Re: [asterisk-users] parsing issue
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 $end, expecting
> '-' or '!' or '(' or ''; Input:
> -1*
>
> The variable "Z" has a negative number, which is the code that I need
> to use in the hangup.
> Any idea how can I do this? There is no ABS() function in Asterisk. I
> already filed a request for it but it turns up that it will cost me
> money. How can I remove the sign from a number?
If you're sure Z is non-null and negative, then
exten => rejected,n,Hangup($[-(${Z})])
--
Barry
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Re: [asterisk-users] Delete "Session timer" ?
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 "Session Timer" ? on asterisk 1.6 > > > > Have you tried 'session-timers=refuse' ? > > > yes i have put this option, but asterisk sent in the Header that he support > the > Session Timers, the sip server of the operator sent a session timer too and > asterisk ignor it. > > my objectifs is asterisk don't sent the session timer I'm not sure about 1.6, but on 1.8 and 10, my INVITEs normally say Supported: replaces, timer *unless* the peer definition has 'session-timers=refuse'. Then I see Supported: replaces and CLI 'sip show channel ' on an outgoing call to this peer shows SIP Options:(none) Session-Timer: Inactive Isn't this what you're looking for? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete "Session timer" ?
On Wed, Apr 18, 2012 at 10:59:21AM -0500, Danny Nicholas wrote: > If I interpret the original question correctly, this will make his call > drop. I imagine this would only happen if the remote end _requires_ RFC 4028 timers, which I don't think is very common. I was thinking he would 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, 2012 10:57 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Delete "Session timer" ? > > 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 Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete "Session timer" ?
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 Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] syntax error from digium fax manual ??
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(FAXCOUNT)})
>
> -- Executing [send@sendPDFasFax:1] NoOp("DAHDI/4-1", " SENDING
> FAX ") in new stack
> -- Executing [send@sendPDFasFax:2] Wait("DAHDI/4-1", "6") in new stack
> -- Channel 4 detected a CED tone from the network.
> [Apr 9 15:29:02] WARNING[2912]: ast_expr2.fl:468 ast_yyerror:
> ast_yyerror(): syntax error: syntax error, unexpected '+', expecting
> $end; Input:
> + 1
> ^
> [Apr 9 15:29:02] WARNING[2912]: ast_expr2.fl:472 ast_yyerror: If you
> have questions, please refer to
> https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
> -- Executing [send@sendPDFasFax:3] Set("DAHDI/4-1",
> "GLOBAL(FAXCOUNT)=") in new stack
> == Setting global variable 'FAXCOUNT' to ''
>
> The error seems to be saying that I need a closing "}" or "]", but it
> looks like it has closing brackets.
>
> Any suggestions?
This is exactly the error you'd get if FAXCOUNT is null or not set.
(Because then the expression would be the invalid '$[ + 1]'.)
--
Barry
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Re: [asterisk-users] Digium FXS specifications and limits Question
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 Card manual states it's not recommended to go > beyond 1500 feet distance for an FXS card, and no line should leave the > building or be bundled. The 2400 Series Manual does not have this same > notice. Should it? > > My potential application will be pushing all of 1500 feet and maybe a > teeny bit more. Analog lines will be bundled into PE-89 type > direct-burial rated cable in capacities ranging from 100-pair to > 500-pair. All cable pairs are 24AWG. All cable is privately owned by the > organization. All cable is professionally terminated on each end to > grounded building entrance protection with lightning blocks / surge > protection. This infrastructure has been in place and working on an > existing "old school" digital switch with port cards that don't have > lightning or surge suppression built in, just like the Digium cards. > > Has anyone run a similar configuration and can speak for or against such > an idear? If you must do this, Mike Sandman (and probably others) has a line of loop extenders. I've used his stuff before to keep old POTS stuff running. http://www.sandman.com/longloop.html These devices may also give you a little more lightning protection than you already have, but I wouldn't count on it. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail2ban restarts
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 file. Depending on your platform, this can be the result of a cron job, an entry in /etc/lograte.d, /etc/newsyslog.conf, whatever. It's normal. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird IPs in Fail2ban list
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 all -- 0.23.20.189 0.0.0.0/0 > > I also get things like, 0.0.5.2, etcFail2ban seems to be working when I > am testing. Are these numbers taken from the SIP packet or the TCP/IP > protocol source because they surely are not valid addresses. > > Thanks See fail2ban-regex(1). It's very useful for discovering what fail2ban makes of your log files. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which device auto-registered an extension?
On Sat, Dec 17, 2011 at 03:55:55PM -0600, Warren Selby wrote:
> Why not try set a variable under each device in sip.conf to the same as the
> endpoint name then Dial(SIP/${CustomVar})?
Thanks. That keeps the device-extension association in one place, at the
cost of a little redundancy if done manually:
[0010b5c1867c]
setvar = HEREITISAGAIN=0010b5c1867c
regexten = 543
I've coded (but not yet seriously tested) an optional feature that would
basically automate just what you suggested. It'll be interesting to see
how many think this is useful.
> On Dec 15, 2011, at 7:03 PM, Barry Miller wrote:
>
> > Hi all,
> >
> > In sip.conf:
> > [general]
> > regcontext = autoreg
> >
> > [devabc]
> > regexten = 543
> >
> > creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc
> > registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the
> > dialplan, because there's no device SIP/543. Now I know I can add a line
> > like "exten=> 543,2,Dial(SIP/devabc)" for each and every device that uses
> > regexten, but it would be a lot cleaner to be able to use something like
> > Dial(SIP/${WHAT_GOES_HERE?}) instead.
> >
> > So is there a way for the dialplan to determine which device caused SIP to
> > auto-register an extension?
--
Barry
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Re: [asterisk-users] Which device auto-registered an extension?
On Fri, Dec 16, 2011 at 05:02:11PM +0100, Olle E. Johansson wrote:
>
> 16 dec 2011 kl. 02:03 skrev Barry Miller:
>
> > So is there a way for the dialplan to determine which device caused SIP to
> > auto-register an extension?
>
> Not really, unless someone else can come up with something.
>
> In Asteirsk, the extension hints are the connection from the dialplan to a
> device,
> used for subscriptions and blinking lamps.
>
> exten => 543,hint,SIP/devabc
>
> then you can use
>
> exten=> _5XX,DIAL(/${HINT})
>
> Which opens up the question on how you enter all the hints...
>
> I know Tilghman added something clever recently to new versions of Asterisk,
> but I
> haven't used it myself so I can't describe how it works.
