Re: [asterisk-users] AGI and forking
On Wednesday 13 April 2011 08:08:03 A J Stiles wrote:
Hi. I just want to make sure I understand this before doing something
that might break things spectacularly for our users and customers :)
We are using Asterisk 1.6.2.9 and my programming language of choice is
Perl.
I want, when a call comes in on someone's DDI number (which the person
who dialled it can only possibly have obtained by dialling 1471 after
we called them), to be able to look up the caller's details from one
of our databases (where the number ought to be stored, because we
already dialled it).
Now, this search is going to take some time; so I'd like for the AGI
script to fork a clone of itself, so the parent process can exit and
the dialplan continue on to ring the person's phone, while the database
lookup is done in the background (the script doesn't need to have any
further contact with Asterisk -- it will initiate any necessary future
communication via other channels).
Is this the sort of thing I need?
## begin code snippet ##
#!/usr/bin/perl -w
use strict;
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my %params = $AGI-ReadParse();
$SIG{CHLD} = IGNORE;
if (my $child_pid = fork) {
# This is executed in the parent process
exit;
}
elsif (defined $child_pid) {
# This is executed in the child process
close STDIN;
close STDOUT;
close STDERR;
# Load some more modules and do some stuff
# that will take a long time
exit;
}
else {
die Could not fork: $!;
};
## end code snippet ##
Am I right in thinking I shouldn't have to worry about zombie processes,
because the parent exits before the child and the init in modern Linux
distros is smart enough to deal with orphaned processes itself?
Almost. You should also set a new session ID to ensure that the child gets
a new processgroup. Otherwise, on some systems, it will still wait for the
child to also exit). In Perl, this is accessible from the POSIX module,
function setsid().
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Re: [asterisk-users] Variable stripping/removing part of string
On Monday 11 April 2011 00:25:35 magnu...@inputinterior.se wrote:
Now i am lost.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,2):0:-1})
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-8,
Martela (fax)) in new stack
-- Executing [0424449...@fax.inputinterior.se:5] NoOp(OOH323/Avaya2-8,
fax)) in new stack
But i am looking for the part before (, in my case: Martela
Oh, sorry. You were right before, then. As far as the :0:-1 nomenclature,
what version of Asterisk are you using? It was not supported before 1.4.
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Re: [asterisk-users] Variable stripping/removing part of string
On Monday 11 April 2011 02:56:03 magnu...@inputinterior.se wrote:
It was a 1.8 but then we started to do a lot of development (ooh323) so
today it is Asterisk SVN-may-ooh323_ipv6_direct_rtp-r311741MS-/trunk.
Can hardly se that we have done any changes that would cause my
problem.
Are you sure there's only a single space separating the name from the
opening parenthesis? The :0:-1 nomenclature only removes a single
byte from the end, and if there was more than a single byte, that might
explain the difference. If that's the case, you may be forced to do a loop
to remove all trailing spaces, if that's still important:
While($[${foo:-1} = ])
Set(foo=${foo:0:-1})
EndWhile
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Re: [asterisk-users] Variable stripping/removing part of string
On Monday 11 April 2011 00:07:08 magnu...@inputinterior.se wrote:
Hi!
I try to get rid of some part of CALLERID(name) but I cant realy figure
out a way to do it. For example: CALLERID(name) = Martela (fax) I am
just looking for the part before “ (“ in my case “Martela”. I can’t
serch for “ “, could be many “ “, but only one “ (“, thought i could do
something like:
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1})
But that gave me “Martela “ so my way of doing it is wrong.
Any that can tell me what I am doing wrong or have any better suggestion
howto do it?
You're almost there. The issue is that CUT uses 1-based offsets, not
0-based offsets, so:
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,2):0:-1})
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Re: [asterisk-users] asterisk hints
On Wednesday 06 April 2011 10:09:07 satish patel wrote:
I used following hint dialplan and i ran show hints but its showing only
one extension what about other 200 phones status ?
exten = _7[456]XX,hint,SIP/${EXTEN}
exten = _7[456]XX,1,Macro(stdexten,${EXTEN},sip/${EXTEN})
shirley*CLI core show hints
-= Registered Asterisk Dial Plan Hints =-
_7[456]XX@ora-cam-extensions : SIP/${EXTEN}
State:IdleWatchers 0
- 1 hints registered
It's actually just showing the pattern, which is not any. In order for the
pattern to generate individual items, something must query an individual
hint state. The usual method of doing this would be for a SIP phone to
subscribe to that extension state, but you can also use EXTENSION_STATE
in the dialplan to query individual extensions.
Just note that if you query an extension that comes back with an invalid
devicename, you've still queried that extension, so the Invalid state will
be preserved in your hint list. The pattern match is intended to be a
shortcut for configuring a lot of phones (and allowing new ones to be
populated on the fly), not a shortcut for making a pretty list for the
command line.
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Re: [asterisk-users] Question About Codecs
On Wednesday 06 April 2011 09:49:17 Jon Farmer wrote: Hi I have a call into a MeetMe conference that when I do a core show channel returns NativeFormats: 0x4 (ulaw) WriteFormat: 0x1000 (g722) ReadFormat: 0x1000 (g722) Can someone explain what the differences between Native, Wite and Read are? Your native format is the format that the phone actually uses (on the wire). The read and write formats are what Asterisk expects to send to and receive from the application, because Asterisk has set up a translation path to ensure that the application gets a format that is more conducive to its purpose. Internally to Asterisk, when you ast_read() a frame from the channel, you should expect that, when the frame is a voice frame, the frame will be in the ReadFormat. And, when you ast_write() a voice frame to that channel, it should be in the WriteFormat. -- Tilghman -- _ -- 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] realtime mysql for 1.8
On Wednesday 06 April 2011 14:53:00 Hans Witvliet wrote: I'm going to have a go with realtime mysql. Just wondering, most examples i came across while googling, was with 1.6 systems. So any drastic changes with 1.8.3, table-layout? other pitfalls? This isn't a pitfall that comes with the upgrade, but you should set wait_timeout internal to the MySQL server to 864000 or higher. This will prevent a number of mysterious crashes that are otherwise possible (and difficult to diagnose) with the threaded MySQL client driver. This is the case, whether you use the native res_config_mysql or the abstract res_config_odbc driver. The usual symptom of this problem is that Asterisk crashes on the first call of the day on Monday morning and then is fine (either for the rest of the week, or until the next morning, depending upon how active calls are on your system). -- Tilghman -- _ -- 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 1.8.3
On Tuesday 05 April 2011 20:10:48 Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. This sounds like a deadlock of some kind. Asterisk has a debugging facility built-in for finding this type of problem, but you will need to compile in DONT_OPTIMIZE and DEBUG_THREADS. Also, it would be helpful, but not entirely necessary, to compile in BETTER_BACKTRACES. Once the problem occurs with the recompiled binary, issuing a core show locks should turn up an indication of where the problem lies. -- Tilghman -- _ -- 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] Dialplan matching
On Monday 04 April 2011 09:09:28 Asterisk User wrote: Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) This one is okay. exten = _01137455.,2,Goto(intl-disabled,s,1) Change this to priority 1. exten = _01137477.,3,Goto(intl-disabled,s,1) Change this to priority 1. exten = _0113749.,4,Goto(intl-disabled,s,1) Change this to priority 1. exten = _011.,5,Goto(intl-disabled,s,1) Change this to priority 1. exten = _011.,6,Playback(all-outgoing-lines-unavailable) exten = _011.,7,Wait(1) exten = _011.,8,Playback(please-hang-up-and-dial-operator) exten = _011.,9,Hangup This looks like it should be starting from priority 1, extension s, context [intl-disabled]. Is this correct or should it be: exten = _011870X,1,Goto(intl-disabled,s,1) exten = _01137455X,2,Goto(intl-disabled,s,1) I tried searching for definitive information on voip-wiki, nerd vittles, but there is a lot of confusion. The major problem in your dialplan is that you WANT to have multiple start points, but the way you have it written, there is only ONE start point. Everything else is simply ignored. Extensions will only start in the dialplan from priority 1. -- Tilghman -- _ -- 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] CDR fields not being written from h extension after Dial command completes.
On Monday 04 April 2011 06:58:23 Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 The h extension should be in the context from which the Macro was called, not in the Macro context itself. -- Tilghman -- _ -- 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] CDR MYSQL missing field data
On Monday 28 March 2011 14:23:25 Tilghman Lesher wrote: On Monday 28 March 2011 07:57:10 Eric W. Davenport wrote: alias start = calldate alias callerid = clid These are fine. I was incorrect, here, and there's nobody to blame but myself. The field is ACTUALLY named clid internally, so when I, by default, set ANOTHER field to the clid column, I shut out the default logic which would have put the callerid into the clid column. This has now been fixed in 1.8 SVN, as a default configuration file fix. If you comment out the line marked alias callerid = clid, it should work fine. -- Tilghman -- _ -- 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] CDR Mysql adaptive Colum
On Wednesday 30 March 2011 14:34:29 Henrique Fernandes wrote:
exten = s,1,set(CDR(teste)=${CHANNEL(audioreadformat)})
And is not working, i thought the only diference it i would need the
colum teste in my cdr table right ?
Correct. Did you restart Asterisk after modifying the table? If you set
core debug to 2 (core set debug 2 cdr_mysql), does it spit out any messages
regarding the MySQL CDR driver? (Note that you'll need to have debug lines
going to the console in logger.conf for this to work.)
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Re: [asterisk-users] CDR Mysql adaptive Colum
Do NOT copy me on replies. I do NOT need two copies of your message.
