No problem Louis...Even though in recent times I've been kind of a jerk
about people not reading the documentation, I've been trying to return
to my original personality on this list, a helpful member of the
community. :-[
On 3/29/2011 12:47 AM, Louis Carreiro wrote:
Wow... completely missed
design your dial-plan for routing a specific number to different context ,
you can try func_odbc for query to DB if you have a large number of setup.
ideally its called click to call but you are made it as, miss call and you
will get a call.
regards
dhaval
On Mon, Mar 28, 2011 at 5:21 PM, Roger
Dear All
I am using Asterisk 1.4.17 in a calling card application. Following
description explains the usage:
A call/request hits asterisk from an ip xxx.xxx.xxx.xxx which opens a
channel for this ip (Lets call it Channel A). Asterisk answers the call and
play IVRs first asking the PIN and then
On Tue, 29 Mar 2011 07:48:08 +0200, magnu...@inputinterior.se wrote:
I was a little unclear, it is not the cell phone that does the call-back, it
is the cell-phone-network.
Makes more sense :-) Thank you.
--
_
-- Bandwidth and
Hi All,
I am having some issues with Asterisk 1.8.3 extensions with a SIP Phone and
an gateway.
My setup is that I have my SIP Phone setup to register with the gateway.
Then the gateway should sent calls to the Asterisk as a type of friend.
This works fine if the SIP Phone configuration
On Tue, Mar 29, 2011 at 12:23 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
design your dial-plan for routing a specific number to different context ,
you can try func_odbc for query to DB if you have a large number of setup.
ideally its called click to call but you are made it as, miss
- upgrade policy - is it intended that someone who has Debian 6 with
the existing Asterisk 1.6 packages (from Debian's maintainer) can just
upgrade to the Digium package without moving or changing any config?
There is nothing specific about the packages that is going to make this
On Mon, 28 Mar 2011 08:20:23 -0400, vip killa vipki...@gmail.com
wrote:
Is anyone using asterisk with fail2ban?
Sorry for hi-jacking the thread, but I was wondering if there were a
lighter alternative that I could run on appliances?
Python uses too much RAM, but I need to find a way to ban
On Mon, 28 Mar 2011 08:20:23 -0400, vip killa vipki...@gmail.com
wrote:
Is anyone using asterisk with fail2ban?
Sorry for hi-jacking the thread, but I was wondering if there were a
lighter alternative that I could run on appliances?
Python uses too much RAM, but I need to find a way to
On Tue, Mar 29, 2011 at 01:59:54PM +0200, Daniel Pocock wrote:
- upgrade policy - is it intended that someone who has Debian 6 with
the existing Asterisk 1.6 packages (from Debian's maintainer) can just
upgrade to the Digium package without moving or changing any config?
There is
Hello list,
I want to get the phone number out of the following P-Asserted-Identity
header :
/BlaBlaBla sip://88779922//@192.168.8.10;user=phone/
I do the following in the dialplan :
/exten = _XXX.,n,Set(PY=${SIP_HEADER(P-Asserted-Identity)})
exten = _XXX.,n,Set(PY2=${CUT(PY,@,1)})/
This
On Tue, 29 Mar 2011 07:31:18 -0500 (CDT), Joe Greco
jgr...@ns.sol.net wrote:
sshguard is *extremely* lightweight compared to most things; it's a very
efficient compiled C application that doesn't have (m?)any dependencies.
Thanks much for the tip. I'll study how to install/configure iptable
and
I have an Asterisk server running 1.6.2.13, where I can't seem to get
the increased logging to save to the /var/log/asterisk/messages file.
I have tried using the standard core set debug 10 and core set
verbose 10, as well as specifically pointing it to the filename with
core set debug 10
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Hoover
Sent: Tuesday, March 29, 2011 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Debugging not going to log
On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover kb7...@gmail.com wrote:
I have an Asterisk server running 1.6.2.13, where I can't seem to get
the increased logging to save to the /var/log/asterisk/messages file.
I have tried using the standard core set debug 10 and core set
verbose 10, as well
That did it. Thanks everyone!
On Tue, Mar 29, 2011 at 9:11 AM, Andrew Latham lath...@gmail.com wrote:
On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover kb7...@gmail.com wrote:
I have an Asterisk server running 1.6.2.13, where I can't seem to get
the increased logging to save to the
Sorry, for some reason I misread it as the forward feature.
