On 08/02/05 17:35:19, Matt Brooks wrote:
Hey guys,
I am just emailing to inform you guys that a new website has been
created for asterisk.org. You can find the beta site up at
http://beta.asterisk.org. It utilizes the drupal portal framework
and allows users to post pages, news, and comm
On Tuesday 02 August 2005 23:04, Frank Tarczynski wrote:
> I'm running a very recent CVS build under Solaris 10.
>
> For some reason Asterisk reports time as per GMT and not as per
> local time. I have TZ set to US/Eastern but this makes no
> difference.
>
> Any one know where to look for a soluti
On Tuesday 02 August 2005 19:28, Ganbold Tsagaankhuu wrote:
> Hello,
>
> I have few questions about Asterisk.
> I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days
> ago.
This is the developer's list, for questions about development of
Asterisk, itself, not for general user questions.
Forgive me if this has been asked before, I'm new to the list. I was
wondering whether PostgreSQL support would be offered by Asterisk 1.2,
for Asterisk Realtime configuration.
I realise it can be done via ODBC, but this seems a bit clunky
considering that the current Asterisk 1.0 branch alrea
"Herman Webley" <[EMAIL PROTECTED]> wrote:
On 08/02/05 17:35:19, Matt Brooks wrote:
> Hey guys,
>
> I am just emailing to inform you guys that a new website has been
> created for asterisk.org. You can find the beta site up at
> http://beta.asterisk.org. It utilizes the drupal portal framew
I'm running a very recent CVS build under Solaris 10.
For some reason Asterisk reports time as per GMT and not as per local
time. I have TZ set to US/Eastern but this makes no difference.
Any one know where to look for a solution?
Frank
___
Aster
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Reply-To: Asterisk Developers Mailing List
To: Asterisk Developers Mailing List ,
[EMAIL PROTECTED]
Subject: Re: [Ast
Mikael Magnusson wrote:
The Reason header can't be used with only SIP cause values. RFC 3326
defines support for both SIP and Q.850 protocols. I believe Q.850 codes
are used by both PRI and H.323. The Reason protocols are registered by
IANA in http://www.iana.org/assignments/sip-parameters.
RFC
On Tue, Aug 02, 2005 at 05:35:19PM -0500, Matt Brooks wrote:
> I am just emailing to inform you guys that a new website has been
> created for asterisk.org. You can find the beta site up at
> http://beta.asterisk.org. It utilizes the drupal portal framework and
Will this be the end of the wil
Hello,
I have few questions about Asterisk.
I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago.
1.I couldn't find Asterisk version using "asterisk -V" command.
How can I to find version information?
2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on
it.
I
On 08/02/05 17:35:19, Matt Brooks wrote:
Hey guys,
I am just emailing to inform you guys that a new website has been
created for asterisk.org. You can find the beta site up at
http://beta.asterisk.org. It utilizes the drupal portal framework
and allows users to post pages, news, and comm
On Mon, 2005-08-01 at 17:39 +0200, Sergio Serrano wrote:
> Hi Srs.,
> I know that this theme is not first time in this list, but I can't
> never see something enough good for little company. I'm going to start
> to develop a low cost high availability system for asterisk with next
> scenario:
>
Title: Mensaje
Hi
Srs.,
I
know that this theme is not first time in this list, but I can't never
see something enough good for little company. I'm going to start to develop
a low cost high availability system for asterisk with next
scenario:
two machines(one master and other slave
I do something similar using the MYSQL dialplan plugin.
- Original Message -
From: "Doug Logan" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, August 02, 2005 10:46 AM
Subject: [Asterisk-Dev] Route Calls Based on Caller ID
Is there a good way to route calls based on CallerID, in an easily
On Tue, 2005-08-02 at 11:46 -0400, Doug Logan wrote:
> Is there a good way to route calls based on CallerID, in an easily updatable
> format?
> Eg, maybe a mySQL database where you have two fields, "pattern" and
> "extension". If the incoming call meets the pattern, then it is automatically
> s
We have AGI scripts that prompt for DTMF input and they work fine if
the caller dials in.
But if we have asterisk dial out over a Zap channel through the PSTN
and invoke the AGI script, it does not hear any DTMF input.
This is true on all our scripts, both those written in bash amd those
using th
> The bit that stops it compiling on Solaris is the implementation of
> vasprintf()
> in utils.c - as far as I can see, it's only used once, in cli.c
And in res_agi.c
> It didn't want to compile because the varargs stuff seemed to be completely
> wrong for solaris.
Yes. it's very broken in seve
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