it be faster to do a SELECT
first and then conditionally do an INSERT or UPDATE, depending upon the results
of that SELECT? It would seem as if one read and one write operation should
always be faster than two write operations.
- Tilghman Lesher
On July 23, 2014, 5:21 p.m., Michael Young wrote
/#comment22987
It looks like you're adding a 3rd and 4th bound value to the statement, yet
you're not actually binding any values to those placeholders. Don't you need
to do that?
- Tilghman Lesher
On July 16, 2014, 10:44 p.m., Michael Young wrote
On July 2, 2014, 3:14 p.m., Tilghman Lesher wrote:
/trunk/UPGRADE.txt, line 51
https://reviewboard.asterisk.org/r/3698/diff/1/?file=61807#file61807line51
I would think that they could continue to use the MSet application if
they wanted some of the old semantics; however, Set
.
/trunk/res/ael/pval.c
https://reviewboard.asterisk.org/r/3698/#comment22702
There are implications to how the language compilation works that are
satisfied by using MSet. Unless you're going to revisit how AEL works, I'd
suggest keeping this as always running MSet in the future.
- Tilghman
On Sat, Jun 28, 2014 at 5:50 PM, Matthew Jordan mjor...@digium.com wrote:
The following modules are currently deprecated in trunk. This e-mail
is a proposal to remove them from trunk in preparation for Asterisk
13. All of these modules have been deprecated for at least two release
cycles (the
://reviewboard.asterisk.org/r/3386/#comment21012
Fix red while you're here.
/branches/12/funcs/func_dialplan.c
https://reviewboard.asterisk.org/r/3386/#comment21013
Fix red while you're here.
- Tilghman Lesher
On March 25, 2014, 12:18 a.m., Corey Farrell wrote
it looks strange. Perhaps make this a macro of the
form NULLABLE(column), so that it clarifies the intent of the code?
- Tilghman Lesher
On March 17, 2014, 12:17 p.m., zvision wrote:
---
This is an automatically generated e-mail
wanted support for NULL-able columns, we'd want to be able to support ALL
column types, not just integers.
- Tilghman Lesher
On March 13, 2014, 10:06 a.m., zvision wrote:
---
This is an automatically generated e-mail. To reply, visit
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3194/#review10827
---
Ship it!
Good changes.
- Tilghman Lesher
On Feb. 7, 2014
On Dec. 17, 2013, 2:28 a.m., Matt Jordan wrote:
Well, we can't just undo r354657, which is what this patch does. However we
choose to fix the regression, we shouldn't do so in a way that just causes
another regression by re-opening ASTERISK-17121.
To quote Kinsey from
On Tue, Nov 26, 2013 at 1:45 AM, alexander merkulov arhe...@gmail.com wrote:
i am gooing to use libfftw3 library in asterisk application.
compilation is ok, but when i am loading module i get error like this:
dev*CLI module load app_amdb.so
Unable to load module app_amdb.so
Command 'module
you retrieve the
value, instead of specifying a variable into which the name is placed? Seems
like a significant regression in spitting things out to channel variables,
instead of using dialplan functions to retrieve values when needed.
- Tilghman Lesher
On Nov. 20, 2013, 9:12 p.m., Mark
On Nov. 21, 2013, 4:40 p.m., Tilghman Lesher wrote:
Shouldn't something like this be a channel function from which you retrieve
the value, instead of specifying a variable into which the name is placed?
Seems like a significant regression in spitting things out to channel
variables
On Thursday 10 January 2008 08:06:20 Nick Gorham wrote:
Michiel van Baak wrote:
On 11:18, Thu 10 Jan 08, Nick Gorham wrote:
Hi,
The change to use prepare_cb() in odbc_log became broken in r88182.
The code calls SQLBindParameter on a local variable (timestr) but as it
now doesn't call
On Tuesday 11 December 2007 09:38:24 Moises Silva wrote:
AGI is a channel-local interface into an active call, allowing you to do
things in that call like playing files, reading DTMF, setting or getting
variables, etc.