On Sun, Mar 27, 2011 at 9:16 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
From the polycom pdf:
divert.fwd.x.enabled
If set to 1, the user will be able to enable universal call
forwarding through the soft key menu.
This
Look at page 311 in that manual
If you disable the soft keys and then reassign the hard key it should
- at least in theory - be possible to accomplish.
On Tue, Mar 29, 2011 at 11:46 AM, C F shma...@gmail.com wrote:
Sorry, for some reason I misread it as the forward feature.
On Sun, Mar 27,
t38 from the itsp to asterisk using FFA = works
t38 from the spa8000 to the local asterisk using FFA = works (no
differance if FAX Passthru Method is set to reinvite or nse)
g711 to a remote system = works(i just turned off t38 on the
spa8000)
just the t38 pt is not working
On Tue, Mar 29, 2011 at 8:19 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello list,
snip
I'm trying this :
*exten = _XXX.,n,Set(PY4=${CUT(PY2,\:,1-)}) *
but this does not change a thing to the string...
Try the following:
*exten = _XXX.,n,Set(PY4=${CUT(PY2,:,2)}) *
--
On 3/29/2011 7:16 AM, Gilles wrote:
On Mon, 28 Mar 2011 08:20:23 -0400, vip killa vipki...@gmail.com
wrote:
Is anyone using asterisk with fail2ban?
Sorry for hi-jacking the thread, but I was wondering if there were a
lighter alternative that I could run on appliances?
Python uses too much
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table for sip endpoints)
Thanks for the idea, but it's not possible, as the Asterisk must
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table for sip
endpoints)
On Tue, 29 Mar 2011, Gilles wrote:
Thanks for the idea, but it's not possible, as the
Thanks again Tilghman
OK now I am back to the original.
[columns]
;static value = column commented now
;aliascdrvar = columncommented now
These are bogus and should never have been uncommented.
alias start = calldate uncommented now
alias callerid = clid uncommented now
Rest are
On 3/29/2011 12:25 PM, Steve Edwards wrote:
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table for sip
endpoints)
On Tue, 29 Mar 2011, Gilles wrote:
Thanks for
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Remember guys, there's a LOT of IP blocks out there that are almost
definitely not going to be somewhere you expect to receive SIP traffic
from.
I agree. Is there a list I could use to check which blocks have
On Tue, Mar 29, 2011 at 2:34 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On 3/29/2011 12:25 PM, Steve Edwards wrote:
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your
On 3/29/2011 12:42 PM, Gilles wrote:
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Remember guys, there's a LOT of IP blocks out there that are almost
definitely not going to be somewhere you expect to receive SIP traffic
from.
I agree. Is there a
I recently configured a SIP peer which i must specify my fromuser as my
ten digit DID. I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is asterisk.
Is this a bug? Or is there some other config I must make ?
register =
On 3/29/2011 12:52 PM, Jeremy Kister wrote:
I recently configured a SIP peer which i must specify my fromuser as
my ten digit DID. I send calls to this peer, but whenever Asterisk
sends an options message, the fromuser is asterisk.
Is this a bug? Or is there some other config I must make
On Tue, Mar 29, 2011 at 12:32 PM, Eric W. Davenport
ewdavenp...@certin.comwrote:
I restarted Asterisk and I still have all the fields outgoing and incoming
except for the CLID field.
Clid is populated in the CSV Simple file as well as the CSV Custom file.
Please, if you would, copy and paste
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Remember guys, there's a LOT of IP blocks out there that are almost
definitely not going to be somewhere you expect to receive SIP traffic
from.
