On Wed, May 02, 2012 at 01:48:04PM -0400, CDR wrote:
I get an error when I execute this code
exten = rejected,n,Hangup($[-1*${Z}])
May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
'-' or '!' or '(' or
On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote:
Hi
can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6
Have you tried 'session-timers=refuse' ?
--
Barry
--
_
-- Bandwidth and
session timers for just those peers that had problems with them,
like one of my ITSPs.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller
Sent: Wednesday, April 18, 2012 10:57 AM
To: Asterisk Users
On Wed, Apr 18, 2012 at 08:52:02PM +0200, Olivier CALVANO wrote:
Le 18 avril 2012 17:56, Barry Miller asterisk-us...@notanet.net a ?crit :
On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote:
Hi
can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6
Have
On Mon, Apr 09, 2012 at 06:21:40PM -0400, sean darcy wrote:
I've cut and pasted from the digium fax admin manual:
exten = send,1,NoOp( SENDING FAX )
exten = send,n,Wait(6)
exten = send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten =
On Sun, Jan 29, 2012 at 09:48:47AM -0500, eherr wrote:
I have fail2ban running on my Asterisk box.
Every so often I receive emails stating that the jails stopped and then
started.
Why does this happen?
Why isn't it just continuously running?
fail2ban is restarted when it switches its log
On Thu, Jan 26, 2012 at 11:49:32PM -0500, asterisk jobs wrote:
Hello everyone,
I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?
Chain fail2ban-ASTERISK (1 references)
num target prot opt source destination
1DROP
manually:
[0010b5c1867c]
setvar = HEREITISAGAIN=0010b5c1867c
regexten = 543
I've coded (but not yet seriously tested) an optional feature that would
basically automate just what you suggested. It'll be interesting to see
how many think this is useful.
On Dec 15, 2011, at 7:03 PM, Barry
On Fri, Dec 16, 2011 at 05:02:11PM +0100, Olle E. Johansson wrote:
16 dec 2011 kl. 02:03 skrev Barry Miller:
So is there a way for the dialplan to determine which device caused SIP to
auto-register an extension?
Not really, unless someone else can come up with something
Hi all,
In sip.conf:
[general]
regcontext = autoreg
[devabc]
regexten = 543
creates exten= 543,1,Noop(devabc) in context autoreg when devabc
registers. But I can't use exten= _5XX,2,Dial(SIP/${EXTEN}) in the
dialplan, because there's no device SIP/543. Now I know I can add a line
like
On Thu, Dec 08, 2011 at 04:47:37PM -0600, Asterisk Security Team wrote:
[...]
Description It is possible to enumerate SIP usernames when the general
and user/peer NAT settings differ in whether to respond to
the port a request is sent from or the
On Thu, Oct 20, 2011 at 04:41:17PM -0500, Danny Nicholas wrote:
... I installed 10.0
and copied my 1.4 configuration files over. With a few tweaks everything
works great except for 1 feature that I specifically went to 10.0 for. When
I do an attended transfer, I still get the receptionists
On Thu, Oct 13, 2011 at 10:13:49AM -0500, Kevin P. Fleming wrote:
On 10/12/2011 07:24 PM, Barry Miller wrote:
Up through 1.8, 'database show' returned results ordered by key. In 10,
the output is unordered (or maybe chronological?). Is this intentional?
It was not intentional, probably
On Thu, Oct 13, 2011 at 02:52:58PM -0500, Terry Wilson wrote:
It was not intentional, probably a side-effect of the switch to SQlite 3
from BDB. Unfortunately, that command was not documented to produce the
database results ordered in any particular order, so this change isn't a
bug,
Up through 1.8, 'database show' returned results ordered by key. In 10,
the output is unordered (or maybe chronological?). Is this intentional?
(I know 'database query' will let me view the AstDB any way I want, but
the output isn't formatted as nicely.)