The problem here comes when you want to do all that stuff based on
On Tuesday 11 December 2007 10:16:31 Steven Critchfield wrote:
On Tue, 2007-12-11 at 09:21 -0600, Tilghman Lesher wrote:
Well, if you're addicted to AGI, you're going to love 1.6, since we've
joined the boundary between live and dead mode and allowed an AGI to make
the transition.
You
On Tuesday 04 December 2007 18:21:44 Daniel Hazelbaker wrote:
I have been going through starting to work on a patch to make the
digit and response timeouts all floating point (i.e. millisecond)
aware. Everything is going fine but I am running into a point of
confusion that I would like to get
On Thursday 29 November 2007 11:46:07 Kevin P. Fleming wrote:
SVN commits to the Asterisk project wrote:
Modified: trunk/include/asterisk/lock.h
URL:
http://svn.digium.com/view/asterisk/trunk/include/asterisk/lock.h?view=di
ffrev=90157r1=90156r2=90157
On Tuesday 27 November 2007 11:26:34 Eliel Sardanons wrote:
On 11/27/07, Russell Bryant [EMAIL PROTECTED] wrote:
Eliel Sardanons wrote:
We could start a janitor for creating a 'foo reload' and we could make
de 'module reload *.so' do a module unload; module load
I would rather not
On Thursday 15 November 2007 03:56:26 Rizwan Hisham wrote:
As you can see in the code which i mentioned in my last email that for
decrementing the call-limit value they are actually incrementing it, and
some how it is called reverse decrementing. I need to know What is the
reason for doing
On Tuesday 06 November 2007 09:00:47 Stephen Kratzer wrote:
I'm new here, so be gentle... While it is probably better to disable CDR
using NoCDR() as needed, there might be instances where users would prefer
to disable CDR using NoCDR() and then re-enable it as needed using
ResetCDR() (without
On Saturday 03 November 2007 06:57:56 Victor Sergeev wrote:
That's a great feature!
Does it mean that Digium decided to replace AEL with LUA?
No.
It seems there'll be no sense to use AEL anymore if you can do the same in
real programming language.
Are you saying that users shouldn't have
On Saturday 03 November 2007 01:09:39 Brian Capouch wrote:
Russell Bryant wrote:
Brian Capouch wrote:
. . . it's doing some wy funky things
to the CLI that never used to happen--putting it into reverse video, and
then coloring up some but not all of the output.
On my terminals
On Tuesday 30 October 2007 07:41:58 Andy Davidson wrote:
Asterisk fires some SQL queries (to get all of them, i temporarily
removed the select privs from the asterisk user) based on the code at
the top of this email, akin to :
SELECT * FROM sip_buddies WHERE name = 'customer5'
SELECT * FROM
On Thursday 25 October 2007 08:30:44 Atis Lezdins wrote:
On Wednesday 24 October 2007 17:18:22 Tilghman Lesher wrote:
I added MeetmeList in trunk for this purpose (thus negating the need to
do a corresponding CLI command from the manager).
Nice to hear about that. Will there be also easy
On Thursday 25 October 2007 06:06:37 Tobias Engel wrote:
When there are about 100 parallel connections to asterisk (1.4.11, btw),
all leaving voicemails, the peformance becomes really abysmal:
Connections take over a minute before they are even accepted, etc.
Since this does not happen when
On Thursday 25 October 2007 10:30:20 Kevin P. Fleming wrote:
Since all this stuff is in the global namespace, it should be prefixed
with ast_ so as not to collide with anything defined locally in modules.
Also, I'd suggest prefixing the two callback functions with '__ast_',
signifying that
On Sunday 07 October 2007 14:38, Corrado Santoro wrote:
I'm writing an Asterisk application and, for some reason, I need to
access the same ast_channel from two different threads: one thread
basically executes ast_waitfor_nandfds and the other thread ast_write.