Thanks. I like to keep devices and extensions separate, for security reasons,
and it's a
pain having to define the association in different places, especially when
sip.conf seems
like the logical place to do this. Do you think if I were to code up a
function like
AUTOREG([context,]exten) that returned 1 for auto-registered extensions, 0
otherwise, and
set a channel variable to the registering device name, it might be useful
enough to be
accepted as a feature?
--
Barry
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[asterisk-users] Which device auto-registered an extension?
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/${EXTEN})" in the
dialplan, because there's no device SIP/543. Now I know I can add a line
like "exten=> 543,2,Dial(SIP/devabc)" for each and every device that uses
regexten, but it would be a lot cleaner to be able to use something like
Dial(SIP/${WHAT_GOES_HERE?}) instead.
So is there a way for the dialplan to determine which device caused SIP to
auto-register an extension?
--
Barry
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Re: [asterisk-users] AST-2011-013: Possible remote enumeration of SIP endpoints with differing NAT settings
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 port listed for > responses in the Via header. In 1.4 and 1.6.2, this would > mean if one setting was nat=yes or nat=route and the other > was either nat=no or nat=never. In 1.8 and 10, this would > mean when one was nat=force_rport or nat=yes and the other > was nat=no or nat=comedia. I see that early this year, VOIPPACK (from the folks who brought us SIPVicious) announced "Additionally we improved vp_sipenumerate to be able to scan Asterisk servers regardless of the alwaysauthreject option". I'm guessing this is how they do it. VOIPPACK isn't free, so it's not as widely used as SIPVicious, but it seems to show that there's at least one exploit already out there. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10.0 CallerID question
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 caller ID on the > transferred phone instead of the incoming callerID. My assumption is that > there is some new users.conf or sip.conf flag I'm not setting to make this > functionality work? Hi. I think sendrpid and trustrpid in sip.conf are what you're looking for. I've meaning to test this since 1.8, but haven't gotten around to it yet. I did bookmark this thread from a PIAF forum a while ago http://pbxinaflash.com/forum/archive/index.php?t-8923.html that claims that, at least with some phones, attended transfer with transferee's callerID now works. I hope this helps. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 10 'database show' CLI command
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, just a side-effect. > > > Thanks. The only time it really matters to me is on my little test box, > > so I'll just keep running a patched db.c there. > > It will be easy enough to change. No real reason for me not to add an 'ORDER > BY key' to the get_tree_all SQL statement near the top of the file. I also added the clause to gettree and showkey stmts, because wildcarding database show abc% database showkey 212% can return many results. But I'm not sure if anyone besides me cares :) -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 10 'database show' CLI command
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 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, just a side-effect. Thanks. The only time it really matters to me is on my little test box, so I'll just keep running a patched db.c there. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10 'database show' CLI command
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk rpm build problem
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 .recno_rec_utils.o.d -MP -pthread -Wall -D__DBINTERFACE_PRIVATE -I. > -I.. -Iinclude -Ihash -Ibtree -Irecno > -I/root/rpmbuild/BUILD/asterisk-1.8.5.0/include -O2 -g -march=i386 [snip] > /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: > undefined reference to `__sync_fetch_and_add_4' > /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: > undefined reference to `__sync_fetch_and_add_4' > ccss.o: In function `ast_atomic_fetchadd_int': > /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: > undefined reference to `__sync_fetch_and_add_4' > /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: > undefined reference to `__sync_fetch_and_add_4' > /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: > undefined reference to `__sync_fetch_and_add_4' > cdr.o:/root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: > more undefined references to `__sync_fetch_and_add_4' follow > utils.o: In function `ast_atomic_dec_and_test': > /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:646: > undefined reference to `__sync_sub_and_fetch_4' > utils.o: In function `ast_atomic_fetchadd_int': > /root/rpmbuild/BUILD/asterisk-1.8.5.0/include/asterisk/lock.h:600: > undefined reference to `__sync_fetch_and_add_4' I do not use centos (or build rpms, for that matter), but at least on OpenBSD the "-march=i386" causes the same undefineds you are seeing. I have to force i686 when configuring asterisk: ./configure --prefix=/usr/local CFLAGS=-march=i686 for the build to succeed. I'm not sure if this applies in your case, but maybe it will help. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requires
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk binaries on CentOS version 6
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 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. Is there a way to save its result for > >>later use? > > From the Makefile: > > > ># If the file .asterisk.makeopts is present in your home directory, you can > ># include all of your favorite menuselect options so that every time you > >download > ># a new version of Asterisk, you don't have to run menuselect to set them. > > > Thanks Barry. > > I just tested on 1.8.5. After I run "make menuselect" and saved when > exit, it generate two new files in the current source directory: > -rw-r--r-- 1 root root4544 Jul 14 11:57 menuselect.makeopts > -rw-r--r-- 1 root root4482 Jul 14 11:57 menuselect.makedeps > > So should I do: > cp menuselect.makeopts /etc/asterisk.makeopts > > I notice the name is different. The Makefile want to see > /etc/asterisk.makeopts, but I only got the menuselect.makeopts. Are they > the same file? Yes. The name has to change because other systems can also use menuselect. E.g., dahdi-tools uses ~/.dahdi.makeopts (and /etc/). Or you could just save menuselect.makeopts somewhere and copy it to each new source you fetch. Remember to save it again if you make any changes you want to keep. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk binaries on CentOS version 6
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. Is there a way to save its result for > later use? >From the Makefile: # If the file .asterisk.makeopts is present in your home directory, you can # include all of your favorite menuselect options so that every time you download # a new version of Asterisk, you don't have to run menuselect to set them. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
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
> > > ${CALLERID(num)} includes '-'. Ex:
> > >
> > > 123-456-7890
> > >
> > > How can I strip the dashes from the number, leaving me with
> > > '1234567890'?
> >
> > I would do this in an AGI written in C -- but that's just me...
> >
> >
>
> Let's assume, "hypothetically", I'm looking for a simpler solution directly
> in the dialplan or via a quick run of System(). :-)
I'm pretty sure that 1.2 has the FILTER dialplan function, so
exten => Set(CALLERID(num)=${FILTER(0123456789|${CALLERID(num)})})
should work.
--
Barry
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Re: [asterisk-users] broken SVN asterisk 1.8 ?
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 last changed revision in the build. To see the difference, try: cd asterisk-src-dir svnversion svnversion -c -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI custom ring cadences in 1.8.3
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, 1, Dial(DAHDI/1) exten=> 202, 1, Dial(DAHDI/1r1) exten=> 203, 1, Dial(DAHDI/1r0) Dialing, in sequence: 201 -> 2000,4000; Good 202 -> 667,1333 ; Good 201 -> 667,1333 ; Huh? 203 -> 2000,4000; r0 seems to be default cadence 201 -> 2000,4000 Do I now need to specify r0 every time I dial an FXS line to be sure it will use the default cadence in case I've previously used a custom one? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong time retrieved from system command
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 Mar 21 19:43:35 HKT 2011
Have you tried "${STRFTIME(${EPOCH},Hongkong,%G%m%d-%H%M%S)}" ?