On Thursday 31 March 2011 12:08:53 Henrique Fernandes wrote:
Found something now! i need first to set the CDR and after make the Dial
Like this.
[default]
exten= _X.,1,set(CDR(teste)=${CHANNEL(useragent)})
exten= _X.,2,Dial(SIP/${EXTEN})
Like this, this would work!!
Now i have a question, if i do not use EXTEN and add an entry for each
number like this
[default]
exten= _600.,1,set(CDR(teste)=${CHANNEL(useragent)})
exten= _600.,2,Dial(SIP/600)
exten= _700.,1,set(CDR(teste)=${CHANNEL(useragent)})
exten= _700.,2,Dial(SIP/700)
exten= _800.,1,set(CDR(teste)=${CHANNEL(useragent)})
exten= _800.,2,Dial(SIP/800)
I would have to do like this or thre is an easier way to set the CDR for
all my calls ?
Simple pattern matching:
exten = _[678]00.,1,set(CDR(teste)=)
exten = _600.,2,Dial(...)
exten = _700.,2,Dial(...)
exten = _800.,2,Dial(...)
Or, better:
exten = _[678]00.,1,set(CDR(teste)=)
exten = _[678]00.,n,Dial(SIP/${EXTEN:0:3})
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Re: [asterisk-users] CDR MYSQL missing field data
On Monday 28 March 2011 07:57:10 Eric W. Davenport wrote: Thanks Tilghman for your response. I have the following in my cdr_mysql.conf I put it in sometime yesterday and did not have it till then. However, it did not make any difference. Did you reload after making the change to the config file? [columns] static value = column alias cdrvar = column These are bogus and should never have been uncommented. alias start = calldate alias callerid = clid These are fine. alias src = src alias dst = dst alias dcontext = dcontext alias channel = channel alias dstchannel = dstchannel alias lastapp = lastapp alias lastdata = lastdata alias duration = duration alias billsec = billsec alias disposition = disposition alias amaflags = amaflags alias accountcode = accountcode alias userfield = userfield alias uniqueid = uniqueid There is no reason to have any of these uncommented, unless the column specified after the arrow is different from the field specified before. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx.c: We were unable to say the number
On Sunday 27 March 2011 14:50:37 Mohammad Khan wrote:
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is it too
large?
Do you see any odd in my dialplan?
What do you have CHANNEL(language) set to at the time? What language
packs do you have installed? What is the exact version of Asterisk you
have installed?
Usually, what this error indicates is that you have one or more sound files
missing, unreadable, or in a format that cannot be transcoded to the codec
you're using.
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Re: [asterisk-users] CDR MYSQL missing field data
On Sunday 27 March 2011 19:36:45 Eric W. Davenport wrote: Hello, I have Asterisk-1.8.3.2, dahdi-linux-complete-2.4.1+2.4.1, and libpri-1.4.11.5 installed and running on a Ubuntu 10.04 server all built from source. Everything is working nicely except one small issue. The CDR records are stored in the CSV file correctly and complete. The MySQL storage is working as it should and is automatically updating all the fields except the CLID field. I have compared and constructed and destructed the system 3 times since Thursday and I cannot figure out why the field does not get populated. If I run the import from csv script it correctly populates the CLID so I believe that tables are setup correct. Has any one seen this and could possibly point me at the offending conf file. I am more familiar than I want to be with cdr_.conf files and I cannot find where the problem is. I have browsed all the wiki's, blogs, and emails looking for a hint and I did not find anything. Anything would be appreciated. Do you have alias callerid = clid in your cdr_mysql.conf file (in the [columns] context)? -- Tilghman -- _ -- 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 1.6.2.10 CDR custom added field
On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote: On 03/24/2011 10:45 AM, Rizwan Hisham wrote: You have to use adaptive cdr for this functionality. In 1.8 the conf file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell you everything. If you are using some other cdr engine then you will have to jump into the code of asterisk to make it log the item you want, which includes creating an extra variable in the cdr data struction, creating a function to set/get its value from dialplan, and then changing the sql command to include the extra variable for insertion into DB. I thought it was possible in asterisk 1.6.2 to add extra mysql-fields ?? In asterisk 1.4 you just have one 'userfield', but in 1.6.2 it is possible to add custom fields... I just don't know how. This is what the wiki (http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql) tells : /Module now permits arbitrary columns to be created and populated, just like cdr_adaptive_odbc, simply by adding the column to the table and defining the corresponding CDR() variable/ Where is the information on this ? Same as always, in the configs/ directory of addons 1.6.2. The sample configuration file contains common examples of the added functionality. Also, there's a note on it in UPGRADE.txt, in the root directory of addons 1.6.2. If you have any further questions, you're welcome to ask this list. -- Tilghman -- _ -- 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 is the most stable version of asterisk?
On Thursday 24 March 2011 11:38:54 Gordon Henderson wrote: On Wed, 23 Mar 2011, Douglas Mortensen wrote: 1.2? 1.4? 1.6? 1.8? 1.2 has been the most stable version for me. Same setups with 1.4 +DAHDI has never been as stable with random crashes and re-starts - however they're not predictable and sometimes months apart. I had one instance of 1.2 run for over a year without a hiccup. I've not even thought about 1.8 yet. There is an inherent danger in running 1.2 code at this point, however. Any security issue that applies to 1.2 won't be patched by the Asterisk team, now that it has passed out of security maintenance mode. You'll need to watch for future vulnerability reports, keeping in mind that some vulnerabilities will only apply to Asterisk 1.2, not later versions (in which case Digium _may_ silently ignore the reports), and you may need to patch those manually. Depending upon your setup, this may or may not be a big concern, but you should at least be aware of it. If your Asterisk 1.2 box is public-facing, this is a potential risk that you should mitigate. -- Tilghman -- _ -- 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] Fwd: asking for some help
On Thursday 24 March 2011 12:02:38 vip killa wrote: If you are new to VoIP, you are better off learning FreeSWITCH And if you're new to analog recordings, you're better off purchasing Sony BetaMax. How is your BetaMax deck, btw? -- Tilghman -- _ -- 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 1.6.2.10 CDR custom added field
On Friday 25 March 2011 04:36:28 Jonas Kellens wrote: On 03/25/2011 08:19 AM, Tilghman Lesher wrote: On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote: On 03/24/2011 10:45 AM, Rizwan Hisham wrote: You have to use adaptive cdr for this functionality. In 1.8 the conf file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell you everything. If you are using some other cdr engine then you will have to jump into the code of asterisk to make it log the item you want, which includes creating an extra variable in the cdr data struction, creating a function to set/get its value from dialplan, and then changing the sql command to include the extra variable for insertion into DB. I thought it was possible in asterisk 1.6.2 to add extra mysql-fields ?? In asterisk 1.4 you just have one 'userfield', but in 1.6.2 it is possible to add custom fields... I just don't know how. This is what the wiki (http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql) tells : /Module now permits arbitrary columns to be created and populated, just like cdr_adaptive_odbc, simply by adding the column to the table and defining the corresponding CDR() variable/ Where is the information on this ? Same as always, in the configs/ directory of addons 1.6.2. The sample configuration file contains common examples of the added functionality. Also, there's a note on it in UPGRADE.txt, in the root directory of addons 1.6.2. If you have any further questions, you're welcome to ask this list. alias start = calldate alias callerid = clid ;alias uniqueid = uniqueid But this is not explained... Alias allows you to rename a standard named column to another column name. I agree that the commented items are confusing. However, both of the uncommented ones are common renames of the standard columns. So please can you confirm how I think it should work : In my dialplan I have : /exten = 600,n,Set(CDR(mycolumn)=myvalue)/ So I should add the following to cdr_mysql.conf : /[columns] static mycolumn = mycolumn/ No, what this will do is add the static definition of the literal value mycolumn to the mycolumn field. What you actually want is to add the field to your table (ALTER TABLE ... ADD COLUMN ...) and add it to your extensions.conf (and reload). That's it. There is literally nothing you have to change in the cdr_mysql.conf file to add an extra column. There is also literally nothing you have to change in the cdr_mysql.conf file to _delete_ a standard column. Just have the column not appear in the backend table (ALTER TABLE ... DROP COLUMN ...) and reload Asterisk. The static definition is for implicit values only. The alias column is just for renaming standard columns. -- Tilghman -- _ -- 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] Back-to-back asterisk PRI issue
On Friday 25 March 2011 15:11:49 Doug Lytle wrote:
satish patel wrote:
group = 0,24
Granted, I'm still running 1.4.x, but I don't believe the above is
valid.