So I would like to ask if the
On Sunday 09 September 2007 06:58:08 Sergio Garcia Murillo wrote:
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com
Sent: Saturday, September 08, 2007 11:46 PM
Subject: Re: [asterisk-dev] AST_FRAME_DIGITAL
On Saturday 08 September 2007 10:28:06 Sergio Garcia Murillo wrote:
In fact the problem is when you want to transmit the data from a
channel to an application. The real case that we have to deal with
is with h234m (i.e. 3g videocalls). The call is received by the
isdn channel (chan_zap in
On Wednesday 29 August 2007 14:00:40 Russell Bryant wrote:
After thinking about it, I decided it would make more sense to attempt to
make a way to use these features for any application, instead of putting
this code in a lot of places. I have a little bit of the code actually
written, but I
On Friday 24 August 2007 11:43:08 Tony Mountifield wrote:
In the previous version of Mantis on the bug tracker it was possible to
click on the link for an attached file such as a patch, and for it to
be displayed in the same browser window as a text file.
Now that doesn't seem possible - all
On Monday 20 August 2007, Matthew Rubenstein wrote:
Is there a way for voice media clients (like SIP phones and POTS/PSTN
phones) that connect their call legs to Asterisk to negotiate a common
codec that they both use at their end, so Asterisk doesn't have to
transcode?
No.
--
On Tuesday 31 July 2007 08:13, Tzafrir Cohen wrote:
On Tue, Jul 31, 2007 at 07:34:54AM -0500, Tilghman Lesher wrote:
On Tuesday 31 July 2007, Tzafrir Cohen wrote:
Asterisk is really the application menuselect was designed for.
However, is there really a point for the common user
On Wednesday 25 July 2007, Russell Bryant wrote:
John Todd wrote:
I would agree with Tilghman here. There are already quite a few
examples of deprecated applications turning into functions, which
tend to break older dialplans already.
Well, all of those examples are done over two
On Monday 23 July 2007, Russell Bryant wrote:
SVN commits to the Digium repositories wrote:
Author: tilghman
Date: Mon Jul 23 14:51:41 2007
New Revision: 76703
URL: http://svn.digium.com/view/asterisk?view=revrev=76703
Log:
Merge the dialplan_aesthetics branch. Most of this patch
On Thursday 05 July 2007 08:56, Senad Jordanovic wrote:
Alex B. wrote:
2007/7/5, Martin Vít [EMAIL PROTECTED]:
Martin Schrott - thinking:systems wrote:
you are right, acting now is not needed, when callbacklogin will
be removed anywhere in future... But thinking how to realice
On Thursday 05 July 2007 12:11, Wolfgang Alper wrote:
I just saw that the filed bug request was closed without fursther
actions and think that this is wrong.
It seems that Asterisk does not allow a nesting of contextes with a
stacksize defined in AST_PBX_MAX_STACK (defaults to 128)
However
On Thursday 05 July 2007 16:31, Steve Murphy wrote:
On Tue, 2007-07-03 at 16:38 -0500, Tilghman Lesher wrote:
The core is adequate at this point. What needs to happen is for
the backends (finally) to take advantage of the CDR core features
that have been there since 1.2 was released
On Wednesday 27 June 2007 07:57, Russell Bryant wrote:
SVN commits to the Digium repositories wrote:
+ /* Create the directory if it does not exist. */
+ dir = ast_strdupa(tmp);
+ if ((file = strrchr(dir, '/')))
+ *file++ = '\0';
The previous two lines can be simplified
On Thursday 21 June 2007, ZhangZQ wrote:
Thank you, can you tell me how, or can you give me more information.
See app_skel.c in the apps/ directory for a sample.
--
Tilghman
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On Wednesday 20 June 2007, Matt Riddell wrote:
What's up with the conversions for delimiter from | to ,?
Is this something we should pay attention to or just in documentation?
I assume you're talking about my dialplan_aesthetics branch?
I'm going through and revamping the dialplan to reduce
On Tuesday 19 June 2007 09:02, [EMAIL PROTECTED] wrote:
I'd like your opinion about increasing the AST_PBX_MAX_STACK value,
currently set to 128.