$ date ; TZ=UTC date ; TZ=Hongkong date
Mon Mar 21 10:13:31 EDT 2011
Mon Mar 21 14:13:31 UTC 2011
Mon Mar 21 22:13:31 HKT 2011
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Re: [asterisk-users] uptime
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 weeks,...,46 seconds is $ TZ=EST5EDT date -d @`dc -e '41 365*7 7*+6+24*3+60*26+60*46+p'` Mon Feb 14 22:26:46 EST 2011 about 13 min. before your post, meaning Asterisk apparently used 0 as its start time when calculating uptime. How do you set the system time? Is it possible that Asterisk starts before the time is set? Do you run "ntpdate -b" at startup, or set the time by some other means before the boot completes? Could this result in some really humorous CDRs? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 MIBs
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/Digium+MIB+Definitions -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] S110M not working
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 this. You'll notice in the config line > that I used fxo signalling. > > I'm just thinking that the failure that dahdi_scan see may be because > the s110 isn't getting power. Sorry. I write faster than I read. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] S110M not working
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 failure and ordered another s110. The replacement > doesn't work either, and now I'm confused. > > dahdi_scan recognizes the s110, but it says it fails. Output looks like this: > > [2] > active=yes > alarms=OK > description=Wildcard TDM400P REV E/F Board 5 > name=WCTDM/4 > manufacturer=Digium > devicetype=Wildcard TDM400P REV E/F > location=PCI Bus 05 Slot 01 > basechan=5 > totchans=4 > irq=169 > type=analog > port=5,FXO > port=6,FXO > port=7,FXO > port=8,FXS FAILED > > I tried moving both the s110s to other positions on the TDM400, but it > fails in all of them. > > Is this a power issue? The system it's in only has sata power > connectors, so I had to splice on an ide power connector. Both the 5v > and 12v lines are connected through all the way from the power supply > to the tdm400 card. > > If I force the issue, and manually configure the s110 like this: > --- > fxoks=8 > echocanceller=mg2,8 > --- > the dahdi startup script throws this error: > > Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 8: Invalid argument > (22) > Selected signaling not supported > Possible causes: > FXO signaling is being used on a FXO interface (use a FXS > signaling variant) > RBS signaling is being used on a E1 CCS span > Signaling is being assigned to channel 16 of an E1 CAS span > > > Have I bungled the power connection and/or the config, or am I just > extremely unlucky and have two bad s110s? 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. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
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 asking myself, why these numbers? Are they > looking > for a voip trunk? Or is it just like a serial number for the scan? What? It's SIPVicious. Before it starts its sequential scan, it makes sure that it can tell the difference between a valid peer and an unknown one. It tries two random peers, expecting a 404 response to at least one (most likely both) of them. Then, if it later gets a 401 during the sequential scan, it knows it's found a good peer name that can be targeted for password guessing. On the other hand, if both random guesses elicit 401 responses to REGISTERs, it knows that it can't winnow out the real peers, and (normally) just gives up right there. That's why 'alwaysauthreject' is so effective at stopping the attacks (as opposed to blocking them). But if the attacker uses the '--force' option, which causes the scan to press on regardless, or something other than SIPVicious, only something like fail2ban will help, but that won't save your bandwidth like 'alwaysauthreject' will. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
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 in our ADSL-connected office. > > What are they after, anyway? Merely cheap international calls? I'm guessing free PSTN access. They don't want to DoS you. The scans are an attempt to collect valid extensions for later password guessing attempts. Every one I've seen has used svwar (from SIPVicious), which by default will give up if it can't tell the difference between trying to register (or invite) an unknown peer and a known one. This is why "alwaysauthreject = yes" is so effective, even though it bends RFC3261 a bit. But keep using fail2ban, too. "svwar.py --force" will cause it to scan regardless of response code. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Load Balancing
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
> searching and have not found a working example.
> Call1 exten =>
> NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
> )
> Call2 exten =>
> NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
> )
> Call3 exten =>
> NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
> )
> Call4 exten =>
> NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
> )
> Call5 exten =>
> NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
> )
> Call6 exten =>
> NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
> )
> Call7 exten =>
> NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8
> )
> Call8 exten =>
> NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4
> )
> ..
If your goal is really load balancing, not just alternating between
providers, you might look at the GROUP* functions. Otherwise, if you
hit a stretch where you have, e.g., several even-numbered calls of long
duration mixed with short odd-numbered calls, most of your traffic will
wind up on the same route.
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Barry
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Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 5 Now Available
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 candidate, please see > >>> the > >>> ChangeLog: > >>> > >>> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4 > >> > >> Apologies, this link should be: > >> > >> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5 > >> > >> -- The Asterisk Development Team > > > > Is it worth mentioning somewhere (ChangeLog? This list?) that all the > > asterisk-core-sounds tarballs were updated today? It would remind someone > > [me!] who's trying to upgrade from an earlier rc to rc5 a chance to do a > > 'make sounds' before stopping asterisk for the install. My test system is > > on a slow link, and waiting for the tarball downloads in the middle of > > installing is frustrating. > > > > If you deselect the sounds from menuselect then you don't have to wait for > them > to download, and you can update them at your convenience later. Thanks. I would not have thought of that. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 5 Now Available
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 > > Apologies, this link should be: > > http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5 > > -- The Asterisk Development Team Is it worth mentioning somewhere (ChangeLog? This list?) that all the asterisk-core-sounds tarballs were updated today? It would remind someone [me!] who's trying to upgrade from an earlier rc to rc5 a chance to do a 'make sounds' before stopping asterisk for the install. My test system is on a slow link, and waiting for the tarball downloads in the middle of installing is frustrating. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find ".gsm" audio file length or duration
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 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" file length or duration in dialplan.
> >
> >Set(DUR=$[${STAT(s,/var/lib/asterisk/sounds/en/demo-congrats.gsm)} /
> > 1650])
> > Verbose(Length (seconds): ${DUR})
> >
> > for asterisk >= 1.6
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Re: [asterisk-users] How to find ".gsm" audio file length or duration
On Sat, Oct 16, 2010 at 04:12:14PM +0530, RAJNIKANT VANZA wrote:
> Hi Friends,
>
> I need to find ".gsm" file length or duration.
>
> *E.g.*
> demo-congrats.gsm
>
> sox demo-congrats.gsm -e stat
>
> Above command is display file length in seconds. like as
> Length (seconds): 27.96
>
> I want to ".gsm" file length or duration in dialplan.