--
Barry
--
On Fri, Jul 22, 2011 at 10:10:01AM +0200, marek cervenka wrote:
hi,
i'm trying build asterisk rpm
normal compilation is ok but rpm building always fail
centos6/asterisk 1.8.5.0
any ideas?
gcc -o recno/rec_utils.o -c recno/rec_utils.c -MD -MT recno/rec_utils.o
-MF
On Mon, Jul 18, 2011 at 11:02:16AM +0530, mahesh katta wrote:
Sorry boss
Best Regards,
Mahesh, I'm afraid that at some point Ashirwad will become annoyed
that you are including the asterisk-users list on these emails.
--
Barry
--
On Thu, Jul 07, 2011 at 12:55:37PM -0500, Tim Nelson wrote:
- Original Message -
On Thu, 7 Jul 2011, Tim Nelson wrote:
On occasion, I have calls coming into an Asterisk 1.2.x system where
the
${CALLERID(num)} includes '-'. Ex:
123-456-7890
How can I strip the
On Sun, Jun 05, 2011 at 04:18:25PM +, satish patel wrote:
Hey guys!
I have just download latest SVN Revision 322051 and compile and install but
my asterisk -V showing still old version :( is it broken ?
/usr/sbin/asterisk -V
Asterisk SVN-branch-1.8-r321926
asterisk -V shows the
When did Dial() with a custom ring cadence replace the default from
indications.conf for subsequent calls?
indications.conf:
ringcadence = 2000,4000
asterisk -rx dahdi show cadences
r1: 667,1333
extensions.conf:
exten= 201,
On Mon, Mar 21, 2011 at 07:45:37PM +0800, asterisk asterisk wrote:
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
it returns 20110321-034329
but it is exactly 8 hours early than the system time when I type date in
linux terminal
Mon
On Mon, Feb 14, 2011 at 10:39:20PM -0500, Jeff LaCoursiere wrote:
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
Bizarre bug?
Hi. I see that 41 years, 7
On Mon, Jan 03, 2011 at 08:04:48PM -0800, Kirill Katsnelson wrote:
Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped?
They're now part of the wiki.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+MIB+Definitions
On Tue, Nov 16, 2010 at 01:17:08PM -0500, Noah Miller wrote:
Hi All -
I pulled from a working system a TDM400 with one s110 fxs and three
x100 fxos. I put it into a new box and the fxs no longer works. The
fxos work just fine. I thought it was odd, but I chalked it up to a
random chance
On Tue, Nov 16, 2010 at 01:55:32PM -0500, Noah Miller wrote:
Hi. FXS cards use FXO signalling, and vice versa. Think of it this way:
FXS cards want to look like a CO when talking to stations, and FXO cards
want to look like a phone when talking to a CO.
Thanks, Barry. I am aware of
On Sun, Nov 07, 2010 at 07:11:43AM -0700, Steve Murphy wrote:
Hey, I'm going thru logs, and I see some very common and interesting things
that the hackers are looking for.
In a whole bunch of scans, I've noticed that the first guess or two for sip
accounts
is usually a 10-digit number. I'm
On Sun, Oct 31, 2010 at 03:23:52AM +0200, Tzafrir Cohen wrote:
On Sat, Oct 30, 2010 at 01:43:49PM -0600, Joel Maslak wrote:
Is there really any benefit to blocking these, if you use good passwords?
Regardless of any threat from those attacks succeeding, they completely
saturated the uplink
On Tue, Oct 19, 2010 at 09:59:34AM -0400, Leif Madsen wrote:
On 10-10-18 11:01 PM, Barry Miller wrote:
On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote:
On 10-10-18 07:54 PM, Asterisk Development Team wrote:
For a full list of changes in the current release
On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote:
On 10-10-18 07:54 PM, Asterisk Development Team wrote:
For a full list of changes in the current release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4
On Sat, Oct 16, 2010 at 04:12:14PM +0530, RAJNIKANT VANZA wrote:
Hi Friends,
I need to find .gsm file length or duration.