On Tue, 29 Mar 2011, Gilles wrote:
I agree. Is there a list
On 3/29/2011 1:56 PM, Sherwood McGowan wrote:
[mypeer](peer)
host=10.0.138.226
defaultuser=211941
fromuser=211941
md5secret=023f30a320a5781e8ffd1af9888012af
incominglimit=10
IIRC, you need to define the fromuser in the peer in order for the
qualify checks (options packets) to
On Tue, Mar 29, 2011 at 1:25 PM, Jeremy Kister asterisk...@jeremykister.com
wrote:
On 3/29/2011 1:56 PM, Sherwood McGowan wrote:
[mypeer](peer)
host=10.0.138.226
defaultuser=211941
fromuser=211941
md5secret=023f30a320a5781e8ffd1af9888012af
incominglimit=10
IIRC, you
On 3/29/2011 2:29 PM, Warren Selby wrote:
It looks like you did to me. Is it just OPTIONS packets that are showing
the wrong fromuser field? In other words, when you send call traffic over
this peer, does it properly create the SIP packets? For some reason, I'm
correct - when i actually
Oh, damn, my bad, I've apparently read too many sip.conf entries today
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
--
_
-- Bandwidth and Colocation Provided by
Le 29/03/2011 19:34, Sherwood McGowan a écrit :
On 3/29/2011 12:25 PM, Steve Edwards wrote:
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table for sip
endpoints)
On
Obviously, the other side of the world wants connections to your side, no
matter what side you are on.
:-)
Cary
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Tuesday, March 29,
On Tue, Mar 29, 2011 at 3:57 PM, Cary Fitch ca...@usawide.net wrote:
Obviously, the other side of the world wants connections to your side, no
matter what side you are on.
:-)
Cary
Exactly
--
_
-- Bandwidth and Colocation
On Tue, Mar 29, 2011 at 3:52 PM, Eric W. Davenport
ewdavenp...@certin.comwrote:
Hi Warren,
Thanks for your help,
I think this is what you want
Please don't mail me (or anyone else) off the list directly without being
specifically asked to. The idea is to keep things on the mailing list
On 03-29-2011 19:25, Steve Edwards wrote:
Really? How many callers are you expecting from North Korea, Libya, China,
Iran, etc?
after reviewing last week's log i'd say around 25-28k/min :)
--
_
-- Bandwidth and Colocation
On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote:
On 03-29-2011 19:25, Steve Edwards wrote:
Really? How many callers are you expecting from North Korea, Libya, China,
Iran, etc?
after reviewing last week's log i'd say around 25-28k/min :)
So it looks like I should check out sshguard
Hi All;
I have an E1 card with two ports for ISDN PRI.
Do I need to install DAHDI in addition to LIBPRI?
For placing outside calls (outgoing) via the PRI, then in the extension.exe
file, I will use the Dial function? But how can I determine that I need to use
the PRI channels and not the
Hello;
I need to use Cisco IP Phones with Asterisk and I have some questions to know
how to use it if someone can advise:
1) How I can assign for each button an extension?
2) How I can assign for specific button a feature to be used (like call forward
or call pickup .. etc)?
3) As you know
On 03-29-2011 19:25, Steve Edwards wrote:
Really? How many callers are you expecting from North Korea, Libya,
China, Iran, etc?
On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote:
after reviewing last week's log i'd say around 25-28k/min :)
On Tue, 29 Mar 2011, Gilles wrote:
So it
Hi Gilles,
Just to provide an alternative to sshguard: you could use BFD[1]
(based on bash scripts) and configure it to use iptables to block the
attacker host.
The default configuration is to check the logs at each 3 minutes
(using a crontab entry).
BFD rules for Asterisk could be found here
Sorry for the top post, responding from my phone...
Yes, you'll need both DAHDI and libpri to make an E1 card work with asterisk.
Yes, you'll most likely use the Dial() command inside extensions.conf in order
to dial out. You'll differentiate your PRI channels from your analog channels
in your
The answer to all of your questions are the same - the config file that you
create for your phone.
Thanks,
--Warren Selby, dCAP
On Mar 29, 2011, at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
I need to use Cisco IP Phones with Asterisk and I have some questions to know
how
On Tue, Mar 29, 2011 at 4:03 PM, Warren Selby wcse...@selbytech.com wrote:
That information does indeed look like what I want and it appears to be
setup correctly. I will be building a comparable test system later today
(using all the same software versions as you) and I'll test to see if I
I listened to your email using DriveCarefully and will respond as soon as I can.
Download DriveCarefully for free at www.drivecarefully.com
--
Sent from my BlackBerry®
Senior Support Engineer
US Numbers: 561-886-0664
Nicaragua Mobile: +505.8488.6876
--
Hi,
I'm using IAX2 between our SIP and PSTN servers, both running Asterisk
1.6.2. Users connect to the SIP server and dial; the SIP server
forwards the call to the PSTN server over IAX2, which then dials out
over the connected PRI. Since users need detailed call progress
feedback, the first
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