Set(DUR=$[${STAT(s,/var/lib/asterisk/sounds/en/demo-congrats.gsm)} / 1650])
Verbose(Length (seconds): ${DUR})
for asterisk >= 1.6
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Re: [asterisk-users] SIP flood attacK
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 svwar.py (part of SIPVicious). It will give up as soon as it realizes it can't tell the difference between attempting to register an invalid extension and a valid one (with an arbitrary password). It's the default in 1.8, but the option goes back at least to 1.4. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available
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/ > > I downloaded this, ran "./configure" followed by "make menuselect" > and I don't seem to have SIP as an available protocol. Is there > something I can do to make it available? It works fine on the most > recent 1.6 version and it's worked on most of the prior 1.8 versions. You probably need to install libssl-dev then rerun ./configure. At least I did (Debian Lenny). Seems chan_sip needs res_crypto which needs libssl. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] http://www.asterisk.org/downloads naming schema
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\
> asterisk-1.4-current.tar.gz
Those are GNU tar options, which {Free,Net,Open}BSD won't like.
--
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Re: [asterisk-users] AGI Delimiter in 1.6
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 and pass to an AGI script. I am now trying to set the > same thing up in 1.6 However because the argument delimter in 1.6 has > changed from pipe to comma this breaks as the AUTH line is also comma > delimited. Thus the AGI sees the AUTH as extra arguments instead of a > single argument. As the AUTH may contain varying number of arguments I > need a new way for a my AGI to access this data. > > Does anyone have any ideas how I might go about this? For an interim fix, setting res_agi=1.4 in the [compat] section of asterisk.conf should work. See UPGRADE-1.6.txt . -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force ip disconnect after register?
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 asterisk does > not force a drop of the connect after so many bad reg attempts I can't > enforce the block until they drop and try again. This allows them to run > the box with reg attempts as long as they maintain their initial connection > or I reset the state tables on the firewall. This is very bad. Is there a > way to force the connection to drop and reconnect after let's say 50 > attempts. Not an exact answer to your question, but if the attacker is using svwar (part of SIPVicious), setting alwaysauthreject=yes in sip.conf will make the probing stop after only TWO tries. svwar first tries registering a few longish, random extensions before it begins a sequential or dictionary scan, to see how you handle unknown extensions. With alwayauthreject set, svwar just gives up, complaining: "ERROR:TakeASip:SIP server replied with an authentication request for an unknown extension. Set --force to force a scan." I still see 3-4 attempts per week from various sites, but now they stop after just two failed registration attempts. Saves lots of wear and tear on my DSL. I still run fail2ban, but after setting alwaysauthreject a few months ago nothing has passed its threshold. And nothing seems to have broken, either. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Archive of security advisories?
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 this list but is there an easier way to find > them than trying to search the list. Recent ones: http://www.asterisk.org/security Back to 2007: http://downloads.asterisk.org/pub/security/ -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SOLVED: What can make G.729a codec hostid change?
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-registration for a simple OS upgrade. > > The README only says that hostid is based on MAC addresses of all NICs, > but that doesn't seem to be true. Does anyone know anything else that > might cause g729 to compute a different hostid? G.729a is happy again. I'm not sure how many people will be affected by this, but here's what fixed it here: Etch loads eth1394.ko, Lenny doesn't. Why, I don't know. But it caused the eth0 PHY to become iface eth1. To get things working again, I added "alias eth0 eth1394" to modprobe.conf, and swapped eth0<->eth1 in udev/rules.d/70-persistent-net.rules and network/interfaces. If there's a simpler way to do this, please let me know. Now my original hostid is back. Thanks, everybody. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What can make G.729a codec hostid change?
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 from etch to > lenny (and to squeeze in no time). > Having a kernel built on purpose to remove some modules is out of line. > > A better solution needs to be provided in cases like these. > > On 7 September 2010 19:15, Roger Burton West wrote: > > > On Tue, Sep 07, 2010 at 10:58:18AM -0700, Dave Platt wrote: > > > > >Note that "ifconfig" will not necessarily show all of your > > >interfaces (hard- or soft-) - only the active, configured ones. > > > > ifconfig -a would help here. Kernel upgrades often seem to bring in new > > default interfaces. > > > > If this turns out to be the problem, rmmod or a custom kernel > > compilation may do the trick. (Of course if you've _lost_ an interface > > you were using under etch this may be more of a problem.) I posted here before contacting Digium. They have been helpful. Here is what I've found : An old Etch dmesg shows this: eth0: VIA Rhine II at 0x1cc00, 00:13:d4:f5:e3:e6, IRQ 10. eth0: MII PHY found at address 1, status 0x786d advertising 05e1 Link 41e1. eth1394: eth0: IEEE-1394 IPv4 over 1394 Ethernet (fw-host0) eth1: link up, 100Mbps, full-duplex, lpa 0x41E1 Current (Lenny) dmesg: [8.257495] eth0: VIA Rhine II at 0x1cc00, 00:13:d4:f5:e3:e6, IRQ 10. [8.258221] eth0: MII PHY found at address 1, status 0x786d advertising 05e1 Link 41e1. [ 39.188147] eth0: link up, 100Mbps, full-duplex, lpa 0x41E1 And a diff between Etch & Lenny /etc/network/interfaces: 9,10c9,10 < allow-hotplug eth1 < iface eth1 inet static --- > allow-hotplug eth0 > iface eth0 inet static So even though the PHY is eth0 on both, it winds up as eth1 on Etch, (because of the Firewire?), but eth0 on Lenny. I have to admit, I never saw that one coming. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What can make G.729a codec hostid change?
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, > > 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 based on MAC addresses of all NICs, > > but that doesn't seem to be true. Does anyone know anything else that > > might cause g729 to compute a different hostid? > > Just one nic? I don't know much about licensing but I do know that if you > swap ethN assignments on network cards the hostid changes. Yes. Just the lone integrated NIC that's always been there. NO hardware changes. Still eth0 with the same MAC address. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What can make G.729a codec hostid change?
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 based on MAC addresses of all NICs, but that doesn't seem to be true. Does anyone know anything else that might cause g729 to compute a different hostid? Console output follows: Connected to Asterisk SVN-branch-1.6.2-r284958M currently running on secundus (pid = 2430) secundus*CLI> g729 show version Digium G.729A Module Version 1.6.2.0_3.1.4 (optimized for k6_3_32) secundus*CLI> g729 show hostid Host-ID: 02:e1:6c:f6:81:a7:06:b6:4d:fc:94:49:83:c5:3e:71:a4:0f:1b:2c secundus*CLI> g729 show licenses 0/0 encoders/decoders of 0 licensed channels are currently in use Licenses Found: File: G729-2028.lic -- Key: G729-2028 -- Host-ID: 98:3e:89:19:af:0c:11:32:49:cc:fc:9b:e4:92:63:bb:fc:0b:26:4d -- Channels: 0 (incorrect host-id) File: G729-4075.lic -- Key: G729-4075 -- Host-ID: 98:3e:89:19:af:0c:11:32:49:cc:fc:9b:e4:92:63:bb:fc:0b:26:4d -- Channels: 0 (incorrect host-id) -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Ben NanoNote?
