need to understand docs are never enough,
so if you're going for a real degree, only reading
the code will give you degree-level results...
and also if you go deep enough perhaps you'll have
some very useful suggestions for the * community...
matteo.
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phpmyadmin :)
Matteo.
Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha
scritto:
Hello list,
Does anyone know about a web/php interface to deal with users in Realtime's
Mysql database (sipusers and sippeers tables) ?
Thanks in advance
Laurent
Hi,
Il giorno gio, 07-04-2005 alle 05:19 -0400, Asterisk Pbx ha scritto:
I am thinking in implementing asterisk into my buisness. I heard all
sorts of good things about it. The question im asking my self is what
linux distribution is best to use? Do you know what distribution they
use for
Hi,
Digium, the service is problematic. Well, I believe that Digium should
services it's channels, the channels should support the resellers and the
resellers should support
The customers. I don't think that any company, no matter what its size or
function is, could support the end users.
Hi,
Il giorno gio, 07-04-2005 alle 13:02 +0200, [EMAIL PROTECTED] ha scritto:
Digium do sell online and so many other of their resellers do. The important
point is that they don't sell lower cost than their resellers, which is the
case.
Please find an hardware producer that sells directly to
Hi,
Thanks answering me, that works with the *8 (and *02
th e pattern in my company works too) but there is a
problem : how do you select the phone ringing to
pickup ? For example phones 23 and 24 are ringing ;
I'm 25 (same pickupgroup as 23 and 24 callgroup), How
do I decide either
Hi,
ast_data is replaced by ARA (Asterisk RealTime Architecture). That is
why you can't compile ast_data on HEAD.
I'm currently testing ast_data on HEAD. the patches applies
and compiles :)
ARA is something different from ast_data:
* ast_data is real realtime: for example extensions.conf
is
Hi,
ast_data is replaced by ARA (Asterisk RealTime Architecture). That is
why you can't compile ast_data on HEAD.
I'm currently testing ast_data on HEAD. the patches applies
and compiles :)
One note: you need to use the SVN dev version for that.
Matteo
Hi,
It's working fine, although I'm not sure if it comes with asterisk or with
bristuff ...
bristuff
Matteo
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Il giorno mer, 12-01-2005 alle 20:29 +0330, Paradise Dove ha scritto:
just to make sure:
when i have zaptel devices on my box and i also use meetme and iax2,
do i need to have USB device enabled and it's modules loaded?
no, no, no
just usbcore+uhci loaded + ztdummy
Matteo.
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agi + dial option T is your friend.
Matteo.
Il mar, 2005-01-18 alle 15:24, Joao Pereira ha scritto:
Hello to all
Im using SER as SIP registrar and Asterisk as GW and billing system but I m
not sure if Asterisk can interupt calls when a client is out of credit. Is
there any way of doing it or
Hi,
Can someone tell me if the timing device is needed for voicemail and
other applications too?.
i'm sure that searching on google and/or voip-info.org can lead
to an answer.
btw, the answer is no. only meetme and iax truking needs a timing
device.
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app_capiRETRIEVE.o
Any suggestions.. and sorry for a very long posting...
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on port X is enabled after a request
by the device on that port. this is for mantaining compatibity
with non POE devices.
so you will need also something that detects the power request
on each port and enables it.
Matteo.
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Hi.
I'm not using snom phones for a while, but
now I want to test again them and I'm gonna
buy a snom 200 105 .
Some times ago I had a snom 100 , and gsm wasn't
working with *. How's now the situation?
the snom gsm works well with * ?
Thanks for any info, Matteo.
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Hi.
About a month ago I made a test with snom200b.
At least then it worked ok with *.
At the moment I'm using mainly g711a. So, there is always a possibility
something
but you also tested gsm ?
Greets,Matteo.
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Web
to the called party, using x as
file
see last param ...
Matteo.
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auth required
* the phone answers back with the logon data.
now the phone is registered.
the mediatrix stops at step2: never
answers to asterisk with logon data after a proxy auth required
any hint?
Matteo.
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* capi.conf was wrong
Matteo.
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.
reverting that, made it register without issues.
Matteo.
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without switches, but making asterisk browse
between multiple sw can be useful)
matteo.