Although it is clearly enough in the 99% of the scenarios, sometimes
it could be a limit. So I am curious about how increasing to 256 or
512, or
On Saturday 09 June 2007, Pari Nannapaneni wrote:
getcontexts(*new) - returns only the list of contexts in a config file -
and NOT their contents like: action=getcontextsfilename=somefile.conf
should return just the list of all context names [context-1]
[context-2]
getcontext(*new) -
On Saturday 09 June 2007, Caio Begotti wrote:
On 09/06/2007, at 16:57, Tilghman Lesher wrote:
Wouldn't adding these commands defeat the intent of the http server in
Asterisk? The intent, as I understand it, is to be lightweight, to
shift all
of the complexity of the configuration
On Friday 01 June 2007 14:05, Russell Bryant wrote:
This bug report was originally reported on this list. The module
unload callback is not called for modules on Asterisk shutdown. I
consider this bug, but I would like a bit of feedback on where it is
appropriate to make this change.
I
On Thursday 10 May 2007 14:35, Andrew Kohlsmith wrote:
On Thursday 10 May 2007 2:26 pm, Olle E Johansson wrote:
chan_pan :-)
PAN/nokia
PAN/peter
PAN = Personal Area Network
Eww... this isn't using the PAN profile at all, so I don't think
that'd be right... chan_HFP or HSP would be
On Sunday 06 May 2007, Christopher Aloi wrote:
On 5/5/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Saturday 05 May 2007, Sergey Okhapkin wrote:
Yes, it's the expected IP stack behavior when the service is bound to
0.0.0.0. Asterisk sends the repy to the address from which
On Saturday 05 May 2007, Sergey Okhapkin wrote:
Yes, it's the expected IP stack behavior when the service is bound to
0.0.0.0. Asterisk sends the repy to the address from which the request
came, it has no control which src address to use.
Actually, it does control it; it uses the Linux routing
On Friday 06 April 2007, Yuan Qin wrote:
The ast_log() will lock a mutex if appropriate, but the mutex may be in
locked state already.
Maybe we should use pthread_atfork() instead of fork() or never call some
functions that hold mutex
before execv() in child process.
Is there something that
On Wednesday 28 March 2007 11:08, Vadim Lebedev wrote:
I hope it can be integrated in mainline
Why not just use the SetMusicOnHold application in the dialplan?
--
Tilghman
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On Thursday 29 March 2007 18:27, Nicholas Campion wrote:
Would it be possible to include the capability for channels to put
out channel specific information? For example, it would be really
nice if the SIP channel would put out the $RTPAUDIOQOS as you
currently have to store this information
On Tuesday 27 March 2007 16:06, Luigi Rizzo wrote:
Googling a bit, it seems that you can get the modification date of
a database table with a SHOW TABLE STATUS query, and this would
at least help to know if the cache is in sync.
That does not appear to be portable. Postgres, for example, has
On Monday 19 March 2007 10:12, Philipp Kempgen wrote:
Tilghman Lesher wrote:
On Monday 19 March 2007 01:43, Philipp Kempgen wrote:
; Tested
[snom-check-cfg]
Event=check-sync;reboot=false
Content-Length=0
but it makes my Snom 360 reboot. It works as expected (get
config without
On Monday 19 March 2007 09:28, Philipp Kempgen wrote:
Leif Madsen wrote:
On Tuesday 13 March 2007 13:25:11 Steve Murphy wrote:
On Mon, 2007-03-12 at 14:33 -0600, Tilghman Lesher wrote:
exten = s,n,Set(ODBC_TEST3(foo)=123\,\\456\,789\\\,012)
What I miss most is real strings as you know
On Friday 09 March 2007 13:58, Kevin P. Fleming wrote:
Admin DeryTelecom wrote:
I am talking about a SIP Dial and not a Zap Dial, it seems that
this option is available on Zap Dial.
Then this will be the responsibility of the SIP endpoint, not
Asterisk.
It cannot. It is a violation of
On Sunday 04 March 2007, Anthony Lamantia wrote:
In this case, we took the action to document that it was fixed and told
users they should upgrade (and why), because I don't believe this
particular issue was reported by an auditing company
it would have been nice to know a problem existed in
On Tuesday 06 February 2007 16:42, Roger Schreiter wrote:
First of all, I would like to know, whether my attempt
is the right way:
I would either introduce a new frame type for IAX, e.g.