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 Semiconductor Co", and it runs Linux. > > > > Does someone know if Asterisk has been ported to that platform? > > The real question is, does it have PCI slots for Digium cards? > > And where do I get one of those HUGE coke cans?! ;-) I'm waiting until the Ben Counterweight project is complete. http://projects.qi-hardware.com/index.php/p/ben-counterweight/ -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
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 the DB function in [globals]. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
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 = ${STRFTIME(${EPOCH},,%m)}
> DAY = ${STRFTIME(${EPOCH},,%d)}
> TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
>
> If I evaluate these variables in the dialplan later, using
>
> exten => ,n,Verbose(${TIMESTAMP} - ${YEAR} - ${MONTH} - ${DAY})
>
> My output is as follows:
>
> -- Executing [7...@phones:3] Verbose("SIP/2625-d5f0", "Tue Jul 13
> 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue
> Jul 13 11:08:42 2010") in new stack
> Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42
> 2010 - Tue Jul 13 11:08:42 2010
>
> However, the following line:
>
> exten => ,n,Verbose(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} -
> ${STRFTIME(${EPOCH},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
> ${STRFTIME(${EPOCH},,%d)})
>
> evaluates with what I expect:
>
> -- Executing [7...@phones:4] Verbose("SIP/2625-d5f0",
> "20100713-110853 - 2010 - 07 - 13") in new stack
> 20100713-110853 - 2010 - 07 - 13
>
> Is what I'm trying to do possible? It seems like it's at least recognizing
> that I'm trying to grab a date, but it's not taking the date format
> parameters that I want.
Try adding "preload => func_strings.so" to modules.conf
--
Barry
--
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Re: [asterisk-users] MAC Address prefixes of Voip equipment
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI FXO calls and the 's' extension. No, Jackie-O doesn't work here--it's just an example. Sheesh!
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 DAHDI/1-3 to go to 3456 ...etc > > What is the CLEANEST way to do this? > [...] > > Is there a way that I can simply specify the extension associated with a > given dahdi channel in dahdi_channels.conf? It would seem logical, but I'm > finding no love. If you also know for sure that there ISN'T a way to do > what I'm asking, I'd like to know that too. > [...] Do you mean chan_dahdi.conf? There you can do something like setvar = EXT=1234 channel => 1 setvar = EXT=2345 channel => 2 and EXT will be passed into your dialplan. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [CRON] Right way to restart Asterisk and Zaptel?
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 gmane because it suppresses duplicates.) Can you please pick one or the other? Thanks. Sorry for interrupting. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
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, > > > > 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 replacing whsvoip.globalipcom.com with, say, > whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make > it work? What is different about the two machines you've tried? Hmm. I see something odd. My initial dig came back with a length of 515 bytes. Then: ; <<>> DiG 9.3.4 <<>> @auth210.ns.uu.net _sip._udp.whsvoip.globalipcom.com SRV +ignore [lines deleted] ;; SERVER: 195.129.12.74#53(195.129.12.74) ;; WHEN: Tue Jun 15 16:39:09 2010 ;; MSG SIZE rcvd: 457 Then: ; <<>> DiG 9.3.4 <<>> @auth210.ns.uu.net _sip._udp.whsvoip.globalipcom.com SRV [lines deleted] ;; SERVER: 195.129.12.74#53(195.129.12.74) ;; WHEN: Tue Jun 15 16:39:59 2010 ;; MSG SIZE rcvd: 515 The difference is that sometimes one more server is returned, pushing the response over 512 bytes. Now it's back to 457. I suspect that they know there's a problem, and are trying to figure out what to do about it. (maybe shorten their server names by one or two characters each?) So depending on when each of your boxes happens to issue the SRV query, they could very well be identical but get different results. I doubt it will help, but you can try quoting RFC 2782, "Currently there's a practical limit of 512 bytes for DNS replies. Until all resolvers can handle larger responses, domain administrators are strongly advised to keep their SRV replies below 512 bytes." -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
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 replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
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 level is working fine. > > Can anyone have an idea about it? I think asterisk only does UDP DNS queries. The response here is too long. $ dig _sip._udp.whsvoip.globalipcom.com SRV ;; Warning: Message parser reports malformed message packet. ;; Truncated, retrying in TCP mode. ; <<>> DiG 9.3.4 <<>> _sip._udp.whsvoip.globalipcom.com SRV ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 63565 ;; flags: qr rd ra; QUERY: 1, ANSWER: 8, AUTHORITY: 0, ADDITIONAL: 0 ;; QUESTION SECTION: ;_sip._udp.whsvoip.globalipcom.com. IN SRV ;; ANSWER SECTION: _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy20.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.de.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy00.uk.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.de.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.nl.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy10.uk.whsvoip.globalipcom.com. _sip._udp.whsvoip.globalipcom.com. 860 IN SRV 0 0 5060 whs-proxy20.de.whsvoip.globalipcom.com. ;; Query time: 3 msec ;; SERVER: 66.92.213.114#53(66.92.213.114) ;; WHEN: Tue Jun 15 11:28:09 2010 ;; MSG SIZE rcvd: 515 -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time variables in system application
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},America/New_York,%F_%T)})
>
> instead of this:
> exten =>
> ,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T)})
> and it's displaying the right time now!!
If all you want is your system's idea of the current local time, you can
simplify it to:
exten =>
,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(,,%F_%T)})
Sorry for replying so late. I somehow missed this thread back in April.