My guess is it should be:
group = 0
No, that's valid. You can have any of groups 0-63 set on a single
group of channels. They are for group selection of channels, as in
Dial(DAHDI/g0/${EXTEN:1})
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Re: [asterisk-users] Back-to-back asterisk PRI issue
On Friday 25 March 2011 14:40:40 satish patel wrote: Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1][PRI]-Cross Cable-[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel = 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_cpe channel = 1-23 Here's one confusing part. You're saying that calls that come from the master to the slave end up in context from-pstn (on the slave), but calls from the slave to the master ALSO end up in from-pstn (on the master). Seems like one of them should be from-internal or the like. I'm sure some of your problem emanate from these settings. satish-desktop*CLI [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) Check the other side for error messages. [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument This problem is due to a misconfiguration. Asterisk cannot handle the local network being addressed as the 0.0.0.0 network. You need to use the full local address. -- Tilghman -- _ -- 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] Back-to-back asterisk PRI issue
On Friday 25 March 2011 16:23:27 satish patel wrote: I just start Pri set debug on span 1 and its showing D-channel is down How do you have the underlying T1 signalling set up in /etc/dahdi/system.conf (on both ends)? -- Tilghman -- _ -- 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] Usage of lock in CDR
On Tuesday 22 March 2011 00:56:05 Nikhil wrote: Hi all In asterisk source code we can see lots of places AST_CDR_FLAG_LOCKED flags is used.This is for CDR purpose. Does anyone what is exact usage of this lock in CDR.If I remove this flags where it will impact,any data overwrite will happen..? Yes. The purpose of the lock is to force a record to become a snapshot at the time the record was locked. If you remove that flag, then almost any update to the CDR will overwrite the entire record with a new snapshot. -- Tilghman -- _ -- 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] Usage of lock in CDR
On Tuesday 22 March 2011 02:20:24 Nikhil wrote: Thanks for reply. I am trying to understand how CDR in asterisk is working(Code wise),because some issue are there in CDR in call feature scenarios like call transfer ,call forward etc.I wanted to fix that issues for that I am planning to rewriting the CDR logic.Do you have any document which explain how CDR works in asterisk and what are its limitations. Good luck with that. That subsystem has been gone over many times in an effort to improve it. The basic problem is that CDR is designed for a much simpler time, when we did not have things like call transfers, 3-way calling, call conferences, and the like, so it is maladapted for modern usage. A much better method is the CEL subsystem, which generates events which can be analysed after the fact to create a much clearer picture about how a call progressed. -- Tilghman -- _ -- 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 Monday 21 March 2011 06:45:37 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
I am looking for help.
Do you have an file (or symlink) in /usr/share/zoneinfo called GMT+8? I
certainly don't, and I'm not running anything different from the standard
set of zone files. If you don't have that entry, then the timezone code
will use UTC (i.e. no local differentiations).
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Re: [asterisk-users] chan_sip.c:3115 __sip_xmit of 0x108d33c0 (len 523) to xxx.xxx.xxx.xxx:0 returned -1: Invalid argument
On Wednesday 16 March 2011 06:09:33 Ishfaq Malik wrote: Does anyone know what this error is about? I've had 0 success in trying to find any reference to it on the internet Well, the most obvious problem is that you cannot send (or bind, or do anything, really) to port 0. -- Tilghman -- _ -- 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] Executing shell commands via AMI
On Wednesday 16 March 2011 13:14:40 Vinícius Fontes wrote: action: command command: ! /bin/ls -l / For security reasons, you cannot do this. This is intentional, not a bug. Consider the command 'rm -rf /' for the reason why. -- Tilghman -- _ -- 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] Executing shell commands via AMI
On Wednesday 16 March 2011 14:11:21 Vinícius Fontes wrote: I understand the concern with security but why not create a separate authorization allowing that instead of hard-coding it? I understand the concern with security but why not create a separate authorization allowing that instead of hard-coding it? Clearly, you don't understand the problem with security, because you're asking that question. If you want to run shell commands on the Asterisk server, create your own SSH connection to the server, become root, and run those commands. -- Tilghman -- _ -- 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] [1.4] Forcing Asterisk/Zaptel to wait untilcalleeanswers?
On Monday 07 March 2011 08:20:26 Danny Nicholas wrote:
On Monday 07 March 2011 08:14:27 Gilles wrote:
1. Why use instead of = to compare the extension with SIP?
exten = s,n,Gotoif($[${EXTEN} SIP]?start)
#1 is Lazy notation to say ${EXTEN} starts with SIP (as opposed to
Local or DAHDI)
Then you probably want ${CHANNEL}, not ${EXTEN}. ${EXTEN} is always
going to be s, which is always greater than SIP.
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Re: [asterisk-users] Inadyn error
On Sunday 06 March 2011 13:35:13 John Novack wrote: Any clue what this means? Mar 6 14:00:30 NCTM user.warn INADYN[1641]: INADYN:IP: Error 0x68 in recv() Mar 6 14:00:30 NCTM user.warn INADYN[1641]: W: DYNDNS: Error 'RC_IP_RECV_ERROR' (0x15) when talking to IP server Mar 6 14:00:30 NCTM user.warn INADYN[1641]: W:'RC_IP_RECV_ERROR' (0x15) updating the IPs. (it 0) Previously INADYN was working Perhaps you'd have more luck emailing the authors of inadyn, instead of emailing the Asterisk users' list? -- Tilghman -- _ -- 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, Sent accountcode between 2 asterisk
On Sat, Mar 5, 2011 at 11:52 AM, Steve Edwards asterisk@sedwards.com wrote: On Sat, 5 Mar 2011, brya...@zktech.com wrote: Send the account code as a custom header variable encode it on A and read it on B. You can send any variables you want using this method. I currently send about 10 variables on switch transfers. If you need an example ping me back and I will send one when I get in the office. Just noticed you are using IAX I don't think my method works with IAX. That is why I use SIP between systems. Someone correct me if there is a way to send custom variables with IAX. You can pass cruft between Asterisk servers via IAX using the caller ID name. In 1.6.2 and above, you can set arbitrary variables with IAXVAR() on one side and retrieve them on the other side. -- _ -- 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] mySQL connection testing
On Thursday 03 March 2011 08:42:42 Andrew Thomas wrote: Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? There is no way to test it. If you want this, you should track the information yourself or don't disconnect anywhere but in the h extension. BTW, the disconnect is not strictly needed in all versions of the addons since 1.4.9. Due to the possibility of a memory leak, the connections are tracked and deleted when the channel is destroyed. See this issue (and the patch) for more information: https://issues.asterisk.org/view.php?id=14757 -- Tilghman -- _ -- 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] mySQL connection testing
On Friday 04 March 2011 02:47:56 Andrew Thomas wrote: If mySQL in the dialplan is so bad - why did Digium include it in the first place? Digium is not responsible for everything that appears in Asterisk. This is a community project, and community volunteers have written large swaths of Asterisk, including the MYSQL command. -- Tilghman -- _ -- 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] mySQL connection testing
On Friday 04 March 2011 03:03:41 Andrew Thomas wrote: Thanks Tilghman - this is exactly what I wanted to hear. As for the 'inclusion' bit - true, but it's still infused in to the addons package at the Digium end (isn't it?). While Digium hosts the repository and the project head (Russell) is a Digium employee, what winds up in the repository is largely up to the Asterisk community, including many non-Digium developers with commit access. While Digium does contribute a great deal to the releases, suggesting that Digium is responsible for everything that ends up in a release is reductionist and diminutive of the many contributions made by the community. -- Tilghman -- _ -- 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] Failover Routing
On Wednesday 02 March 2011 07:06:31 Andrew Thomas wrote:
It seems like it is a v1.8 only function at present (unless a backport
is released).
From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
-
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,channel-name)}
Asterisk 1.8 also comes with a 'use_q850_reason' configuration option
for generating and parsing, if available: -
That will give you what you want if you consider upgrading to v1.8.
A backport on this is not possible. It depends upon some core
functionality introduced in the 1.8 branch.
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Re: [asterisk-users] How do I find a phone numbers issued by Rogers?