#define AST_FRAME_RAW 11 /* digital raw data */
When asterisk does link an ISDN data call channel to an
On Tuesday 06 February 2007 18:18, Roger Schreiter wrote:
Tilghman Lesher schrieb:
...
Seems like a cheap hack to get around the fact that you have no
idea what you're sending across the wire. I'd prefer some decoding
on the
yes, that's right, I'm looking for a mean to transport
any
On Tuesday 06 February 2007 19:09, Roger Schreiter wrote:
Tilghman Lesher schrieb:
...
And I'm saying that you shouldn't. IAX (and any other intelligent
Ok, what are the alternatives?
Trying to figure out, what type of data both parties are
sending over the ISDN+IAX line?
Correct
On Saturday 27 January 2007 06:37, Gregory Boehnlein wrote:
We began experiencing an issue yesterday whereby one of our TDM
gateways, which handles IAX2 - Zap handoffs was dropping calls. The box
in question has 18 IAX2 peers defined, never has more than about 30
concurrent calls and
On Saturday 27 January 2007 20:17, Sean Bright wrote:
I was building 1.4 earlier today and ran into a problem with
./configure --with-tds=/usr/local. I resolved it by changing line 787
of configure.ac from:
case `grep TDS_VERSION_NO ${FREETDS_DIR:-/usr/include}/tdsver.h` in
to:
On Tuesday 23 January 2007 05:27, SF Markus Elfring wrote:
You may abort the current call, which should free memory for other calls.
Are you sure that enough space will become available for further processing
of a retry?
No, but other threads should observe the same principle. If they are
On Monday 22 January 2007 04:45, Oded Arbel wrote:
From what I understand, the uniqueid field in the ast_channel is
unique for a single installation of Asterisk, and is composed of the
unix timestamp at the creation of the channel plus some running
counter.
The problem I'm having, is that I
On Monday 22 January 2007 16:10, SF Markus Elfring wrote:
The bug report add checks for calloc calls
(http://bugs.digium.com/view.php?id=8295) results in a opportunity to
think again about the used approaches for error handling. Your
current policy seems to be wrong.
Fatal program conditions
On Friday 12 January 2007 02:29, Ian Hailey wrote:
Tilghman Lesher wrote:
On Thursday 11 January 2007 04:25, Ian Hailey wrote:
I have created a new codec (i.e. codec_xxx.so) for Asterisk and would
like to know if I need to also create a new format module (i.e.
format_xxx.so) supporting
On Thursday 28 December 2006 03:33, Tony Mountifield wrote:
Tilghman Lesher [EMAIL PROTECTED] wrote:
On Wednesday 27 December 2006 07:46, SF Markus Elfring wrote:
Some of my suggestions are also in the waiting queue that can be seen
in the bug/feature request tracker
On Thursday 28 December 2006 07:14, Jonathan k. Creasy wrote:
Tony Mountifield wrote:
Tilghman Lesher wrote:
On Wednesday 27 December 2006 07:46, SF Markus Elfring wrote:
Some of my suggestions are also in the waiting queue that can be
seen in the bug/feature request tracker
On Tuesday 12 December 2006 18:54, Song Zhi Feng wrote:
I downloaded the app_conference code via:
http://www.eflo.net/files/VD_app_conference_0.6.zip, and modified
app_conference.c to add include file module.h (module.h is from Asterisk
1.2 and copied to the app_conference directory. The
On Saturday 02 December 2006 23:03, Peter Beckman wrote:
On Sat, 2 Dec 2006, Tilghman Lesher wrote:
On Saturday 02 December 2006 15:16, Peter Beckman wrote:
Is there a formal definitions of the verbosity levels and what is
appropriate at each level?
There is not, but generally you
On Sunday 03 December 2006 16:51, Tim Panton wrote:
But Kevin's point still applies, IAX doesn't support multiple instances of
the same IE type in a message.
Is that by spec or by implementation? If by spec, then my IAX2 vars
implementation was a big Oops on my part.