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hi
I have set up a * box supporting 3 different companies but have some
questions regarding MOH. Can MOH support multiple context or classes.
Reason I ask each company would like to have different MOH sound files.
Is this possible?
yes, just specify multiple moh classes in musiconhold.conf
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, and the difference is
remarkable...
since is zaptel based, it shares same zaptel routines for EC,
as far as I know.
Matteo.
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...
I get always CRC error, and can't load the firmware
into the phone (via tftp o web interface)
Any hint?
Matteo
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: lunedì 3 marzo 2003 18.09
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users] Re: [Asterisk] phones being autoanswered?
Matteo Brancaleoni wrote:
Hi.
I'm experiencing a strange issue with *.
I have a dev kit, aka a T100P + a zhone cb.
Sometimes, on certains phones (on the fxo ports
the license is needed only with cisco
callmanager. so you can ignore it and use
both ports with asterisk ;-)
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
Michiel Betel
Inviato: giovedì 6 marzo 2003 10.43
A: [EMAIL PROTECTED]
Oggetto:
is the snom
and hangsup on the snom side are ok and hangsup on the
gate are not.
P.S. The gate works ok with any other channel (zap, for ex.)
Matteo Brancaleoni
[EMAIL PROTECTED]
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EspiA - EMMEGI Srl - e*solution provider
Uffici: Via Pascoli, 37
20129 Milano - Italy
Sede
,
with a repository of all * patches. tell me
if anyone is interested into that)
Matteo Brancaleoni
[EMAIL PROTECTED]
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EspiA - EMMEGI Srl - e*solution provider
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Sede Legale: Corso Sempione 67
20149 Milano - Italy
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Fax. +39
Hi, I have a mixed sip-zap evironment
in my office. I was wondering if is
possible to do remote call pickup
from a sip phone, like from zap.
Any hint?
Matteo Brancaleoni
[EMAIL PROTECTED]
Emmegi System Administrator
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20129 Milano
use the System app,
like
exten = s,1,System(/path/to/your/application)
matteo.
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
Christoph Schütz
Inviato: giovedì 13 marzo 2003 22.46
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users] Execute shell
Here in italy we have 3 running like a charm!
matteo.
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
Lars Fredriksson
Inviato: mercoledì 19 marzo 2003 10.57
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users] X100P in Sweden / Europe
Hello
the URL as much as you like.
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with rfc2833 dtmfmode? They work as espected with inband dtmf.
lele
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TDM40B by Digium ;
look @ their website.
matteo
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
David Luyens
Inviato: lunedì 24 marzo 2003 16.46
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users] 4 port FXS card
Does anybody know of a 4 port
=5060
From: sip:[EMAIL PROTECTED];user=phone;tag=0775a502
To: snom100 sip:[EMAIL PROTECTED];tag=p8r4rh552w
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: snom100-1.15e
Contact: sip:[EMAIL PROTECTED]:5060;line=1
Content-Length: 0
9 headers, 0 lines
Message is BYE
astro*CLI
Any idea??
Matteo
://lists.digium.com/mailman/listinfo/asterisk-users
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?
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Hi.
I've found a problem when I pickup a remote sip phone with *8.
There're both budgetones 102 and are both in the same group.
When one sip phone is ringing, I can pickup the call from
another sip phone, but the first one keeps playing a loud
busy signal... that don't go away until I receive
Hi.
Today I was asked about a function of asterisk.
That's what it should to:
a call arrives - put it in a queue - remain here ;)
Then, when someone wants to answer, just dial an extension
and the older call that's in the queue is picked up.
A sort of app_queueonly + app_pickolderqueuecall .
As
There's anyone that's using chan_local that would
provide to the community some example on how to
use it or what's needed for ?
Just for knowledge.
Matteo.
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mmh, I think I found the problem.
seems that my agi script that I use to lookup channels
on a db doesn't close the pipe properly.
So fd descriptors starts to pile up until the process
max files is reached...
My script is written in php-4.3.2 (cli). Is very simple,
and I open close sockets
You can do it with some Agi scripting and call spooling.
Look in app_agi app_qcall.
Matteo
Il ven, 2003-06-27 alle 08:52, Michiel Betel ha scritto:
Some switches provide the functionality to try a number till it
becomes available. Thus when one dials a number and get a busy, one
enters a
the exten is *8 only.
didn't know if 0.4.0 has already that features on sip .
