhaps you forgot to put a filter between the line and your X100P?
-Tilghman
On Tuesday 03 February 2004 11:46, john wrote:
I'm still getting a sementation fault with mpg123. I have tried
Ah, adventures in the pubic school system.
This GDB was configured as i386-redhat-linux-gnu.
Core was generated by `asterisk -vvvfg'.
Program terminated with signal 11, Segmentation
Do you have a DSL filter on your X100P? Just like any other telephone
device it needs a DSL filter to keep it from messing up your DSL
service.
On Tue, 2004-02-03 at 12:14, Michael Zheng wrote:
Hi, all
When I use x100p card, my DSL modem can not connect
with ISP. Is my card bad or all
linphone is available for ipaqs running familiar/opie linux.
It's on my todo list to try it out via wifi
William Waites ([EMAIL PROTECTED]) wrote:
On Tue, Feb 03, 2004 at 10:46:50AM -0700, [EMAIL PROTECTED] wrote:
This thread got me thinking of other servers that would run asterisk. The
Comments below.
Rich Adamson wrote:
I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank T1 card suggestions,
please. I've also just completed an eval of
Does anyone know, is there a way to get current status of device
From * using some variable or similar in relation to qualify=XXX
statement.
I am referring to qualify= which qualifies and monitors if device
is reachable.
I need this in order to include it in my dial plan so that incoming
On Tue, 3 Feb 2004, Chris Albertson wrote:
Smallest Asterisk server? No. That old Gateway box must
be about 2 cubic feet. 1.5 ft^3 at a minimum. I've got one
that is about 0.2 ft^3 a factor of maybe 10 smaller.
Hehehe.. As far as Form Factor goes, I'm sure there are smaller boxes
out
On Tue, 3 Feb 2004, Steven Critchfield wrote:
On Tue, 2004-02-03 at 12:01, Chris Albertson wrote:
I read a report of Asterisk running on a Microsoft X-Box.
That's kind of a stunt as you could buy a decent PC for
the price of a Linux-capable XBox. Id's still like to
see Asterisk run on
Here is my experience so far to treat some issues I have been having with
Digium hardware (t100p, and x100p's.) I am not 100% certain these are
fixxes, but just something for people to try if they are expierencing
issues with the hardware performing quirky.
1st) Do NOT use Promise Array ATA
Possible Mediatrix 1204 fxo sip gateway workaround
Need some feedback from experienced * users relative to this workaround
please please please.
Problem: The mediatrix 4-port fxo gateway does not provide any mechanism
for * to select which port an outbound pstn call will use. (See lots
of
Check here for list of small Asterisk implementations mentioned on the mailing list.
http://www.voip-info.org/wiki-Asterisk+setup+minimum
Jim
James H. Thompson
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for
that.
The Linux bit is all free, and only a couple of PCB work to disenable the
protection.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Albertson
Sent: 03 February 2004
Yes...myself. I can be contacted at the email above or on (021) 1387245.
Kind regards,
Matt Riddell
Are there any New Zealand Asterisk users/contractors out there - we're
looking to install a small business pnx and are interested in Asterisk
as a solution.
Thanks, I don't know what is different from all steps I have followed
several times. I did all this before, believe me. Now, I said to myself
that I'll do it once again, and it worked.
Thanks once again!
Tomica
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I cant wait to see the asterisk on an xbox page!!, but the link seems broken
http://nlug.org/mail/nlugb2003_12/0094.html
I've tried removing the b with no luck
Anyone know what the link should be ?
Thanks
Panny
- Original Message -
From: David J Carter [EMAIL PROTECTED]
To: [EMAIL
Clif,
I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank T1 card suggestions,
please. I've also just completed an eval of the Mediatrix 1204 which
How about a PCMCIA Zapata interface?? Asterisk and its strength kick
ass as a test unit. Can't do some of the things a T-byrd can do but the
Telco techs freak when you tell them its your PBX!!!
)
10. Re: The Smallest Asterisk Server Ever? (Panny Malialis)
Message: 10
From: Panny Malialis
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