--
Tilghman
On Sunday 03 December 2006 22:43, Lucas Barbuto wrote:
Hi all,
I'm interested in adding withdrawal codes or not ready reasons to
the Asterisk™ ACD system. That is, when an agent elects to go on
pause, they can/must first select a reason for that pause.
I was able to make a few small
On Friday 17 November 2006 17:55, Steve Murphy wrote:
Defaults are only a good idea when there is a really obvious
choice that has almost no chance of being the wrong thing in
practice. Offhand, I can't think of anything that would meet
that criteria where CUT
On Wednesday 15 November 2006 12:32, Luigi Rizzo wrote:
I think they make this part of the code a lot more readable and
consistent and less error prone, especially because we can do the
same parsing on fields of the same type.
While it certainly makes the code smaller, I don't find it
On Thursday 09 November 2006 06:10, Tony Mountifield wrote:
Is it possible from within a function to get a backtrace of where it was
called from? I would do this if I determined I was closing an unexpected
fd. Or perhaps I just do a crash and use gdb on the core dump.
Unfortunately, people are
On Thursday 09 November 2006 05:09, Ale wrote:
And can asterisk send CPC signal with wctdm driver on a TDM400 over FXO?
Any hint is appreciate.
Have you tried passing the driver parameter opermode=ITALY while loading
the wctdm driver?
--
Tilghman
On Monday 06 November 2006 09:12, Moises Silva wrote:
Well, now knowing the fundamental cause, im going to the hardest
part, finding a solution for the issue on app_conference code, I dont
think is a good idea rely on 20ms frames. Or is it possible to make
Asterisk use only 20ms frames?
On Monday 06 November 2006 12:36, Dan Austin wrote:
'Fixing' app_conference is a worthy endevour, but convincing
chan_iax to honor framing limits on both the send and receive
legs of a channel would be a big win. Even better would be
the addition of an IE to convey the desired framing/payload
On Thursday 02 November 2006 10:09, Ramu Yadav wrote:
All of sudden asterisk disappeared(running asterisk process died) and
it didn't generate the core. We are very sure that we started
asterisk with -g option.
You also need to run 'ulimit -c unlimited' prior to running asterisk.
Asking
On Sunday 22 October 2006 15:08, Roy Sigurd Karlsbakk wrote:
Well, technically, it is a new feature. I'm pretty sure that Trunk is
in
feature freeze mode.
No, there is a branch for Asterisk 1.4 already, so trunk is open
for all kinds of changes.
This patch was submitted half a year
On Monday 09 October 2006 23:00, Austin Seipp wrote:
Greetings! For a while now, I've had a very large interest in telephony
systems in general, and seeing Asterisk I was immediately drawn to it. I
got myself through Asterisk: The Future of Telephony (what a quest, I
know), and I was wondering
On Thursday 05 October 2006 04:30, Brian Candler wrote:
On Thu, Oct 05, 2006 at 03:47:22AM -0400, Jeremy McNamara wrote:
I suggest you purchase the Asterisk book. You seem to be missing
quite a few major core concepts of how Asterisk functions.
http://www.oreilly.com/catalog/asterisk/
On Tuesday 03 October 2006 20:12, Matt Florell wrote:
As for a SVN patch, the code masters won't even look at it til you
post a SVN-development patch although I personally feel that adding
it to the 1.2.X trunk should be allowed since this does not alter
the functions that are in 1.2 and it
On Sunday 24 September 2006 12:58, Jay Hoover wrote:
Thanks, that makes sense. One thing that I don't understand is what
situations in normal Asterisk operation would cause a SIGHUP to get
sent to the daemon. I'm getting a lot of these deadlocks, and I'm
suspicious that there is a problem
On Wednesday 23 August 2006 16:58, Race Vanderdecken wrote:
While there have been fixes to app_voicemail lately that help; one of
the things you will need to watch out for is the mysterious
lost/extra .txt file that gets left behind in
/var/spool/asterisk/voicemail/../INBOX sometimes on larger
On Saturday 12 August 2006 10:38, Tzafrir Cohen wrote:
While I realize that readline support is not getting into asterisk
any time soon...