--
Barry
--
_
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Re: [asterisk-users] iax calls via checkbox.cc
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 either, not replying to emails.. > >> > > > > > >I don't know about that company, but since you are sounding unhappy with > >them, have you looked into callwith.us? > > > >-Brandon Broyles > > I think I will go back to callwith.us. > I think they only take paypal > > checkbox.cc is not customer friendly. CallWithUs dropped IAX when they converted to FreeSWITCH. But I still use them at home, and their email-only support and call quality remain good. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing storm-prevention behaviour in logger.conf
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 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 '"" > >> ' failed for 'xx.xx.xx.xx' - Wrong password > >> [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 > >> times > >> > >> This IS a good thing to do, but i want to disable this behaviour. We are > >> using fail2ban to ban scripts and people from the Asterisk system. On > >> version 1.4.23 this worked fine, but now this mechanism is in place, i > >> cannot use fail2ban anymore. > >> > >> Is there any option to disable this behaviour, or even better, add it to > >> logger.conf so anybody can decide what to do? I just want all logging and > >> it seems impossible now. > >> Maybe a patch on the source? > > > > If you use a newer version of rsyslogd to do your logging, there is a > > global configuration directive: > > > > $RepeatedMsgReduction off > > > > that will do what you are asking. The issue #14905 patch you mention is > > not in 1.6.2.x. > > > Hi, > > Well, this sounds fair, but this happened after an upgrade to 1.4.29 from > 1.4.23. Nothing else changed in my setup after that. > > My logger.conf : > > [general] > dateformat=%F %T > > [logfiles] > console => notice,warning,error > messages => notice,warning,error > > This tells me i'm not using the syslog feature at all and > /var/log/asterisk/messages is generated by Asterisk and not by syslogd Hi. First, I'm sorry I didn't look more closely at your in your example. Of course you're not using syslog, but rather asterisk's own logging. Second, I just downloaded 1.4.29. The patch that does the "message repeated" stuff is just not there, as Tilghman said. Is it possible that someone applied that patch to your source? Have you tried downloading the 1.4.29 tarball again and recompiling? If you installed asterisk as a package from somebody's repo, I can't really say, but it seems highly unlikely that the patch would be present. I hope this helps a little bit. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing storm-prevention behaviour in logger.conf
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 '"" > ' failed for 'xx.xx.xx.xx' - Wrong password > [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 > times > > This IS a good thing to do, but i want to disable this behaviour. We are > using fail2ban to ban scripts and people from the Asterisk system. On > version 1.4.23 this worked fine, but now this mechanism is in place, i > cannot use fail2ban anymore. > > Is there any option to disable this behaviour, or even better, add it to > logger.conf so anybody can decide what to do? I just want all logging and it > seems impossible now. > Maybe a patch on the source? If you use a newer version of rsyslogd to do your logging, there is a global configuration directive: $RepeatedMsgReduction off that will do what you are asking. The issue #14905 patch you mention is not in 1.6.2.x. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Fax over TDM410P
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 > group=0 > channel => 1 > >> Not a FXS...can you explain to me what were you trying to say? > > > > http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#SignallingType > > > > Yep. > > If you say that's an fxo port, that's a disagreement between what you > > told me and what you told the DAHDI layer. > > You told DAHDI it's fxs. > > Try changing the config to say fxo and tell us what happens. > > Of course, after re-reading what I just wrote, I think I have it backwards. You do. FXO ports want fxs signalling, and vice-versa. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID and distinctive ring detection
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 1st DID works normally, but I get no CallerID on the 2nd because the call is picked up before the FSK spill is sent. In both cases, the spill is sent about 2.8 secs after the start of the 1st ring, and 0.7 secs after the (1st or 2nd) ring ends. But after the default cadence, DAHDI waits for the spill. After the dring cadence, the pickup is almost immediate (about 0.5 sec). Anybody have any suggestions? distinctiveringaftercid doesn't help. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] string length in dialplan
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 to set it to a known value or insert 0's at the beginning until it
> > is 10 digits in length.
> > My PRI provider needs it set to 10 digits always.
>
> ...stuff before...
> exten => _NXXNXX,n,GoSub(set_cid,1())
> ...stuff after...
>
> exten => set_cid,1,NoOp()
> exten => set_cid,n,Set(CURRENT_CID_LENGTH=${LEN(${CALLERID(num)})})
> exten => set_cid,n,GotoIf($[${CURRENT_CID_LENGTH} = 10]?skip_modify_cid)
> exten => set_cid,n,While($[${LEN(${CALLERID(num)})} < 10])
> exten => set_cid,n,Set(CALLERID(num)=0${CALLERID(num)})
> exten => set_cid,n,EndWhile()
> exten => set_cid,n(skip_modify_cid),Return()
>
>
> There is likely a more efficient way of doing that, but I haven't gone
> through
> and looked at the functions to see if there might be a way of avoiding the
> loop :)
His provider wants 10 digits always, so
exten => set_cid,n,Set(FOO=00${CALLERID(num)})
exten => set_cid,n,Set(CALLERID(num)=${FOO:-10})
would work, but in that case he's likely going to present annoying CIDs
like 000666 to his callees.
--
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Re: [asterisk-users] agi debug in Asterisk 1.6?
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 the CLI help but > found nothing similar. Both my 1.6 boxes do not have the command but > my 1.4 box does. Syntax changed from 1.4. In 1.6 it's "agi set debug on". -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial script
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 -0800, Steve Edwards wrote: > On Sat, 6 Feb 2010, Michelle Dupuis wrote: > > [snip] > > > Oh wait, the advent of computers has allowed us to conveniently insert > > the most recent text at the TOP of a message, to prevent people from > > having to reread the same stuff every time. > > A: Because we read from top to bottom, left to right. > > Q: Why should i start my reply below the quoted text? > > Just because you can do something with a computer doesn't mean you should. > > > Just because this list's moderator has chosen bottom posting doesn't > > make it right, logical, common sense, etc. > > (These lists are not moderated...) > > The list owner's choice did not make bottom posting right, logical and > common sense. It was all of those things already. The list owner just made > the right choice. > > > How about we don't belittle people who don't notice? > > (I don't think I belittled anyone for top-posting.) > > We all make mistakes, but once we are informed, continuing to make the > same mistakes indicates either a lack of consideration or stupidity or > both. > > Simple courtesy would be to only include relevant sections of previous > posts and to reply below the quoted text. > > If you don't like the rules of the playground, find another playground. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gotoif Question
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 -0600, Danny Nicholas wrote:
> Hi Gang,
>
> I'm working on a lumenvox app and am having "fun" with the
> Gotoif's on speech/DTMF recognition. If you're using DTMF to enter a number
> instead of speech to enter a numeric value, the engine will often return a
> "confidence score" of 1000 instead of 1-999. Therefore this Gotoif fails:
>
> exten => s,n,GotoIf($["${SPEECH_SCORE(0)}" <= "${THRESHOLD}"]?:tag)
>
>
>
> ${THRESHOLD} was set to 850.
>
> If SPEECH_SCORE(0) <= 850 (1-850) fails
>
> 851-999 succeeds
>
> 1000 fails
>
>
>
> My workaround was this:
>
> exten => s,n,GotoIf($["${SPEECH_SCORE(0)}" = "1000"]?tag)
>
> exten => s,n,GotoIf($["${SPEECH_SCORE(0)}" <= "${THRESHOLD}"]?:tag)
>
>
>
> Is there a better way than using to Gotoif's per evaluation?