*E.g.*
demo-congrats.gsm
sox demo-congrats.gsm -e stat
Above command is display file length in seconds. like as
Length (seconds): 27.96
I want to .gsm
On Sat, Oct 16, 2010 at 06:46:20PM +0100, Tiago Geada wrote:
r you would have to convert that gsm to another format first like ogg
Why on earth would you have to do that? Did you even try doing what
I suggested?
On 16 October 2010 18:23, Barry Miller asterisk-us...@notanet.net wrote
On Sun, Oct 03, 2010 at 02:19:35PM -0600, Greg Saunders wrote:
Hello all. I was recently the victim of a SIP flood attack. I'm wondering
what is the best method to prevent such things in the future.
In sip.conf:
[general]
alwaysauthreject = yes
The attacking program is probably
On Fri, Sep 24, 2010 at 10:25:01PM -0700, Ira wrote:
At 01:14 PM 9/23/2010, you wrote:
The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
On Wed, Sep 22, 2010 at 09:50:00AM -0700, Steve Edwards wrote:
Still, for scripting and portability, I'd recommend specifying the
decompressor and using the long option form:
tar\
--list\
--[un]gzip\
--file\
On Thu, Sep 16, 2010 at 07:44:23PM +0100, Jon Farmer wrote:
Hi
I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
I do on INVITES is to re-authenticate the user from OpenSER. Then when
the INVITE gets passed to Asterisk I capture the AUTH to a variable in
the dialplan
On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote:
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
I have been having issues with hackers doing registration scanning against
our server. We block their address at the fire wall but since
On Thu, Sep 09, 2010 at 12:25:03PM -0500, Carlos Chavez wrote:
Is there an archive of security advisories for Asterisk? We recently
upgraded a customer from 1.2 to 1.4 and now they are asking for
documentation of all security and bug related fixes. I know the
advisories get published
On Tue, Sep 07, 2010 at 04:02:29AM -0700, Kyle Kienapfel wrote:
On Mon, Sep 6, 2010 at 12:32 PM, Barry Miller
asterisk-us...@notanet.netwrote:
After upgrading my small test system from Debian Etch-Lenny via a
complete reinstall, I find my g729 hostid has changed. Same machine,
same CPU
On Tue, Sep 07, 2010 at 07:43:45PM +0100, Tiago Geada wrote:
Hi,
I don't have any g729 codec license. But by reading Barry's complaint I get
to think that it is really unfair that Digium can't renew his license or
something.
I am a Debian user myself and I understand the need to upgrade
On Mon, Sep 06, 2010 at 03:32:35PM -0400, Barry Miller wrote:
After upgrading my small test system from Debian Etch-Lenny via a
complete reinstall, I find my g729 hostid has changed. Same machine,
same CPU, same NIC! It doesn't seem reasonable that I have to burn
my one no-hassle re
After upgrading my small test system from Debian Etch-Lenny via a
complete reinstall, I find my g729 hostid has changed. Same machine,
same CPU, same NIC! It doesn't seem reasonable that I have to burn
my one no-hassle re-registration for a simple OS upgrade.
The README only says that hostid is
On Tue, Aug 10, 2010 at 11:24:18AM -0700, C. Chad Wallace wrote:
At 1:42 PM on 10 Aug 2010, Gilles wrote:
I just read an article on the tiny Ben NanoNote:
http://en.qi-hardware.com/wiki/Ben_NanoNote
As CPU, it uses a JZ4720 366 MHz MIPS compatible processor from
Ingenic
On Tue, Jul 13, 2010 at 11:30:44AM -0500, Warren Selby wrote:
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH =
On Tue, Jul 13, 2010 at 01:07:34PM -0400, Barry Miller wrote:
Try adding preload = func_strings.so to modules.conf
Ah, sorry. I just saw your earlier response that said you're on 1.4 -
I was remembering that after I migrated from 1.4 - 1.6, I had to preload
func_db.so so that I could use
On Mon, Jul 12, 2010 at 04:41:16PM +0100, Frank Church wrote:
Is there a database of MAC address prefixes used the common VoIP
devices. I see the Linksys Sipura devices state with 00:0E.
