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
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
--
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
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 “
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/${EXTEN})
shirley*CLI
On Wednesday 06 April 2011 09:49:17 Jon Farmer wrote:
Hi
I have a call into a MeetMe conference that when I do a core show
channel returns
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x1000 (g722)
ReadFormat: 0x1000 (g722)
Can someone explain what the differences between Native, Wite
On Wednesday 06 April 2011 14:53:00 Hans Witvliet wrote:
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other pitfalls?
This isn't a pitfall that comes with the
On Tuesday 05 April 2011 20:10:48 Bryant Zimmerman wrote:
I have deployed several 1.8.3.2 systems as upgrades of customers systems
and now I am seeing random crashes. For some reason the builds lock up
and stop taking sip connections. Existing calls stay on but when the
user hangs up no new
On Monday 04 April 2011 09:09:28 Asterisk User wrote:
Hello all, I am trying to figure out the logic in on prefix matching for
Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT
calls to 011870, 01137455 and so on.
exten = _011870.,1,Goto(intl-disabled,s,1)
This one is
On Monday 04 April 2011 06:58:23 Arjan Kroon | Mobillion wrote:
Hi,
Does anybody have a solution to this problem?
Because in this issue the solution is not mentioned.
https://issues.asterisk.org/view.php?id=18522
The h extension should be in the context from which the Macro
was called,
On Monday 28 March 2011 14:23:25 Tilghman Lesher wrote:
On Monday 28 March 2011 07:57:10 Eric W. Davenport wrote:
alias start = calldate
alias callerid = clid
These are fine.
I was incorrect, here, and there's nobody to blame but myself. The
field is ACTUALLY named clid internally, so
On Wednesday 30 March 2011 14:34:29 Henrique Fernandes wrote:
exten = s,1,set(CDR(teste)=${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
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)=${CHANNEL(useragent)})
exten=
On Monday 28 March 2011 07:57:10 Eric W. Davenport wrote:
Thanks Tilghman for your response.
I have the following in my cdr_mysql.conf
I put it in sometime yesterday and did not have it till then.
However, it did not make any difference.
Did you reload after making the change to the
On Sunday 27 March 2011 14:50:37 Mohammad Khan wrote:
Here is the dialplan in macro:
exten = s,n,SayNumber($[${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
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
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
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
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
--
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
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.
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
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
--
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
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
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
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
--
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
--
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
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)
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
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
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
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
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
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
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
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
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
--
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
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
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
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
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
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
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
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
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
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
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 {
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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.
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
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
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
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
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.
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
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
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
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
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.
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
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
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
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
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
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
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
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
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
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: $CONTEXT
Extension:
On Monday 20 December 2010 11:35:21 Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
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
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
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
say you're missing an equal '=' sign. Without that, nothing is
actually getting executed.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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
application.
--
Tilghman Lesher
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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--
Tilghman Lesher
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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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
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
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