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 $end, expecting
> '-' or '!' or '(' or
On Wed, Apr 18, 2012 at 08:52:02PM +0200, Olivier CALVANO wrote:
> Le 18 avril 2012 17:56, Barry Miller a ?crit :
> > On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote:
> >> Hi
> >>
> >> can i don't sent into the SIP invite the "Ses
ould
refuse session timers for just those peers that had problems with them,
like one of my ITSPs.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller
> Sent: Wednesday, April 18, 20
On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote:
> Hi
>
> can i don't sent into the SIP invite the "Session Timer" ? on asterisk 1.6
Have you tried 'session-timers=refuse' ?
--
Barry
--
_
-- Bandwidth and Coloc
On Mon, Apr 09, 2012 at 06:21:40PM -0400, sean darcy wrote:
> I've cut and pasted from the digium fax admin manual:
>
> exten => send,1,NoOp( SENDING FAX )
> exten => send,n,Wait(6)
> exten => send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
> exten => send,n,Set(FAXCOUNT=${GLOBAL
On Thu, Mar 01, 2012 at 05:35:34PM -0700, Nunya Biznatch wrote:
> Howdy All,
>
> I'm considering Asterisk / Digium as a replacement to my existing phone
> switch. I need to continue to be able to push analog lines between
> multiple buildings in a campus environment.
>
> The Digium Analog 410 C
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 lo
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
> 1
f done 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
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
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
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 receptionis
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
>
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?
&g
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 .recn
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 14, 2011 at 12:07:05PM -0700, J Gao wrote:
> On 11-07-14 11:42 AM, Barry Miller wrote:
> >On Thu, Jul 14, 2011 at 10:51:02AM -0700, J Gao wrote:
> >>Sorry for hijack this topic, but I have a different question:
> >>
> >>Every time I install Asteris
On Thu, Jul 14, 2011 at 10:51:02AM -0700, J Gao wrote:
> Sorry for hijack this topic, but I have a different question:
>
> Every time I install Asterisk I have to "make menuselect" and to
> select/deselect some items. Now every time I have to write down what I
> selected for future reference. I
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
> > >
> >
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
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=> 20
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
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 year
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
https://wiki.asterisk.org/wiki/display/AST/Di
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
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 ch
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.
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 up
On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
> I have a very simple setup with two SIP routes to my carrier. I need to have
> every other phone call placed to that carrier go to a different address.
>
> This is what I need the call flow to look like. I have spent many hours
> searchin
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
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
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 wrote:
>
> > On
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 wa
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 probabl
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/ast
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 dialpl
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 s
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 publishe
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
> m
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 upg
On Tue, Sep 07, 2010 at 04:02:29AM -0700, Kyle Kienapfel wrote:
> On Mon, Sep 6, 2010 at 12:32 PM, 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,
>
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
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
> > I
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
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 = ${S
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
>
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 gm
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:
> >
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, doe
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 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 =>
> ,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},Americ
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
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.aster
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
On Tue, Apr 13, 2010 at 06:59:01PM -0400, David Backeberg wrote:
> On Tue, Apr 13, 2010 at 6:55 PM, David Backeberg wrote:
> > On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias wrote:
> >> What do you mean with problems on my configuration?
> >> ?This is a FXO port on zapata:
> signalling=fxs_ks
>
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. The
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 digit
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 scan
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 -06
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=22725&type=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 04:42:44PM +, --[ UxBoD ]-- wrote:
>
> - "Barry Miller" wrote:
>
> > On Mon, Jan 11, 2010 at 03:23:33PM +, --[ UxBoD ]-- wrote:
> > > Hi,
> > >
> > > why would Asterisk core dump with the following test d
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 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 "
>
> username=woodsy
> type=friend
> 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
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 prod
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 TDM842
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
On Thu, Sep 10, 2009 at 05:06:01PM -0500, Doug Bailey wrote:
>
> - "Doug Bailey" wrote:
>
> > - "Doug Bailey" wrote:
> >
> > > - "Barry Miller" wrote:
> > >
> > > > On Fri, Sep 04, 2009
On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote:
> > > ----- "Barry Miller" wrote:
> > >
> > > > Hi,
> > > >
> > > > Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840
> > work
> > > >
On Wed, Sep 02, 2009 at 09:44:05AM -0500, Doug Bailey wrote:
>
> - "Barry Miller" 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 DAH
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(nu
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(".
___
-- Ban
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