See http://standards.ieee.org/regauth/oui/oui.txt
--
Barry
--
On Thu, Jul 01, 2010 at 10:19:08PM -0500, Karl Fife wrote:
Calls that come in on DAHDI FXO ports are routed to [context], extension 's'
INSTEAD, I would like to route specific ports to specific extensions, For
example:
I want DAHDI/1-1 to go to 1234
I want DAHDI/1-2 to go to 2345
I want
Hi Gilles. You appear to be both posting to newsgroup
gmane.comp.telephony.pbx.asterisk.user AND sending the same message
directly to the asterisk-users list. This means that we list subscribers
see two copies of all your messages: one from gmane, one from you. (They
don't show up that way on
On Tue, Jun 15, 2010 at 07:50:36PM +0500, Faisal Hanif wrote:
Hi,
We are using Asterisk 1.6.2 and it is continually failing to resolve Verizon
SRV and sending following message,
WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup
'whsvoip.globalipcom.com'
DNS settings on OS
On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote:
Hi,
I am also wonder that same SRV record is working fine on one machine but not
on 2nd while both have same asterisk version.
It may be some missing OS utilities which asterisk using to resolve SRV?
Could be. To test, does
On Tue, Jun 15, 2010 at 10:40:06PM +0500, Faisal Hanif wrote:
Till now I am not able to find any difference between both machines.
Can you please tell me how I can try to resolve it on OS level using some
utility like dig?
On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote:
Hi,
On Wed, Jun 02, 2010 at 10:26:12AM -0400, khalid touati wrote:
Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =
On Tue, May 11, 2010 at 04:42:30PM -0600, Joseph wrote:
On 05/11/10 17:22, lists-asterisk-us...@yoinks.net wrote:
Joseph wrote:
Is anybody using checkbox.cc to make iax2 calls?
They have recently did some changes and my calls no no longer go through.
They don't have a best service
On Sun, Apr 18, 2010 at 08:21:57PM +0200, Remco Bressers wrote:
On Apr 18, 2010, at 12:40 AM, Barry Miller wrote:
On Sat, Apr 17, 2010 at 11:14:23PM +0200, Remco Bressers wrote:
Dear List,
According to https://issues.asterisk.org/view.php?id=14905 there is a storm
prevention
On Sat, Apr 17, 2010 at 11:14:23PM +0200, Remco Bressers wrote:
Dear List,
According to https://issues.asterisk.org/view.php?id=14905 there is a storm
prevention mechanism in newer Asterisks. If i look in my logfile, i see :
[2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from
On Tue, Apr 13, 2010 at 06:59:01PM -0400, David Backeberg wrote:
On Tue, Apr 13, 2010 at 6:55 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote:
What do you mean with problems on my configuration?
?This is a FXO port on
Using distinctive ring detection with bell202 cid, is there any way to tell
DAHDI to sometimes expect the cid after the 2nd ring, other times after the
1st?
I just added RingMaster service (2nd DID w/ distinctive ring) to a TDM800P
FXO line. No problem setting dringcontext for the 2nd DID.
On Mon, Feb 22, 2010 at 12:57:30PM -0500, Leif Madsen wrote:
Jerry Geis wrote:
I am trying to find out how I can tell the length of a string actually
CALLERID(num) in the dialplan.
How is that done?
If need to test the length of the CALLERID(num) if its less the 10 digits I
need
On Sun, Feb 14, 2010 at 04:50:06AM +0100, Alejandro Recarey wrote:
Much to my surprise I tried to debug an AGI script today with agi
debug on the Asterisk CLI and it did not work. Plus, I could find no
reference on lie of it being removed.