On Wednesday 02 March 2011 19:17:03 Robert Augustyn wrote: Is there a way of finding out what block of phone numbers were issued to Roger’s business customers in my end of the woods? You can find out from NANPA, the registry which assigns blocks of phone numbers. Note that due to phone number portability, however, this only will tell you the numbers that were originally allocated to Rogers, as customers are free to request existing numbers to be ported to them, and former customers are free to port their numbers away from Rogers. -- Tilghman -- _ -- 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] [1.4] Still can't get it to call back
On Fri, Feb 25, 2011 at 4:54 AM, Gilles codecompl...@free.fr wrote: Is there a way to launch a script asynchronously, so that Asterisk proceeds to the next step immediately, and the script will then wait 10 seconds so that the channel is available again? In Perl, the line would be: fork and exit; I'm sure there's an equivalent in lua, but the basic idea is that you want to fork a child, which takes over. When the parent process dies, control returns to the dialplan. -- _ -- 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] NVFaxDetect causing segfault
On Monday 21 February 2011 21:11:28 Shamus Rask wrote: 3. Is it true that Digium is sidelineing IAX2 and only focusing on SIP? Should I be looking to migrate to SIP trunks instead? Is it true that space aliens stole your brain and replaced it with a head of cabbage? -- Tilghman -- _ -- 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] pbx_ael.so: undefined symbol: ast_compile_ael2
On Friday 18 February 2011 05:29:56 Borin wrote: Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Make sure res_ael_share.so is loaded first. -- Tilghman -- _ -- 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] application for voice modulation
On Thursday 17 February 2011 14:13:04 Albert wrote: On 17.02.2011 20:21, Paul Belanger wrote: On 11-02-17 07:47 AM, Albert wrote: Hi guys, i am looking for application to modulate voice of speaker. This is supposed to be FUN type of service, where user can call a premium number and from IVR menu choose woman's or men's voice type and then call friend to make him a joke :) Is there such application in standard asterisk's applicartions ? And if no maybe there is some add-on. *CLI core show function PITCH_SHIFT Thanks Paul. This is what I've been looking for :) Am gonna play around with this baby Watch the values. I was trying unsuccessfully to hear the pitch shift while playing around one day, left it in place, and was greeted the next morning to fits of laughter from the rest of the team on a conference call, because I sounded like Alvin of the chipmunks. -- Tilghman -- _ -- 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] Lua extensions are not working on asterisk 1.8.2.3
On Tuesday 15 February 2011 11:06:32 Carlo Pires wrote: Hi, After compiling a installing asterisk 1.8.2.3 I wanted to play with lua but I noticed that extensions created in extensions.lua was not being registered with asterisk. uga1*CLI dialplan show [ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ] 's' =1. NoOp() [app_queue] [ Context 'parkedcalls' created by 'features' ] '700' = 1. Park() [features] [ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ] 's' =1. NoOp() [app_dial] [ Context 'local' created by 'pbx_lua' ] Alt. Switch ='Lua/' [pbx_lua] [ Context 'demo' created by 'pbx_lua' ] Alt. Switch ='Lua/' [pbx_lua] [ Context 'default' created by 'pbx_lua' ] Alt. Switch ='Lua/' [pbx_lua] -= 3 extensions (3 priorities) in 6 contexts. =- uga1*CLI uga1*CLI dialplan show demo [ Context 'demo' created by 'pbx_lua' ] Alt. Switch ='Lua/' [pbx_lua] -= 0 extensions (0 priorities) in 1 context. =- uga1*CLI Need I enable something to get lua extensions to be created? No, that's how Lua extensions work, with the switch statement. Your extensions are still being evaluated by Lua. The only difference is that pbx_lua now doesn't see any need to create extensions, because it will see every extension when it hits the switch. -- Tilghman -- _ -- 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] uptime
On Tuesday 15 February 2011 12:13:37 Jeff LaCoursiere wrote: On Tue, 15 Feb 2011, A J Stiles wrote: On Tuesday 15 Feb 2011, 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? I'm guessting, this is a brand new machine on its first ever boot, with no last bootup time information saved anywhere. So it assumes the last bootup date was 1970-01-01 00:00:00, i.e. zero time on all Unix-like systems. That would explain the 41 years, anyway. No, it is a few months old now with lots of reboots. It is my experiment to build a reliable PBX out of Seagate Dockstar hardware and USB sticks. I've now blown up about three 4G sticks, presumably for heat issues (?), and have just bought an SLC based stick (which was not easy to find) that supposedly has a better heat range and longer write life. Once I got it working I saved a dd image of the stick and have just been imaging the new sticks as I try them. This *is* the first boot on this new stick, but the filesystem itself is old, with several reboots. I don't really care about the uptime calculation, just thought it was funny. It is strange that asterisk and the OS don't agree... so how does asterisk compute it? The system has a monotonically incrementing integer (jiffies), which is used for uptime calculations. Asterisk just stores the time the process started, then performs a simple subtraction from the current time. If your system clock doesn't have large jumps while Asterisk is running, it's a good calculation. BTW, the Linux uptime counter also has an inherent problem: at some point, the counter will overflow and uptime will return to 0. On older Linux systems, this occurred at 497 days. -- Tilghman -- _ -- 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] On-Hold Music
On Monday 14 February 2011 08:23:08 Danny Nicholas wrote: Might not be your question to answer, but if I did get a BMI license, this would allow me to use virtually any music I wanted for MOH? The answer is, as long as the music publisher for each piece of music has an agreement with BMI to license their music, yes. You can verify each individual title here: http://www.bmi.com/search/ -- Tilghman -- _ -- 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] CDR with unix time.
On Thursday 10 February 2011 12:33:40 Rodrigo Lang wrote: 2011/2/10 Tilghman Lesher tilgh...@meg.abyt.es On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote: I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? The answer is, it depends upon the backend version you're using. With cdr_pgsql and cdr_mysql from 1.6.2 forward, if the column type is integer or float, then the unix timestamp will be used. Without any modification? Only with the column type, Asterisk will modify the common date to unix time? The idea behind this is that we don't want to lose any information. Thus, if the datatype is numeric, then the only way to ensure that we don't lose information during the insert is to set the data to a unixtime format. Note that we can even store fractions of a second in this way, if the column type supports it (i.e. decimal or float). -- Tilghman -- _ -- 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] On-Hold Music
On Friday 11 February 2011 16:37:49 Danny Nicholas wrote: Hi gang, In 500 words or less (if possible), please explain what is a legal music-on-hold file? My boss hates the stuff provided with the distribution and I figure that I'm asking for trouble if I take my Les Mis tracks and run them through Audacity and SOX to make new files. The proper licensing authority in the United States for hold music is BMI (Broadcast Music Inc). If you use music for MOH which is not royalty- free, then BMI requires a payment for each trunk line per year which is using such music. If you want to use for-royalty music, it is very possible, but it will be a continual expense. Not paying the fees upfront will cost you dearly in legal fees at the point at which you are caught (it's really only a matter of time). http://www.bmi.com/licensing/entry/534929 For future reference, I now work for a company which gets paid with fees generated by the music business (including MOH), so fair warning: if you announce that you're illegally evading such royalties (note that the use of royalty-free music, as is distributed with Asterisk, is perfectly legal), you may get a visit from the BMI enforcement division shortly thereafter. -- Tilghman -- _ -- 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] CDR with unix time.
On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote: I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? The answer is, it depends upon the backend version you're using. With cdr_pgsql and cdr_mysql from 1.6.2 forward, if the column type is integer or float, then the unix timestamp will be used. -- Tilghman -- _ -- 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] Call Recording audio file quality query
On Wednesday 09 February 2011 03:50:51 Sherwood McGowan wrote: Tilghman, When you say reformat the audio, do you mean sample rate and bits per sample, etc...or do you mean the format in which each packet of data is structured ? I just want to make sure I know which one I'd be dealing with if recording a call that was using one of the higher quality codecs that was metioned earlier. I *think* you mean just the structure version of the format options I presented, because for example: Microsoft PCM (wav) files can be of varying quality levels (192Khz, 256Khz..8bit 16 bit 24...32)..This is true (as you know, I'm more than sure) of almost every audio file format... So, is it Structure of data/packets or sample rate, bitrate, etc' ? That would be structure of data stored in the file. At the point where the file format comes into play, the samples are already in their final stage of computation. The only thing that remains is how the samples are wrapped for storage. -- Tilghman -- _ -- 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] Callback through extensions.conf?
On Wednesday 09 February 2011 06:28:43 Gilles wrote: On Wed, 09 Feb 2011 11:47:09 +0100, Gilles wrote: Unfortunately, I checked how the uClinux kernel was configured for compiling, and the inotify is indeed selected by default :-/ Greping the Asterisk source code for inotify only returned a couple of hits, in binaries (./main/logger.o and ./main/asterisk). So I guess Asterisk doesn't use the inotify Linux feature. Inotify for spoolfiles is supported starting in Asterisk 1.8. -- Tilghman -- _ -- 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] AEL Eswitches
On Wednesday 09 February 2011 13:31:36 Thiago Maluf wrote:
Hi List,
Would someone can to explain me the main difference in SWITCHES or
ESWITCHES in AEL.
context default {
switches {
DUNDi/e164;
IAX2/box5;
};
eswitches {
IAX2/context@${CURSERVER};
};
};
A switch evaluates variables at load time. An eswitch evaluates variables
locally at query time. An lswitch sends the variables intact through to
the switch backend.
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Re: [asterisk-users] Call Recording audio file quality query
On Tuesday 08 February 2011 06:34:56 Sherwood McGowan wrote: On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote: But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will firehose the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device sees and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated recording server...all it has to do is find the signaling packets for each call and then just dump the payload from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the DECoding portion is there, there's almost GOT to be the enCOding functionality... Actually, the writing of encoded voice has nothing to do with codecs. The format modules simply expect a particular type of packet to be fed in, and they simply reformat the audio (without transcoding) to be stored on disk. One caveat is that the format in which they are stored on disk is not guaranteed to be a standard format that is at all useful to outside utilities; just that Asterisk can read it off disk and reassemble the packets. -- Tilghman -- _ -- 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] Zaptel slow to detect remote hangup
On Saturday 05 February 2011 03:24:11 Gilles wrote: Hello I hooked up an Asterisk appliance to an analog phone line, and I notice the phones keeps ringing twice after the remote caller has hung up. /etc/zaptel.conf has the right country parameter set. /etc/asterisk/indications.conf has locale information, but it's apparently used by the pbx_indications module. Is there a way to tel Zaptel detect this type of event faster? No. It is a limitation of the protocol. If you want faster notification, use a digitally signalled line. -- Tilghman -- _ -- 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] asterisk18 rpm issues
On Wednesday 02 February 2011 14:21:50 Jason Parker wrote: On 02/02/2011 02:14 PM, Frank Liu wrote: Hi there, Per the instruction from http://www.asterisk.org/downloads/yum , I setup the yum repository on my Centos 5 x86_64 machine and did a yum install asterisk18 asterisk18-configs then I startup the asterisk (with no changes to config) just to see if it runs, but see below errors in the /var/log/asterisk/messages: [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'res_pktccops': /usr/lib/asterisk/modules/res_pktccops.so: cannot open shared object file: No such file or directory [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'chan_mgcp.so': /usr/lib/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc I checked the system and can't find the file /usr/lib/asterisk/modules/res_pktccops.so at all. I double checked the rpm file downloaded by yum and res_pktccops.so is not in any rpms. Asterisk should still load fine with this warning. chan_mgcp wouldn't work, but that isn't used very often. I will take a look at it. This is not true for CentOS 5 and other distributions where the version of GCC does not support attributes weak or weakref. See this issue: https://issues.asterisk.org/view.php?id=17707 I'm guessing a packaging error simply did not include that new module as an oversight. -- Tilghman -- _ -- 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] AGI script exits non-zero when running system command
On Tuesday 01 February 2011 23:43:34 Charles Solar wrote: Hey guys I was hoping I could get a few pointers on a problem I have been trying to debug for the last couple of months regarding asterisk AGI scripts and unexpected termination. I have this agi script that accepts incoming faxes using RxFax on the latest asterisk 1.4 branch. Its written with perl and it works fine except for one line that causes the entire script to terminate unexpectedly. snip AGI Tx 200 result=0 AGI Rx VERBOSE Converting /tmp/1296624119.53.tiff to /tmp/1296624119.53.pdf 1 fax.agi: Converting /tmp/1296624119.53.tiff to /tmp/1296624119.53.pdf AGI Tx 200 result=1 Really destroying SIP dialog '371b80c6324ece0c779653c34d2e88a2@XXX' Method: INVITE == Spawn extension (from-trunk, XX, 3) exited non-zero on 'SIP/trunk-0035' This isn't the script terminating non-zero. It's the channel hanging up. One possible problem might be that your script is not properly handling the SIGHUP signal sent to the AGI process when a hangup occurs. If that is the case, then your script may be terminating early due to the signal. The best way to handle that is to set a signal handler in your script (this is dependent upon the language you're using), although there's also a workaround for people who are unwilling or unable to set a signal handler. Just remember that prior to Asterisk 1.6.2, once you receive the SIGHUP, you may no longer interact with the Asterisk process. That includes setting and retrieving variables and using the VERBOSE command. Starting with Asterisk 1.6.2, an AGI is free to continue interacting with Asterisk (the setting of final variables is likely the most productive task). -- Tilghman -- _ -- 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] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
On Tuesday 01 February 2011 02:24:20 Benny Amorsen wrote: Tilghman Lesher tilgh...@meg.abyt.es writes: Correct; and Asterisk needs to be started as root, even if it will drop privileges after startup. Do this, and there should be no problems. Starting as root + dropping privileges is fine. Running configure as root is not so fine; that basically makes building RPMS impossible. Alternatively, if you can set ulimit -n 32768 in your RPM build environment (this needs to be set as a login requirement), you can sidestep the need for configure to run as root. The only reason it needs root is to expand the file descriptor limit so it can test using a file descriptor beyond 1023 (the usual limit). -- Tilghman -- _ -- 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] Return variables from func_odbc calls?