It's better to use latest cvs version.
Matteo.
Il mar, 2003-07-01 alle 11:52, Louis-David Mitterrand ha scritto:
Hello,
We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186
phones.
All sip
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show application monitor in the cli
Matteo.
Il mer, 2003-07-02 alle 09:22, Herv Thibaud ha scritto:
hi
is there a simple way to record a conversation with asterisk ?
thanks
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If the ML appreciate that, we can ask Mark to add it to cvs
But before that, I would appreciate someone else testing...
(in my environment works ok ...)
Matteo.
Il ven, 2003-07-04 alle 09:53, Florian Overkamp ha scritto:
At 07:57 4-7-2003 +0200, you wrote:
It's an offence ;) ?
yes I know
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experienced with the new firmware .77 ?
There's Day Light Saving time now, but haven't
time to play with it, till now.
Matteo.
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or the budgetones?
I haven't found any doc on the net that reports what values to use...
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or tel:17005662458 - ext 911
) is THE first and original
H.323 channel driver for Asterisk. The other is
based on this one.
Regards.
Michael.
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*,# or even A,B,C,D (detected as one '1')
Basically is the same thing that happens into rtp.c
Works great for me.
Patch against latest cvs.
Yup! This does the trick. * and # work now.
I can't test A through D, but, well, so life goes. -rt
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Scrive Paul Cheng [EMAIL PROTECTED]:
What baffles me is why SIP INFO doesn't work with Asterisk AppVoicemail.
I'm using DTMF = info with my budgetones and they works great,
voicemail is ok, dtmf on remote ivr are ok too.
what * version are you using?
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Hi.
Johnson-Perkins, Robert wrote:
If you are just doing VoIP (i.e. no FXO/FXS Cards involved) you should
be able to run up multiple virtual copies of Linux * in VMWare or
Virtual PC.
Though I guess you would need a pretty pokey machine
User Mode Linux is way better for that use, much more
module, the card
works without problems at all on the
remaining 3 modules.
using latest zaptel cvs.
anyone is experiencing that or have
a workaround ?
thanks a lot,
Matteo
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, drop me an email off line,
so will be able to meet at the expo and chat a bit ;)
Matteo.
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and pretending to be you if they should start leaving ring-back
numbers.
JT
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be enabled).
feel free to use it ;)
P.S.
the mirror is updated daily, at 05.00 Rome Time (italy).
Digium staff: hope that isn't wrong for you ;)
Matteo.
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Like what PEARS (php libs) do for db backends?
Matteo.
Garry Adkins wrote:
I am trying a scenerio where the * will take the email and mailbox
number from the Mysql database for sendming mail to a voicemail user. I
have seen vmdb.sql file but is not able to determine its use.
You can't
the above patches are included.
also other settings can be added into mysql... but I see only
the auth part as very useful...
Matteo.
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Hi,
Il giorno lun, 22-11-2004 alle 08:49 -0500, Jason p ha scritto:
Fatal error: Unknown function: mssql_get_last_message() in
/var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on
also here...perhaps they're switching away from mssql ? :)
Matteo
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Ciao,
l gio, 2004-12-02 alle 14:10, Leonardo Tramontina ha scritto:
snip
-- Registered channel 23, PRI Signalling signalling
Dec 2 11:07:15 ERROR[5209]: chan_zap.c:6447 mkintf: Channel 24 is reserved
for D-channel.
Dec 2 11:07:15 ERROR[5209]: chan_zap.c:9274 setup_zap: Unable to
!
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dial the number directly...
this is called DID , aka direct inward dialing... sure
it works with asterisk
Matteo.
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, providing infos on how to enable it.
In this way you don't have to worry about user complaints about
privacy (hey, you've turned on! isn't that by default),
Also not all systems could have a open internet
connection... so sending infos is impossible at all.
Matteo.
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Hi.
another (stupid) thing.
don't call that function motv. motv is a name
for another opensource project.
Matteo.
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.
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:
if I use g729 (and sip info dtmf) for sip phones - asterisk communication
will asterisk be able to regenerate real tones when going out to the
pstn ?
