This are load_module()'s return values, I am not changing nothing, this is
just the way "it should be", read chan_sip.c and you will notice how
'return AST_MODULE_LOAD_SUCCESS' is being used.
But if you take a look at some modules (chan_gtalk.c), they don't use this
return values defined in the e
On Thu, Oct 25, 2007 at 02:09:02PM -0300, Eliel Sardanons wrote:
> I think we could start a new janitor project to replace every 'return' on
> load_module's with:
>
> enum ast_module_load_result {
> AST_MODULE_LOAD_SUCCESS = 0,/*!< Module loaded and configured */
> AST_MODULE_L
I think we could start a new janitor project to replace every 'return' on
load_module's with:
enum ast_module_load_result {
AST_MODULE_LOAD_SUCCESS = 0,/*!< Module loaded and configured */
AST_MODULE_LOAD_DECLINE = 1,/*!< Module is not configured */
AST_MODULE_LOAD_
The CentPBX project is looking for a few good developers to help out with
the project. It's complete Open Source and we're looking for some
volunteers to help us build and grow the online community and distribution.
Please contact me @ [EMAIL PROTECTED] if you would like to get
involved J
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 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 wh
Patrick Markert - Amerisave wrote:
> I have heard that they have been problems with Realtime Queus in
> Asterisk. What are the problems? Do they still exist? Do they exist
> in the most current version of 1.2? Has anyone been able to use this
> technology reliably in a high call volume envir
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 e
SVN commits to the Asterisk project wrote:
> Added: team/mmichelson/forward-loop/include/asterisk/global_datastores.h
> URL:
> http://svn.digium.com/view/asterisk/team/mmichelson/forward-loop/include/asterisk/global_datastores.h?view=auto&rev=87017
> ==
Sounds like reinventing the wheel to me.
- Original Message -
From: Zoltan Gaspar
To: asterisk-dev@lists.digium.com
Sent: Wednesday, October 24, 2007 1:17 PM
Subject: [asterisk-dev] Asterisk AGI 2.0 - with doc
ASTERISK AGI 2.0
This document presumes basic knowledge of asterisk dial pla
On Wednesday 24 October 2007 17:18:22 Tilghman Lesher wrote:
> On Wednesday 24 October 2007 08:58:02 Atis Lezdins wrote:
> > On Wednesday 24 October 2007 16:37:25 you wrote:
> > > On Wednesday 24 October 2007 08:22:37 Atis Lezdins wrote:
> > > > On Wednesday 24 October 2007 15:54:36 Tilghman Lesher
I have heard that they have been problems with Realtime Queus in
Asterisk. What are the problems? Do they still exist? Do they exist
in the most current version of 1.2? Has anyone been able to use this
technology reliably in a high call volume environment (Over 1000 calls
per day)?
Hi
Could someone tell me what's the current asterisk policy for network
monitoring? Last iax draft states:
"The IAX2 protocol has various tools to determine the network load. It
uses the POKE message to monitor reachability of remote peer and the
LAGRQ message to measure the quality of a _CURRENT
Hi,
I am currently evaluating the performance of voicemail when using ODBC
voicemail storage and ODBC realtime voicemail config.
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 b
I am afraid that we're going to sacrifice the "multiprotocol" aspect
of Asterisk
if we put DNS support in the dial plan. You have to be able to say
"call this URI, use these SIP servers for the call, time out a
transaction with this time"
Which is a very SIP-specific thing to do. We will have
> "JT" == John Todd <[EMAIL PROTECTED]> writes:
JT> Here's why I ask: I've had first-hand experience with
JT> prioritized/weighted SRV records that cause serious problems.
JT> Someone puts "10 10 _sip._udp.inside-proxy.foo.com" as their first
JT> SRV record for foo.com, and "20 20
JT> _sip._ud
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