What do you mean, not anytime soon? It was ALREADY in Asterisk
previously and was removed due to licensing concerns. The current
editline code is in there
On Saturday 12 August 2006 11:48, Thomas Andrews wrote:
On Sat, Aug 12, 2006 at 11:25:24AM -0500, Kevin P. Fleming wrote:
- Thomas Andrews [EMAIL PROTECTED] wrote:
Do you really think so? I would have thought that the socket on
the odbc interface would block if there was a lock on the
On Friday 11 August 2006 12:27, Andrew Kohlsmith wrote:
Honestly what is wrong with
exten = foo,n,IAXVAR(CHANNELTYPE)
exten = foo,n,IAXVAR(DNID)
exten = foo,n,IAXVAR(RDNIS)
exten = foo,n,Dial(IAX2/box-b/${DESTINATION})
Just nitpicking, but this should be:
Set(IAXVAR(CHANNELTYPE)=SIP)
On Friday 11 August 2006 13:40, Stefan Gofferje wrote:
before opening a bug I would like to make sure, this is an issue.
I am trying to Set(_CALLINGTON=...) before making an outgoing call
via an HFC-S Bri card. I have booked CLIP no screening and therefore
can set an arbitrary CLID.
On Tuesday 08 August 2006 13:29, Peter Beckman wrote:
Noticed that Asterisk doesn't use libwrap. Any reason? Could it be
added? It would be handy.
The fact that Asterisk does not have external dependencies now is
entirely by design. Unless there's a convincing argument put through
to start,
On Tuesday 08 August 2006 16:15, Peter Beckman wrote:
In a completely somewhat-related question, is this lack of
dependence on external libraries, other than the list below, the
reason why Asterisk does not use the seemingly common ./configure
scripts?
It does in trunk.
--
Tilghman
On Wednesday 02 August 2006 11:56, Christian Richter wrote:
I'm far from an expert with SVN so I have to admit I'm mightily
confused about all the different branches of Asterisk. Is there a
place where this is explained?
1.2 = stable
trunk = highly experimental with new features
1.4 will
On Wednesday 26 July 2006 14:09, John Martin wrote:
In trunk of app_queue.c since r35504(22nd June):
Have I got the purpose of eventwhencalled wrong or should the second
strcasecmp in the following code be a !strcasecmp...
} else if (!strcasecmp(param, eventwhencalled)) {
On Thursday 20 July 2006 18:15, Josh McAllister wrote:
Out of immediate neccessity I hacked together a new AGI command to
stream a file in the background (exec Background doesn't seem to
work... Does not return immediately). This has been briefly tested,
and it does indeed work. Usage:
The
On Tuesday 18 July 2006 01:29, Sergio García Murillo wrote:
John Martin wrote:
Hi Devs,
I was just watching the SVN commits going by and I was
wondering why the H.263 buffer size in format_h263.c had to be
increased to 32kB?
I really don't see the point if it either (perhaps someone
On Thursday 06 July 2006 10:45, Joseph Benden wrote:
Hello,
I am working on the Solaris drivers and have a conflict with the way
that Solaris works verses Linux.
My question is:
How many people use a mix-and-match of different cards in their
systems?
I for instance use a single port t1
On Saturday 27 May 2006 09:51, Denis Smirnov wrote:
What about issue #6725?
Given that your patches are for oej's branch, specifically, you
need to talk to oej.
--
Tilghman
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asterisk-dev
On Thursday 27 April 2006 08:13, Ken Ouellette wrote:
Instead of a complex query, my thought was adding a column to the
required database schema that would contain the dialpad equivalent
of the last three letters of a user's last name. This column could
be indexed and accessed via a simple
On Sunday 23 April 2006 18:46, Peter Beckman wrote:
On Sat, 22 Apr 2006, Tilghman Lesher wrote:
It must add a lot of code to parse out all of that stuff, and
having an inconsistent method of calling functions make both
documentation confusing and the user/admin confused. It makes
On Friday 21 April 2006 06:09, Julian Lyndon-Smith wrote:
FWIW, I would say that unknown is the best.
Actually, that should be Unknown , since the number field
should be empty.
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Tilghman
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