>
>
>
> Regards,
>
>
>
> Danny Nicholas
>
>
>
> --
> _
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Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0- Unknown)
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 Fife wrote: > I have had the exact same problem for over a year on my server sporting the > TDM800 but NOT on my server with the TE212P. Both servers run the same > version of Linux, Asterisk and DHADI. > > The problem has remained consistent through all versions of DAHDI 2.0.x > through 2.2.0.2, and every version of Asterisk which I have I've tried which > includes various iterations of 1.6.0, 1.6.1, and 1.6.2. Currently 1.2.6.1. > Surprisingly I also observe that I can even compile & install NEW versions > of Asterisk and/or DAHDI, and NOT observe the bug provided that I do NOT > bounce the server. > > A developer (not an asterisk developer) named Jim Duda posted this issue to > the list back in October of 08. (Asterisk 1.6.0-beta9 & DAHDI 2.0.x > originally for him). After what he described as considerable effort he > found that by changing one line in chan_dahdi.c the issue appeared to be > resolved (below). His simple patch (below) has works (for me too) as a > stop-gap. > > I posted this the DEV list back January of 09, and the issue was reopened > and then closed as 'fixed' . It would appear the issue needs to be > re-reopened, as it's now appearing less specific to my hardware or > configuration. > > https://issues.asterisk.org/view.php?id=13786 > Duplicate issue to 13927 > > If I do not patch chan_dahdi (below), this is what I (still) observe: > ONLY after a system reboot, any attempt(s) to dial from a device attached to > an FXS port on my TDM800P, result in the following error : > > WARNING[2975]: app_dial.c:1502 dial_exec_full: Unable to create channel of > type 'DAHDI' (cause 0 - Unknown) > == Everyone is busy/congested at this time (1:0/0/1) > > BUT after the first INBOUND call to any FXO port on the device, the FXS port > works normally until the next reboot. > > (Asterisk 1.6.2.1 & DAHDI 2.2.0.2 (& earlier ) Centos 2.6.18-164.11.1.el5 #1 > SMP Wed Jan 20 07:39:04 EST 2010 i686 i686 i386 GNU/Linux > > Does anyone else observe this? Could it be specific to certain > (mis)configurations? It's possible that others have the issue but do not > know it. With any inbound call volume it may be nearly transparent :-) > > -Karl > > JIM's ONE-LINE FIX > On line 8730 (I think it's still on this line) of chan_dahdi.c > replace a "return 0" with return "1". > > if (par.rxisoffhook) > return 1; > else > - return 0; > + return 1; > > > > - Original Message - > From: "Walter Arguello" > To: > Sent: Thursday, January 28, 2010 6:15 PM > Subject: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0- > Unknown) > > > > Hi, > > > > I have a tdm22b (2 fxs / 2 fxo) > > > > When Asterisk is just started, outbound calls routing to fxo port, do not > > working with error: > > > > Unable to create channel of type 'DAHDI' (cause 0 - Unknown) > > > > Inbound calls to fxo port work fine. > > > > After first inbound call, the outbound calls starts working. > > > > CentOS 5.4 > > asterisk 1.6.0.21-1 > > dahdi 2.2.1.-1 > > > > Can anybody help me to identify what is the possible cause of problem? > > > > Thanks, > > > > Walter. > > > > > > > > > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk core dumps when using PrivacyManager
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 dialplan
> > extension ?
> > >
> > > exten => 8100,1,Answer()
> > > exten => 8100,n,Set(CALLERID(all)="")
> > > exten => 8100,n,PrivacyManager()
> > > exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid)
> > > exten => 8100,n,NoOp(Number is ${CALLERID(num)})
> > > exten => 8100,n,Hangup()
> > > exten => 8100,n(nocid),Playback(vm-goodbye)
> > > exten => 8100,n,Hangup()
> > >
> > It shouldn't coredump, but what happens if you remove the extra '{}'
> > from the GotoIf, like so:
> >
> > exten => 8100,n,GotoIf($[${PRIVACYMGRSTATUS} = FAILED]?:nocid)
> >
> It still core dumps .. Have opened
> https://issues.asterisk.org/view.php?id=16576
>
> --
> Thanks, Phil
>
I've got to read the whole thing. Sorry. The ':' also has to go.
exten => 8100,n,GotoIf($[${PRIVACYMGRSTATUS} = FAILED]?nocid)
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Re: [asterisk-users] Asterisk core dumps when using PrivacyManager
On Mon, Jan 11, 2010 at 03:23:33PM +, --[ UxBoD ]-- wrote:
> Hi,
>
> why would Asterisk core dump with the following test dialplan extension ?
>
> exten => 8100,1,Answer()
> exten => 8100,n,Set(CALLERID(all)="")
> exten => 8100,n,PrivacyManager()
> exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid)
> exten => 8100,n,NoOp(Number is ${CALLERID(num)})
> exten => 8100,n,Hangup()
> exten => 8100,n(nocid),Playback(vm-goodbye)
> exten => 8100,n,Hangup()
>
It shouldn't coredump, but what happens if you remove the extra '{}'
from the GotoIf, like so:
exten => 8100,n,GotoIf($[${PRIVACYMGRSTATUS} = FAILED]?:nocid)
--
Barry
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Re: [asterisk-users] Feature Request: GotoIfTimeWithOffset
On Fri, Dec 18, 2009 at 12:23:22AM +0100, Olivier wrote:
>
> Today, this IVR is using function AEL GotoIfTime in several places.
> The problem is if it's 11pm at the moment I'm testing this IVR, I can't
> nicely test the 9am or 2pm branch.
>
> Suggestions ?
How about setting, say, LUNCHTIME to "23:00-23:59" and using
GotoIfTime(${LUNCHTIME},...) until you're ready to go live?
I try to do this when there are multiple GotoIfTime's referencing
the same interval in a dialplan. Companies change their minds about
things like lunch times, working hours, shift changes all the time,
and this makes it easier to change down the road.