On Tuesday 01 February 2011 11:49:51 Paul Belanger wrote: On 11-01-26 02:59 PM, Tilghman Lesher wrote: On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote: [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); That assumes you have only one call in existence at a time. If two calls came in and executed the query at about the same time, it's possible for both reads to return the same value. Yup, didn't even think of that. My testing of ODBC was a single channel. Guess I need another method to return the last ID of the record that was just inserted. Assuming you were using a MySQL backend that supported transactions, you could use the transaction layer in Asterisk 1.6.2 and greater to ensure that each channel got a serialized view. That would make this approach work. -- Tilghman -- _ -- 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] Return variables from func_odbc calls?
On Tuesday 01 February 2011 12:36:46 Jose P. Espinal wrote: Paul Belanger wrote: On 11-01-26 02:59 PM, Tilghman Lesher wrote: On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote: [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); That assumes you have only one call in existence at a time. If two calls came in and executed the query at about the same time, it's possible for both reads to return the same value. Yup, didn't even think of that. My testing of ODBC was a single channel. Guess I need another method to return the last ID of the record that was just inserted. In this case, does the Asterisk connection to MySQL through odbc counts as a unique 'client', or does each call to a function will count as a 'client'? The first. But you need to also understand that unless you use transactions, and specifically the transaction support in Asterisk, each channel is not guaranteed to be using the same connection on the second query. Or even if they all use the same connection, the queries are not serialized in the way that you might otherwise expect. The transaction support introduced in Asterisk 1.6.2 allows a connection to be reserved exclusively to a single channel, thus ensuring that the second query on a channel really was the very next query on the connection. -- Tilghman -- _ -- 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] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
On Monday 31 January 2011 07:26:25 Benny Amorsen wrote: Sorry for resurrecting an old thread... Tilghman Lesher writes: Out of curiosity, what platform are you running on? On most platforms that are able to run Asterisk, with the possible exception of Solaris, increasing the maximum file descriptor for use with select(2) is possible. I am not entirely sure yet, but it looks like Asterisk 1.8.x fails to increase the maximum file descriptor when running on Linux, if configure is not run as root. If configure is run as root, everything works as expected. Correct; and Asterisk needs to be started as root, even if it will drop privileges after startup. Do this, and there should be no problems. -- Tilghman -- _ -- 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] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
On Monday 31 January 2011 15:16:13 cov...@ccs.covici.com wrote: Benny Amorsen benny+use...@amorsen.dk wrote: Sorry for resurrecting an old thread... Tilghman Lesher writes: Out of curiosity, what platform are you running on? On most platforms that are able to run Asterisk, with the possible exception of Solaris, increasing the maximum file descriptor for use with select(2) is possible. I am not entirely sure yet, but it looks like Asterisk 1.8.x fails to increase the maximum file descriptor when running on Linux, if configure is not run as root. If configure is run as root, everything works as expected. Not so, I always run ./configure as root and I get the message that 32768 exceeds ... And what platform are you running on? Are you perhaps running on an SELinux platform, where root isn't really root? Or are you running on something that isn't Linux and never bothered to allow a single process to readily handle more than 1024 file descriptors at once? I need an answer if we're going to ever solve this issue, not just whining about how it doesn't work. -- Tilghman -- _ -- 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] Can a duration limit be specified in spool call file?
On Saturday 29 January 2011 05:07:49 DHAVAL INDRODIYA wrote: On Sat, Jan 29, 2011 at 6:19 AM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Friday 28 January 2011 18:27:15 Bruce B wrote: Hi Everyone, I don't see any parameter for limiting duration of a call in the .call file for Asterisk spool outgoing directory. I'd rather not use a MeetMe to drop the call in a conference room and to then limit the call duration as that complicates things unnecessarily. I am wondering if there is anything else I can do or if the Channel parameter take call duration like the DIAL parameter? No, but you can specify a Local channel as the channel in the call file and then set a TIMEOUT(absolute) for the call, before you Dial() the actual channel you want to use. Keep in mind that the actual channel could be specified by a Set variable in the callfile. what about this *WaitTime: number* Seconds to wait for an answer. Default is 45 http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out That only limits the amount of time the callfile allows for the channel to be answered, not the duration of the overall call. -- Tilghman -- _ -- 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] Reducing number of Asterisk processes?
On Saturday 29 January 2011 04:52:02 Gilles wrote: Hello On a uClinux-based appliance, ps aux shows multiple Asterisk processes: 380 root 11990 S asterisk -f 381 root 11990 S asterisk -f 383 root 11990 S asterisk -f 384 root 11990 S asterisk -f 385 root 11990 S asterisk -f 386 root 11990 S asterisk -f 387 root 11990 S asterisk -f 388 root 11990 S asterisk -f 389 root 11990 S asterisk -f 390 root 11990 S asterisk -f 391 root 11990 S asterisk -f 392 root 11990 S asterisk -f 393 root 11990 S asterisk -f 394 root 11990 S asterisk -f 395 root 11990 S asterisk -f 396 root 11990 S asterisk -f 397 root 11990 S asterisk -f 398 root 11990 S asterisk -f 399 root 11990 S asterisk -f 400 root 11990 S asterisk -f 401 root 11990 S asterisk -f I was wondering... 1. Why have more than one? 2. Provided each process is indeed using 11.990 bytes, is it possible to reduce the number of concurrent processes, considering the fact that this appliance will not handle more than a couple of concurrent calls? 1. uClinux has no fork(2) call, only a vfork(2) call. Therefore, these amount to multiple processes sharing the same address space. In fact, it's very likely that these are multiple threads, not processes at all. 2. The unit is in kilobytes. These processes take up 12 MB, not 12KB. 3. Your questions are probably more appropriate to the uClinux mailing lists. They should, at the very least, be able to more completely answer your queries about the behavior of non-Asterisk system utilities. -- Tilghman -- _ -- 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] Reducing number of Asterisk processes?
On Sunday 30 January 2011 02:28:29 Tilghman Lesher wrote: On Saturday 29 January 2011 04:52:02 Gilles wrote: 2. Provided each process is indeed using 11.990 bytes, is it possible to reduce the number of concurrent processes, considering the fact that this appliance will not handle more than a couple of concurrent calls? 1. uClinux has no fork(2) call, only a vfork(2) call. Therefore, these amount to multiple processes sharing the same address space. In fact, it's very likely that these are multiple threads, not processes at all. 2. The unit is in kilobytes. These processes take up 12 MB, not 12KB. 3. Your questions are probably more appropriate to the uClinux mailing lists. They should, at the very least, be able to more completely answer your queries about the behavior of non-Asterisk system utilities. By the way, you are likely to have trouble running Asterisk on uClinux, anyway. There are a lot of assumptions in the code related to fork(2) creating a separate address space. As this is not true with vfork(2), there are parts of Asterisk that will mysteriously fail. Unless you are comfortable delving into the C code and working on these issues, uClinux will probably never be an appropriate system for you with which to run Asterisk. -- Tilghman -- _ -- 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] CDR issue - Problem logging CDR(userfield) in Master.csv
On Friday 28 January 2011 05:34:21 Athanasia Tsertou wrote:
In my dialplan, right before my Hangup() call, I have put the following
(am using AEL, but I guess this is irrelevant)
Set(JITTER=${CUT(RTPAUDIOQOS,\;,4)});
Set(CDR(userfield)=${CUT(JITTER,\=,2)});
Did you put it in the h extension? That is where it needs to be executed
in order for it to go into the CDR. If it's not in the h extension,
you're updating the CDR after it has already been posted.