Tnx for any help ... currently I havent got g729 licenses so I cant
test it out by myself.
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Hi.
What versions does everyone use without problems.
0.59r is PERFECT
0.59r here. all ok.
matteo.
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off
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.
perhaps voicemail2 exists only as an alias to voicemail
to make the transition smoother
Matteo.
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CPU 2.53GHz
SMP Motherboard.
512 MB Ram
Linux-2.4.26 (Compiled as SMP)
Latest Asterisk Devel CVS version
mpg123 Version 0.59s-mh4
Hisax card: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 2).
Wildcard TDM400P
Wildcard X100P
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Matteo Brancaleoni [EMAIL PROTECTED]
Espia
fxoks = 1-31
loadzone = us
defaultzone = us
zapata.conf is simply
transfer=yes
echocancel=yes
threewaycalling=yes
signalling=fxo_ks
context=interni
channel=1-31
any hint on where I can search for problems?
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Matteo Brancaleoni [EMAIL PROTECTED]
Espia - Emmegi Srl
for guest is like a big it expo without
internet connections...)
I have some bandwidth here, so can
set up quickly a server for .it conference
termination...
so, bring on and demostrate to the world
what asterisk can do!
Matteo :)
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Matteo Brancaleoni
System
/me too
this morning was all okie, now I can't connect.
I have an asterisk server ready for replicate the conference
here in .it, as soon as the link will be up with someone,
I'll post the IAX2 url
Matteo.
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Matteo Brancaleoni
System Administrator
.
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Matteo Brancaleoni
System Administrator
[EMAIL PROTECTED]
EspiA Srl - e*solution provider
Via Pascoli, 37
20129 Milano - Italy
SIP:[EMAIL PROTECTED]
Tel. +39 0270633354
Fax. +39 0245487890
IAXTEL: 17005662458
http://www.espia.it
.
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Matteo Brancaleoni
System Administrator
[EMAIL PROTECTED]
EspiA Srl - e*solution provider
Via Pascoli, 37
20129 Milano - Italy
SIP:[EMAIL PROTECTED]
Tel. +39 0270633354
Fax. +39 0245487890
IAXTEL: 17005662458
http://www.espia.it
Hi,
Sometimes I see in a context NoOp
What is the purpose of NoOp (no operation) if it does nothing?
Exactly that. Doing nothing :)
btw, noop could be a placeholder for future instructions,
or if you need to delete an application from the dialplan,
saves you from renumbering the priorities.
Or
Hi,
Il ven, 2004-11-12 alle 07:16, Jeremy McNamara ha scritto:
Brian West wrote:
So all you Sysmaster owners run strings on the 'voipgw' binary that runs on
those boxes and you'll see that its asterisk. If you have doubts I'll post
more proof.
snip
...
I too demand sysmaster either pay
Hi,
If the facts are really obvious that they are hiding Asterisk in their
system best would be to make it public I think.
That means informing the well-known online magazines and see what
happening ;-)
what's about slashdot?
Matteo
--
Matteo
it until
that's usefult for someone... since is a very special
features that probably no one will ever use
lemme know.
Matteo.
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Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 911
IAX(2): [EMAIL PROTECTED
this is defined? I looked through it but could not find
where.
If anybody else can find it please let me know.
Regards,
Andres
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Matteo Brancaleoni
to be due to lost packets? I guess I'm asking if its worth getting a
X100P for timing?
-Nathan
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This message was sent using IMP, the Internet Messaging Program
it rings
through and then talk...
Any thoughts on how to do this?
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Web
/listinfo/asterisk-users
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Espia System Administrator
Email : [EMAIL PROTECTED]
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458 - ext 201
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Asterisk
for experienced users (alot), perhaps newbies will not wait
for that post, as soon as they're subscribed and so on.
I vote yes on B) and why not using a footer like
mplayer-users ML, where it says something like
RTFM! link to some docs here :)
Matteo
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)) ?
Bart
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or group of
employees cover while the receptionist is on break.
TL
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Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458 - ext 201
Hi,
Anyone has experience using * through
128 kbs (or bigger) satelite link?
One word of caution: you may have latency problems. Even at the speed
of light, the information has a LONG way to travel...
yup, normally latencies over a satellite link is
about 200/500 msec ... so you will notice
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