--Barry
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Re: [asterisk-users] AXVoice Server Hacked.. accounts info leaked
On Fri, Nov 20, 2009 at 12:32:31AM +0500, baba jigger wrote: > AXvoice server hacked. Here are few working accounts > USE XLITE to make calls > > Registrar/Proxy > magnum.axvoice.com:9060 > > Free Sample account > username=xMaxwellSmartx > secret=thanksapache > > 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 also apologized if I was the 10,000th person to do so. --Barry Miller ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - DECT SIP Phones
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 , Does > anyone know where that may (Still|Ever) exist ? Try: http://gigaset.com/shc/0,1935,hq_en_0_152411_rArNrNrNrN,00.html --Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI congestion problem
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 got: > > > > WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to create channel of > > type 'DAHDI' (cause 34 - Circuit/channel congestion) > > > > Restarting all of Asterisk or 'dahdi restart' didn't matter. Neither FXO > > card would make an outgoing call until it had received an incoming one. > > This with DAHDI 2.2.0.2. > > > > This sounds like https://issues.asterisk.org/view.php?id=15429. It is. Thanks. (I'm the guy whose post is blamed on Danny Nicholas.) This: https://issues.asterisk.org/file_download.php?file_id=22725&type=bug against DAHDI SVN-branch-2.2-r7039M fixes it. What's interesting about this is that my normal post-boot testing, which starts with "does incoming PSTN work?", actually served to mask the problem. There's a lesson there somewhere. --Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI congestion problem
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 DAHDI works fine afterwards. > > In your case: is the problem reset by restarting asterisk? 'dahdi > resstart'? > > Also, please look at the reply he gets: > > 0/1/0 (Busy/Congested/Not-Available) - the call was marked as > "congested". Is that the same for you? 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 got: WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) Restarting all of Asterisk or 'dahdi restart' didn't matter. Neither FXO card would make an outgoing call until it had received an incoming one. This with DAHDI 2.2.0.2. --Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1 + TDM840 FSK MWI problem
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 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 > > > > > > > > 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 MWI work? > > > > > > > > [snip] > > > > > > > > > > > > > > The only thing I can think of that would be preventing the > > output > > > > would be > > > > > problems in the interface chip with the On-Hook transfer mode. > > > > > > > > > > If you run a dahdi_monitor on the channel that should be > > sending > > > the > > > > FSK > > > > > spill and look at the results in a program like audacity, you > > can > > > > see if > > > > > the MWI FSK spill is actually reaching the interface SLIC IC. > > > > > > > > > > Something like "dahdi_monitor 1 -t spilloutput.raw" (Monitors > > the > > > > output > > > > > going to dahdi channel 1.) > > > > > > > > Hmm. With both 1.4 & 1.6, without touching > > /etc/[asterisk|dahdi], > > > > I used a butt-set to go off-hook, then back on. I got: > > > > > > > > 1.4.26.1: dahdi_monitor captured stutter dialtone, 4.5 seconds > > of > > > > silence, then the FSK spill. And that's what I heard. > > > > > > > > 1.6.1.6: dahdi_monitor captured stutter dialtone, 1.5 seconds > > of > > > > silence, then the FSK spill. Sounds good with audacity. But > > > > all I heard through the butt-in was stutter dialtone. No FSK > > > > spill at all. > > > > > > > > Here's hoping this tells you more than it does me :) > > > > > > > Actually it does tell me a lot. > > > > > > The problem appears in how the interface chip is being programmed. > > > > > For some reason, the interface chip is not being set to on-hook > > > transfer mode which would allow for the mwi spill to go out on the > > > actual fxs port lines. > > > > > > I am looking to see where the problem lies. (It is either in > > > chan_dahdi > > > or in the driver.) I hope to have more information later. > > > > > > > The problem lies in a race condition between chan_dahdi making an > > ioctl call to > > set the VMWI state (performed in the do_monitor loop ) and a a > > subsequent call > > to set the channel in ONHOOK transfer mode (performed in > > mwi_send_init). This > > requires the driver to send successive commands to the SLIC interface > > chip > > linefeed register. If the command required for the VMWI mode is not > > completed > > by the time the ONHOOK transfer mode is requested, the ONHOOK transfer > > request > > is thrown away and the MWI spill does not get sent. > > > > I will be fixing this in the drivers trunk branch and hope to have it > > committed > > soon. I'm not sure when it will be released. > > > > Another option is to comment out the ioctl call for VMWI in the > > do_monitor loop > > (especially if you do not care about line reversal MWI.). > > > > See https://issues.asterisk.org/view.php?id=15875 for more information Thanks much! I would not have figured that out. Eliminating the ioctl works fine for now-- rpas is not an issue here. --Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1 + TDM840 FSK MWI problem
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 > > > > 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 MWI work? [snip] > > The only thing I can think of that would be preventing the output would be > problems in the interface chip with the On-Hook transfer mode. > > If you run a dahdi_monitor on the channel that should be sending the FSK > spill and look at the results in a program like audacity, you can see if > the MWI FSK spill is actually reaching the interface SLIC IC. > > Something like "dahdi_monitor 1 -t spilloutput.raw" (Monitors the output > going to dahdi channel 1.) Hmm. With both 1.4 & 1.6, without touching /etc/[asterisk|dahdi], I used a butt-set to go off-hook, then back on. I got: 1.4.26.1: dahdi_monitor captured stutter dialtone, 4.5 seconds of silence, then the FSK spill. And that's what I heard. 1.6.1.6: dahdi_monitor captured stutter dialtone, 1.5 seconds of silence, then the FSK spill. Sounds good with audacity. But all I heard through the butt-in was stutter dialtone. No FSK spill at all. Here's hoping this tells you more than it does me :) --Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1 + TDM840 FSK MWI problem
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 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 MWI work? > > > > The ability to do line reversal MWI was added into the 1.6.2 branch. > Looking through the 1.6.1 code base, I don't see anything other than fsk > MWI (with and without Ring Pulse Alert Signalling.) > > In any case, this is set by defining "mwisendtype" in chan_dahdi. > The default for this is fsk spills. > It can be set to "nofsk" if you want to disable the fsk spills. > > The line reversal is set by specifying > mwisendtype=lrev > > Regards, > Doug Bailey > Thanks, but that's not the problem. I _want_ FSK. A few ast_debug's in chan_dahdi tell me that after calling vmwi_generate(), it's taking the MWI_SEND_SPILL path through mwi_send_thread(), and happily sending about 9K bytes of spill, 160 bytes at a time. But my phones (and a butt-set) tell me that nothing is being received. I don't understand the DAHDI ioctls very well. Is it possible that the TDM840 is not in the correct state when the spill is transmitted? Thanks again, --Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.1 + TDM840 FSK MWI problem
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 MWI work? Thanks, --Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan syntax error: need new eyes
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)} : ${MAINNUMBER} )})
> >
> > Change "IF (" to "IF(".
>
> Same result.
Sorry. This time I actually tested it. *After* de-spacing the " = ",
exten => test,n,Set(CALLERID(num)=${IF( $[${CALLERID(num)} > 140] ?
${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
exten => test,n,NoOp(${CALLERID(num)})
behaved properly. At least with 1.4.19.1. FWIW, every time I try to use
whitespace to make a dialplan more readable, it jumps up and bites me.
Again, sorry for jumping in with an untested response.
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Re: [asterisk-users] dialplan syntax error: need new eyes
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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