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Re: [asterisk-users] Can a duration limit be specified in spool call file?
On Friday 28 January 2011 18:27:15 Bruce B wrote: Hi Everyone, I don't see any parameter for limiting duration of a call in the .call file for Asterisk spool outgoing directory. I'd rather not use a MeetMe to drop the call in a conference room and to then limit the call duration as that complicates things unnecessarily. I am wondering if there is anything else I can do or if the Channel parameter take call duration like the DIAL parameter? No, but you can specify a Local channel as the channel in the call file and then set a TIMEOUT(absolute) for the call, before you Dial() the actual channel you want to use. Keep in mind that the actual channel could be specified by a Set variable in the callfile. -- Tilghman -- _ -- 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] Return variables from func_odbc calls?
On Wednesday 26 January 2011 03:02:19 Sherwood McGowan wrote: This is primarily aimed at Sir Lesher, whose name graces the source code for func_odbc that I'm currently trying to read to answer this question. Tilghman (or anyone else who has determined the answer to this query), I have googled, searched wikis, and I'm currently perusing the source code, but the long and short of it is that I cannot seem to find any reference to variables set by func_odbc calls such as something that would indicate if a query worked so that I can (in the dialplan) handle errors on the fly. Another item I'm trying to determine is the LAST_INSERT_ID... Thoughts/Comments? I hope very much that I haven't overlooked something, but then again I'm no longer a spring chicken either Well, it depends upon what type of query you're performing. If it is a query which inserts/updates, then ODBC_ROWS will contain an integer specifying the number of rows affected. -1 is reserved for a statement which failed, since it is perfectly possible for an UPDATE to succeed, yet affect 0 rows. For SELECT queries, however, that is a much more difficult question, since it depends upon the particular query. Again, it is perfectly possible for a SELECT query to successfully run, yet return 0 rows. Or it might be that with your dataset, you should never get 0 rows returned. These are questions that must be pondered by the particular data administrator, not answers that I can provide as the author of the tool. As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported, since it is not portable across database types. -- Tilghman -- _ -- 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] Return variables from func_odbc calls?
On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote: On 11-01-26 04:56 AM, Tilghman Lesher wrote: As far as LAST_INSERT_ID, that is a MySQL-ism that is not supported, since it is not portable across database types. While, LAST_INSERTID(); is a MySQL-ism, I've been able to use it with func_ODBC. Of cource, my database is MySQL and this function would not work on anything else. [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); That assumes you have only one call in existence at a time. If two calls came in and executed the query at about the same time, it's possible for both reads to return the same value. -- Tilghman -- _ -- 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] Crossover cable for E1 ?
On Monday 24 January 2011 03:46:18 A J Stiles wrote: On Saturday 22 Jan 2011, Tim Panton wrote: I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card. Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable? If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to stock such a thing. It's easy to make an ISDN crossover cable. Cut one end of a standard network cable. Get a new RJ45 plug, rubber boot (not strictly necessary, but makes it look neater) and crimping tool. Push the rubber boot onto the end of the cable first. Strip about 2cm. of outer sheath and separate the inner pairs, then arrange in this order from left to right with the brass contacts uppermost: white/blue blue white/green orange white/orange green white/brown brown This is incorrect. The pairs should be: blue white/blue white/green white/orange orange green white/brown brown Wire 1 MUST swap with 4 and Wire 2 MUST swap with 5. To do as you have shown above switches the polarity on each electrical circuit. It is especially important that you do not switch the polarity, as some equipment does not auto-correct for reversed polarity. -- Tilghman -- _ -- 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 SSH Console Colors Debian Lenny
On Monday 24 January 2011 04:09:31 Olivier wrote: 2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote: On Thursday 20 Jan 2011, JR Richardson wrote: Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. In short: A. Don't re-invent start-stop-daemon. B. Let's just move to upstart/systemd so there won't be a need for this stupid guardian safe asterisk. All these reasons seem fine for me. So the remaining question is how can we still get colors with ssh console ?. Is it compliant with start-stop-daemon, for instance ? Why not just use the start script included with Asterisk? I solved this exact problem a while back, so unless somebody has broken the script since, it should still be working. -- Tilghman -- _ -- 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] FUNC_ODBC and ARRAY
On Saturday 22 January 2011 19:46:16 Sherwood McGowan wrote:
Gentlemen,
I have googled, searched the mailing list archives, and even spoke on
the IRC channel, but have not found an answer to the following
problem. I am attempting to retrieve multiple columns in an ODBC query
using ARRAY per the solutions offered by many individuals. My dialplan
code is as follows:
exten = _.,n,Set(ARRAY(var1,var2,var3)=${ODBC_LOOKUP(${KEYVAL})})
exten = _.,n,Verbose(2,var1 = ${var1})
exten = _.,n,Verbose(2,var2 = ${var2})
exten = _.,n,Verbose(2,var3 = ${var3})
Here's the func_odbc.conf code:
[LOOKUP]
dsn=mysql-asterisk
readsql=SELECT col1, col2, col3 from table1 WHERE keycol = '${ARG1}'
and here's the full log:
[Jan 22 20:12:50] VERBOSE[32348] pbx.c: -- Executing
[123@dolookup:8] Set(SIP/sip1-inbound-0f99,
ARRAY(var1,var2,var3)=96829,-3,Name Unavailabl) in new stack
[Jan 22 20:12:50] DEBUG[32348] func_strings.c: array
(var1,var2,var3=96829) [Jan 22 20:12:50] DEBUG[32348] func_strings.c:
array set value (var1=96829) [Jan 22 20:12:50] DEBUG[32348]
func_strings.c: array set value (var2=(null)) [Jan 22 20:12:50]
DEBUG[32348] func_strings.c: array set value (var3=(null)) [Jan 22
20:12:50] WARNING[32348] pbx.c: MSet: ignoring entry '-3' with no '='
(in +17322761300@getcnam:8
Add:
[compat]
app_set=1.6
to your asterisk.conf and restart.
Basically, someone a long time ago decided that making Set take multiple
key value pairs would be a good idea. This misfeature led to a great deal
of dialplan confusion, with lots of escaping needed to make it work
correctly. This is what I attempted to fix in 1.6, with 1.4 upgraders
getting the old behavior by default (so their dialplans would not break on
the upgrade) and new users getting the new behavior by default. This is
all detailed in UPGRADE.txt, by the way.
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Re: [asterisk-users] Ongoing problem with 1.8
On Tuesday 18 January 2011 01:05:20 Ira wrote: I have tried installing many of the beta versions and most of the release versions of 1.8. I have 3 SIP phones which we use for all our calls. After installing 1.8 the first thing I try is calling out port one of my Digium TDM04 back into port 2. I can see that the call comes in and tries to call all three SIP phones but the phones never ring. Eventually the call goes to voice mail and these error messages pop up. I've read doc/sip-retransmit.txt and as far as I can tell, there's nothing there for me to try. Is there anything else I might try or do to help troubleshoot this. Try running a tcpdump for udp port 5060 while this is occurring. Also, what type of SIP phones are you using? -- Tilghman -- _ -- 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] Ongoing problem with 1.8
On Tuesday 18 January 2011 11:31:07 Ira wrote: At 01:00 AM 1/18/2011, you wrote: On Tuesday 18 January 2011 01:05:20 Ira wrote: I have tried installing many of the beta versions and most of the release versions of 1.8. I have 3 SIP phones which we use for all our calls. After installing 1.8 the first thing I try is calling out port one of my Digium TDM04 back into port 2. I can see that the call comes in and tries to call all three SIP phones but the phones never ring. Eventually the call goes to voice mail and these error messages pop up. I've read doc/sip-retransmit.txt and as far as I can tell, there's nothing there for me to try. Is there anything else I might try or do to help troubleshoot this. Try running a tcpdump for udp port 5060 while this is occurring. Also, what type of SIP phones are you using? Aastra 480i-CT phones. Is tcpdump port 5060 the syntax you'd like me to use? Nope, tcpdump 'udp port 5060'. And I may have neglected to point out, the same system has been running since 1.2.11 or so with basically no issues. While that's a useful data point, it's not relevant to the problem. A significant portion of the SIP stack was re-implemented in 1.8, and Polycom phones are on the desktops of nearly every Asterisk developer. Since you aren't using a Polycom, the SIP stack on that device is implemented differently, causing possible incompatibilities. This is why the tcpdump will be helpful: to figure out what is different and why it doesn't work. -- Tilghman -- _ -- 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] Top Posting
On Sunday 16 January 2011 21:18:54 William Kenworthy wrote: Peoples email clients, work habits and environment mean that people to work the way thats comfortable to them. You want your mails read, you work with them, not get on a soap box and say YOU MUST BOTTOM POST. That was exactly my original point. If the list administrators are the experts, and they say to bottom post, then pissing off the experts is a way to ensure that you get the least help, when asking a question. Follow list etiquette to get the best possible answers. -- Tilghman -- _ -- 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] Top Posting
On Sunday 16 January 2011 20:47:56 James Miller wrote: When you get over 500 emails a day on your blackberry you have make a decision on what is or is not worth reading at that moment. Clearly, then, the problem is your blackberry. Ditch it. Or stop subscribing to list email on a device which is clearly not up to the task. Or would you say that since it's inconvenient for you to clean up your dog poo, you shouldn't have to pick it up? And leave it where the rest of us might step in it? If you cannot be bothered to clean up after your dog, maybe you shouldn't be taking your dog to the park. Similarly, we may not be able to fine you for failing to obey list rules, but the rules still apply to you, like it or not. -- Tilghman -- _ -- 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 4 ports used for a single call?