Is there another name for that command? I scanned
Here's a .sig from the m...@openbsd list, (which I couldn't resist
top-posting.)
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
A: Top-posting.
Q: What is the most annoying thing in e-mail?
On Sat, Feb 06, 2010 at 10:13:42AM
I think the quotes cause the values to be compared as strings, not
numbers. The old shell programmer's trick (which allows for empty
strings):
exten = s,n,GotoIf($[0${SPEECH_SCORE(0)} = 0${THRESHOLD}]?:tag)
ought to cause a numeric comparison.
--
Barry
On Thu, Feb 04, 2010 at 09:42:18AM
Interesting. I had the same problem last Sept with a TDM800, DAHDI 2.2.0.2.
Shaun Ruffel of Digium pointed me to
https://issues.asterisk.org/file_download.php?file_id=22725type=bug
which fixed it for me. This fix is already in 2.2.1.
--
Barry
On Thu, Jan 28, 2010 at 07:30:57PM -0600, Karl
On Mon, Jan 11, 2010 at 03:23:33PM +, --[ UxBoD ]-- wrote:
Hi,
why would Asterisk core dump with the following test dialplan extension ?
exten = 8100,1,Answer()
exten = 8100,n,Set(CALLERID(all)=)
exten = 8100,n,PrivacyManager()
exten = 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} =
On Mon, Jan 11, 2010 at 04:42:44PM +, --[ UxBoD ]-- wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
On Mon, Jan 11, 2010 at 03:23:33PM +, --[ UxBoD ]-- wrote:
Hi,
why would Asterisk core dump with the following test dialplan
extension ?
exten = 8100,1
On Fri, Dec 18, 2009 at 12:23:22AM +0100, Olivier wrote:
Today, this IVR is using function AEL GotoIfTime in several places.
The problem is if it's 11pm at the moment I'm testing this IVR, I can't
nicely test the 9am or 2pm branch.
Suggestions ?
How about setting, say, LUNCHTIME to
secret=haramikuttasala
username=wumingzi
type=friend
secret=kickyourass
Enjoy!
B.R
BaBa Jigger
I forwarded this to techsupp...@axvoice.com, just in case they didn't
already know. I also apologized if I was the 10,000th person to do so.
--Barry Miller
On Sun, Oct 18, 2009 at 10:46:05AM -0800, Mr. James W. Laferriere wrote:
http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/
Thank you for creating this site keeping up the info available there .
But I'd -really- like to see Siemens Data Sheet on the product
On Mon, Sep 28, 2009 at 02:26:29PM +0200, Tzafrir Cohen wrote:
On Mon, Sep 28, 2009 at 04:56:01AM -0700, Landy Landy wrote:
I have a similar problem with DAHDI. If my server gets rebooted, I can't
make any calls until the a call come in from outside. From there I can
answer the call and
On Mon, Sep 28, 2009 at 04:09:43PM -0500, Shaun Ruffell wrote:
On 09/28/2009 01:06 PM, Danny Nicholas wrote:
Funny. The first thing I always do after a reboot is call in from my
cell to make sure things work. But last night I rebooted and immediately
tried dialing out (with a TDM842B) and
On Thu, Sep 10, 2009 at 05:06:01PM -0500, Doug Bailey wrote:
- Doug Bailey dbai...@digium.com wrote:
- Doug Bailey dbai...@digium.com wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote
On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840
work
fine.
With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a
line
On Wed, Sep 02, 2009 at 09:44:05AM -0500, Doug Bailey wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work
fine.
With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line
polarity reversal
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine.
With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line
polarity reversal. Stutter dialtone is generated as expected.
Has anyone else seen this? Is there anything special I need to do for
1.6.1 to make FSK
On Sat, May 24, 2008 at 12:01:50AM -0400, sean darcy wrote:
Barry Miller wrote:
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
Change IF ( to IF(.
___
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