On Friday 14 January 2011 15:12:29 Bruce B wrote: Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. http://www.justfuckinggoogleit.com/search.pl?query=top+posting It's why most of the experts in here ignore your posts. If you haven't got the good sense to follow etiquette, the Delete key becomes the first line of defense. -- Tilghman -- _ -- 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] environment variable + res_mysql.conf
On Sunday 09 January 2011 23:05:14 Chandrakant Solanki wrote: Hi All. I have export some db parameter in /etc/bashrc as follows ... export DB_NAME=xyz export DB_IP=1x.1x.1x.1x export DB_PWD=dkjfaoi Now, I want use these all environment variable into /etc/asterisk/res_mysql.conf file. Is there any way to do this..?? No, we do not support variable interpolation in that file. You could, however, turn on execincludes in asterisk.conf and execute a command that referred to the environment variables, then output a valid configuration syntax. -- Tilghman -- _ -- 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] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On Wednesday 05 January 2011 07:07:10 Steve Underwood wrote: On 01/05/2011 03:29 PM, Bruce B wrote: Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. The Siren7 and Siren14 codecs in Asterisk are licensed code from Polycom, so they are indeed the Siren7 and Siren14 codecs. They will interoperate with any other vendor who has licensed those codecs from Polycom. -- Tilghman -- _ -- 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] VoIP PoE phones for restaurant (kitchen)
On Wednesday 05 January 2011 06:50:19 Andy Graybeal wrote: I'd definitely look into a phone mounted to the wall that has no actual handset, but merely buttons and a speaker grille. It should probably additionally be stainless steel, as I suspect it will need a good cleaning at least daily. The Polycom phones look great on a desk, but they are not industrial in design. What is this dream phone you speak of? Please help me in located it. I don't want to make a mistake with purchasing the wrong thing. I've never seen such a thing. We've got two noisy kitchens that need to talk back and forth. This is what I first imagined I would find, but I've not found this yet. Top link on Google for stainless steel SIP intercom: http://www.adamtelco.com/valcom-vip-172l-st-stainless-steel-sip-intercom- doorphone.html Cyberdata appears to have another, too: http://www.alloy.com.au/010935.htm Yet another: http://www.zenitel.com/en/Stentofon/Products/Tamper--Vandal-Resistant- Substations/SIP-Vandal-Resistant-Substation/ -- Tilghman -- _ -- 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] Realtime SIP, multiple AX servers question
On Wednesday 05 January 2011 09:39:00 Bryan Field-Elliot wrote: On Jan 4, 2011, at 12:26 PM, Tilghman Lesher wrote: It wasn't designed to do this. While you can have the same sippeers table for multiple servers, you really should have a separate sipregs table for each backend server. The reason why is that some mappings depend implicitly on the host to which it was registered. For example, if a phone is behind a NAT, then the external port is dependent upon the same host responding. If a different host tries to communicate to that external port, some NAT devices will not route the packet properly. This is especially true for SIP over TCP, but it's also true for UDP packets. (Routing packets back through a NAT without verifying the sending IP is a security risk.) Probably more appropriate for your case is to use DUNDi to coordinate your machines as to which server presents holds the registration for any specific phone. We have one table which is serving both purposes (peers and reg). When we want to route a call to an ATA, we first look up that ATA's regserver in that table, and then construct a SIP URI based upon that regserver address. In that way, we route the call through the server to which the ATA is currently registered. So I guess we're covered already in the scenario you describe. It seems like not a great design to have to have a private sipregs table for every server in our pool, especially given that the pool will grow (or maybe shrink) over time. Is that really the recommended design? I haven't seen any articles describing that setup for RealTime in a multi-server environment. Sorry, but a private table for sipregs for each server was exactly what it designed for, in order to separate out values which change per-server from general configuration (same for every server). While I understand that you're presently using a separate lookup into that table, DUNDi is the (scalable!) protocol meant to perform this task for you. Clearly, using a shared sipregs table has its own set of problems; rather than sticking to your flawed configuration, I would think that you would jump at the chance to fix it. -- Tilghman -- _ -- 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] Realtime SIP, multiple AX servers question
On Tuesday 04 January 2011 09:40:56 Bryan Field-Elliot wrote: Thanks Olle. Do you suppose I am the first Asterisk user to discover this behavior? I would find that hard to believe that I'm the first person to notice... It wasn't designed to do this. While you can have the same sippeers table for multiple servers, you really should have a separate sipregs table for each backend server. The reason why is that some mappings depend implicitly on the host to which it was registered. For example, if a phone is behind a NAT, then the external port is dependent upon the same host responding. If a different host tries to communicate to that external port, some NAT devices will not route the packet properly. This is especially true for SIP over TCP, but it's also true for UDP packets. (Routing packets back through a NAT without verifying the sending IP is a security risk.) Probably more appropriate for your case is to use DUNDi to coordinate your machines as to which server presents holds the registration for any specific phone. -- Tilghman -- _ -- 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] VoIP PoE phones for restaurant (kitchen)
On Tuesday 04 January 2011 16:15:54 Andy Graybeal wrote: The Polycom 321 has not been EOL'd and supports VLAN. It is, however, lacking a 2nd ethernet port if you were to go that route. -M Thanks for the response Mark. I see the 331 has two ports and the same features as the 321. I'm wondering what phone would be best being used as an intercom in a busy kitchen. I asked this some months ago; but this time around I'm writing it into this years budget. I see the 335 has HD Voice and the 321 has Clarity by Polycom. Which would be best in a noisy kitchen using the devices speaker phone? Should I seek another device for the kitchen all-together? I'd definitely look into a phone mounted to the wall that has no actual handset, but merely buttons and a speaker grille. It should probably additionally be stainless steel, as I suspect it will need a good cleaning at least daily. The Polycom phones look great on a desk, but they are not industrial in design. -- Tilghman -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Thursday 23 December 2010 09:16:26 Bryant Zimmerman wrote: In the voip-info posting Right here is why you fail. Voip-info is very often wrong. Refer to the documentation that comes with Asterisk for definitive information. In this case, the h extension should be in the calling context, not within the Macro itself. -- Tilghman -- _ -- 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 and Realtime
On Thursday 23 December 2010 12:52:48 CB wrote: Could anyone recommend some documentation regarding Asterisk 1.8 and the realtime architecture? Specifically I want to know if it is possible to set a priority label or to use n as a priority for realtime extensions in Asterisk 1.8? My understanding is that is not possible with Asterisk 1.4 and I wonder if it's changed? It has not changed, and it is unlikely to change. -- Tilghman -- _ -- 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 and Realtime
On Thursday 23 December 2010 17:55:40 Carlos Chavez wrote: On Fri, 2010-12-24 at 07:52 +1300, CB wrote: Could anyone recommend some documentation regarding Asterisk 1.8 and the realtime architecture? Specifically I want to know if it is possible to set a priority label or to use n as a priority for realtime extensions in Asterisk 1.8? My understanding is that is not possible with Asterisk 1.4 and I wonder if it's changed? Realtime extensions for Asterisk is horrible. You need to use extensions.conf and use the switch statement (between other inconvenient things). I really recommend against using that part of the Realtime engine. Actually, you can use the 'overrideswitch' statement, and the named switch will be consulted first for all extension contexts. The solution to this would be to use Realtime static which with a little patience lets you use the dialplan without many modifications and with all the helpers like n and same. This is what we use at the moment for our configuration interface. Also, you should use an interface such as func_odbc to abstract your data away from your logic. Store the logic in the dialplan, but store the dynamic information in the database. -- Tilghman -- _ -- 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 produces *many* zombie processes on Debian.
On Wednesday 22 December 2010 08:23:19 MrHanMan wrote: On Wed, Dec 22, 2010 at 6:41 AM, Steve Davies davies...@gmail.com wrote: On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote: On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. I know what the issue is. Please open a report on https://issues.asterisk.org and I'll get a patch uploaded pronto. Please let us know the issue number once raised - I'd like to follow this one. I happened to see it pop up on the bug tracker. Issue #0018515. Very funny error message in the patch. It's a forward-port of a section of code that was in res_agi in 1.4. It was no longer needed in res_agi because AGIs can now continue to interact with Asterisk after a hangup event, transitioning gracefully into DeadAGI. -- Tilghman -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 11:42:33 Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either 1.6.2
or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may
be ; retrieved inside of of this extension. The default value is no.
endbeforehexten=no
The default is set to no so why can't I store any CDR values in my h
extension.
exp..
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
I need the cause code stored.
Sounds like your h extension is in the wrong context. Try including some
information about where you are putting the h extension and what includes
you're doing.
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Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
On Wednesday 22 December 2010 12:20:36 Olivier wrote: 2010/12/22 Bryant Zimmerman brya...@zktech.com Giorgio You could buy just a couple of licenses 3 to 5. It would get rid of the messages for the most part and it would give you the ability to transcode for voicemails and other items requiring transcode. The reason you are likely getting the messages is there is some kind of transcode required that it can't do and you are getting the warring. If you shut off all in the middle functions like recording, voicemail, and feature codes you may be able to get rid of them but you would also loose the functions. You will likely waste more than the $30 to $50 dollars in time and you get the option to transcode to boot. Just my 2 cents. Mayba I'm hijacking this thread, but what about virtual machines ? At the moment, let say you're using an hardware platform on which you launch virtual machines (one per project but only one at a time). Would a single licence be usable on each virtual machine (same (virtualized?) processor and mac addresses) ? No, because each virtual machine gets its own virtualized Ethernet MAC address. -- Tilghman -- _ -- 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] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 21:08:56 Bryant Zimmerman wrote:
My h extension is in the same context as my Dial commands. Here is a
cut back version of the code.
The cause_code value is never stored in the mysql databae even when set
in the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = h,1,NoOp(Cause Code = ${HANGUPCAUSE})
exten = h,n,Set(CDR(cause_code)=${HANGUPCAUSE})
exten = h,n,Goto(rc-${DIALSTATUS},1)
There's the problem. The h extension should be in whatever context is
calling the Macro, not in the Macro context itself.
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Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were testing this server with 8 SCCP phones, making up to five simultaneous calls through the DAHDI interface (a Digium Wildcard TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) start logging on and we get around 7 or 8 simultaneous DAHDI calls, Asterisk starts producing zombie processes at a high rate. I know what the issue is. Please open a report on https://issues.asterisk.org and I'll get a patch uploaded pronto. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SOLVED: Re: Setting `userfield` from within a callfile
On Tuesday 21 December 2010 04:49:42 A J Stiles wrote:
On Monday 20 Dec 2010, Olivier wrote:
2010/12/20 A J Stiles asterisk_l...@earthshod.co.uk
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial
application (written by someone else before me) which sets up
calls by creating files of
the general form
Channel: SIP/$INSIDE_NUMBER
Context: $CONTEXT
Extension: $OUTSIDE_NUMBER
Priority: 1
CallerId: $INSIDE_NUMBER
in /var/spool/asterisk/outgoing/ .
It works very well. However, it would be nice to be able to attach
an additional piece of information along with the call record
There is a userfield in the SQL database, which is a VARCHAR(255)
and would be plenty for what we need. Is there a way to set the
userfield of the CDR database from within such a callfile?
Yes, adding a Set field in your call file (see
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out), you'll be
able to pass everything you need to your dialplan, and then, from
there, write everything you need to your CDR.
I've got it working now! Thanks Olivier and Tilghman.
Now, for the benefit of anyone who may be searching the archives of this
mailing list at some point in the future, here's what I did.
I have modified the callfile-generating CGI script to added an extra
line to the callfile, something like;
Set: uid=$UID
and made sure that the calls it places are in a context of their own.
In my extensions.conf, I then have as part of that context, a line
ending with the command
... ,Set(CDR(userfield)=${uid})
and it all Just Works Beautifully.
You shouldn't need to. You should be able to just do:
Set: CDR(userfield)=$UID
in your spoolfile and it should work from there.
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Re: [asterisk-users] Setting `userfield` from within a callfile
On Monday 20 December 2010 10:33:33 A J Stiles wrote: Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application (written by someone else before me) which sets up calls by creating files of the general form Channel: SIP/$INSIDE_NUMBER Context: $CONTEXT Extension: $OUTSIDE_NUMBER Priority: 1 CallerId: $INSIDE_NUMBER in /var/spool/asterisk/outgoing/ . It works very well. However, it would be nice to be able to attach an additional piece of information along with the call record There is a userfield in the SQL database, which is a VARCHAR(255) and would be plenty for what we need. Is there a way to set the userfield of the CDR database from within such a callfile? As is stated within sample.call (in the root directory of the Asterisk source): # # You can set channel variables that will be passed to the channel. # This includes writable dialplan functions. To set a writable dialplan # function, the module containing this function *must* be loaded. # #Set: file1=/tmp/to #Set: file2=/tmp/msg #Set: timestamp=20021023104500 #Set: CDR(userfield,r)=42 -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unexpected dialplan match
On Monday 20 December 2010 11:35:21 Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config]
3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]
4. Set(CDR(accountcode)=${accountcode}) [pbx_config]
7. ResetCDR() [pbx_config]
8. ...
You. . is a short-circuit operator; everything after it is ignored.
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Re: [asterisk-users] res_odbc dependeny issue
On Wednesday 15 December 2010 13:53:12 satish patel wrote: I have issue with res_odbc.so module Asterisk 1.8 not allowing me to enable because its depended on generic_odbc and ltdl I did install unixodbc and ltdl but still same error Make sure you re-run ./configure after you add/remove packages, as the configure script is what determines what packages are found. Additionally, ensure you installed the -devel (-dev on Debian/Ubuntu) packages, as it is the headers in these packages which are needed. -- Tilghman -- _ -- 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] Downloading the Asterisk as tar.gz file
On Sunday 26 September 2010 12:22:28 bilal ghayyad wrote: We were using a link before to be able to browse the different asterisk versions and download the needed one as tar.gz file, but really I am not able to find this link again. Anyone can advise me for that link where I can browse the different deliveries (1.2, 1.4, 1.6, 1.8 versions) and select a one to download it as a tar.gz file? Something like subversions. http://www.google.com/search?q=download+asterisk First search result, last section on that page is Older Versions/Direct Access. Not exactly hidden. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Need to pick your brain for recom mendation on using 2.5 or 3.5 HDDs for Asterisk server...
On Sunday 26 September 2010 17:09:40 Hans Witvliet wrote: On Sun, 2010-09-26 at 23:16 +0200, Dmitry Nedospasov wrote: Asterisk TFOT actually has a chapter on this, though it might be a bit outdated [1]. It's Chapter 2, Preparing a System for Asterisk. afaicr, i though that the magnificant book would get un update for the 1.8 release... (or are these plans abandonned ?) No, they are quite active, currently, working on finishing up the book, hopefully for publication just after 1.8 is officially released. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 ODBC Insert issue
On Sunday 26 September 2010 13:08:43 Neeraj Chand wrote:
Hi guys,
Having issues with doing an insert statement using ast 1.4.24:
[START]
dsn=mssql-asterisk
write=INSERT INTO testdb (callarrival,callerid) VALUES
('${VAL1}','${VAL2}')
SET(ODBC_START()${TIMESTAMP},${CALLERID(num)})
I'd say you're missing an equal '=' sign. Without that, nothing is
actually getting executed.
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Re: [asterisk-users] Unexplained message in 1.6.2
On Tuesday 21 September 2010 08:42:04 CDR wrote: Every time I start Asterisk or do a simple reload I see this message: Cannot open maximum file descriptor 32767 at boot? No such file or directory Does anybody have some idea of what can it be? It did not happen in version 1.4. Philip Essentially what this is saying is that you've raised your per-process file descriptor limit higher than your booted kernel will allow in a single process. This should almost never happen. See the value here: bash% sysctl fs.file-max -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting 'fname_base' variable doesn't affect 'automon' result file.
On Monday 20 September 2010 11:01:36 Jose P. Espinal wrote:
Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of
'Monitor' application affect the file name generated through 'automon'
feature?
I initialized this variable with a value as follows:
Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%
H%M%S)})
a. Should I use 'fname_base' in uppercase (FNAME_BASE)? or...
b. Is this variable independent of the 'automon' feature?
Where did you get the idea that fname_base is even a variable for you
to set? It's always been a parameter to the Monitor application.
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Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] 3rd party app store
On Friday 17 September 2010 12:51:16 Dean Collins wrote: I recently came across this email that I wrote in May 2008 .. http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html It's such a shame that Digium manhandled the project away from the community only to then bury it and not allow it to proceed. I really wonder when I look at the Apple iphone development community as to where the 3rd party Asterisk development community could have been if Digium didn't kill this project. It's not buried. You can find the link on asterisk.org, under Applications: http://www.asteriskexchange.com/ -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] 3rd party app store
On Friday 17 September 2010 16:53:58 Dean Collins wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, 17 September 2010 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 3rd party app store On Friday 17 September 2010 12:51:16 Dean Collins wrote: I recently came across this email that I wrote in May 2008 .. http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html It's such a shame that Digium manhandled the project away from the community only to then bury it and not allow it to proceed. I really wonder when I look at the Apple iphone development community as to where the 3rd party Asterisk development community could have been if Digium didn't kill this project. It's not buried. You can find the link on asterisk.org, under Applications: http://www.asteriskexchange.com/ Wow when did that happen? Shockingly, it happened two years ago, at Astricon, shortly after the email that you referenced. It was even part of a keynote address at Astricon. I'm not sure why you weren't aware of this, as a ton of publicity went out surrounding it. Perhaps you've just forgotten that it existed in the interim? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] 3rd party app store
On Friday 17 September 2010 22:52:02 Dean Collins wrote: Tilghman Lesher wrote: On Friday 17 September 2010 16:53:58 Dean Collins wrote: Tilghman Lesher wrote: On Friday 17 September 2010 12:51:16 Dean Collins wrote: I recently came across this email that I wrote in May 2008 .. http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.h tml It's such a shame that Digium manhandled the project away from the community only to then bury it and not allow it to proceed. I really wonder when I look at the Apple iphone development community as to where the 3rd party Asterisk development community could have been if Digium didn't kill this project. It's not buried. You can find the link on asterisk.org, under Applications: http://www.asteriskexchange.com/ Wow when did that happen? Shockingly, it happened two years ago, at Astricon, shortly after the email that you referenced. It was even part of a keynote address at Astricon. I'm not sure why you weren't aware of this, as a ton of publicity went out surrounding it. Perhaps you've just forgotten that it existed in the interim? Any thoughts on why the lack of traffic? No. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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
