[asterisk-dev] Re: Installation and Compilation (Jared Smith)
bilal ghayyad
bilmar_gh at yahoo.com
Fri Jan 12 11:39:44 MST 2007
Dear Jared;
Who create the Makefile? Is it coming after I do the make command or it will be existed after extracting the tar.gz files?
Regards
Bilal
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Subject: asterisk-dev Digest, Vol 30, Issue 24
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Today's Topics:
1. Re: Queue Problem losing variables value on calls waiting
(Andrea Cristofanini -- [Gedam Europe])
2. Installation and Compilation (bilal ghayyad)
3. Out-of-band DTMF (Nitin Ahuja)
4. Re: Installation and Compilation (Tzafrir Cohen)
5. Re: Installation and Compilation (Jared Smith)
6. Re: rate_engine (serva)
7. Re: [svn-commits] file: trunk r1807 - /trunk/zaptel.c
(Russell Bryant)
8. Re: [svn-commits] kpfleming: trunk r50538 -
/trunk/main/channel.c (John Todd)
----------------------------------------------------------------------
Message: 1
Date: Thu, 11 Jan 2007 21:41:49 +0100
From: "Andrea Cristofanini -- [Gedam Europe]" <andrea at gedameurope.com>
Subject: Re: [asterisk-dev] Queue Problem losing variables value on
calls waiting
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <45A6A10D.4000500 at gedameurope.com>
Content-Type: text/plain; charset=ISO-8859-1
Hi list
i have seen this problem :
i use a dialer for outbound callcenter.
I set callerid(name) to recordid for some reason.
When the calls arrive in queue directly to an agent all is sweet
when a call have to wait because the agents are busy, and then is sent
to a agent again it loos the callerid(name) and set it to <unknown>
PLS help!!
--
Cheers Andrea
Andrea Cristofanini
Gedam Europe Srl
Gedam Advanced Communication Ltd
Torino, Italy
C.so Re Umberto 21
Mobile : + 39 329 1871756
PSTN : + 39 011 19824516
FreeVoip: 6838601
http://www.gedameurope.com
http://freevoip.gedameurope.com
------------------------------
Message: 2
Date: Thu, 11 Jan 2007 14:04:07 -0800 (PST)
From: bilal ghayyad <bilmar_gh at yahoo.com>
Subject: [asterisk-dev] Installation and Compilation
To: asterisk-dev at lists.digium.com
Message-ID: <20070111220407.55372.qmail at web54601.mail.yahoo.com>
Content-Type: text/plain; charset=ascii
Hi List;
To create the symbolic link, I read in the documenation that I have to type this command:
# ln -s /usr/src/'uname -r' /usr/src/linux-2.4
1) What it means by 'uname -r'?
2) Why I have to create such symbolic link to do pointing for the kernel? For what exctly will be used with asterisk?
3) What is the relation between creating such symbolic link and build directory?
Any advise.
Regards
Bilal
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------------------------------
Message: 3
Date: Thu, 11 Jan 2007 14:54:54 -0800
From: "Nitin Ahuja" <Nitin at ingenio.com>
Subject: [asterisk-dev] Out-of-band DTMF
To: <asterisk-dev at lists.digium.com>
Message-ID:
<F825F1BE7B552F4893E87AEF8F9F7D530258AAF2 at KEENX03.keencorp.keen.com>
Content-Type: text/plain; charset="us-ascii"
Hi,
I have a 3PCC (b2bua) that accepts calls from an Asterisk and sends them
out to a proprietary mediaserver. The b2bua keeps itself in the SIP
signaling path and needs to "listen" to the DTMF coming out of the
Asterisk.
I have configured the Asterisk with dtmfmode=info but the problem is
that the mediaserver also needs to know about the DTMF and it cannot
handle INFOs.
As far as I could tell there is no way to do both rfc2833 and INFO for
dtmf?
Is it possible to do it via an app or AGI?
If I wanted to add a new dtmfmode to do both what should I be looking
at? Chan_sip.c or something else
Thanks
-n
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Message: 4
Date: Fri, 12 Jan 2007 01:29:09 +0200
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-dev] Installation and Compilation
To: asterisk-dev at lists.digium.com
Message-ID: <20070111232908.GS31960 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
First off: this is a is a questions for asterisk-users. Answering to
there.
On Thu, Jan 11, 2007 at 02:04:07PM -0800, bilal ghayyad wrote:
> Hi List;
>
> To create the symbolic link, I read in the documenation that I have to type this command:
>
> # ln -s /usr/src/'uname -r' /usr/src/linux-2.4
>
> 1) What it means by 'uname -r'?
`uname -r`
The output of the command uname -r
You kernel revision. Modules get loaded from /lib/modules/`uname -r`
> 2) Why I have to create such symbolic link to do pointing for the kernel? For what exctly will be used with asterisk?
Actually those intructions are probably obsolete. What distribution do
you use? Have you built your own kernel?
What is the output of:
ls -l /lib/modules/`uname -r`/build
ls -l /lib/modules/`uname -r`/build/
Chances are you don't need that link.
> 3) What is the relation between creating such symbolic link and build directory?
Check zaptel/Makefile: the build link is used. If it does not exist, the
/usr/src dirs are used.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir at jabber.org
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 5
Date: Thu, 11 Jan 2007 19:38:24 -0500
From: "Jared Smith" <jaredsmith at jaredsmith.net>
Subject: Re: [asterisk-dev] Installation and Compilation
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Message-ID:
<f8a99b4f0701111638n1402e4e9ua6a9e656c54a473a at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 1/11/07, bilal ghayyad <bilmar_gh at yahoo.com> wrote:
> To create the symbolic link, I read in the documenation that I have to type this command:
>
> # ln -s /usr/src/'uname -r' /usr/src/linux-2.4
>
> 1) What it means by 'uname -r'?
You've got it slightly wrong... it's `uname -r` (with back-ticks, not
single quotes). The backticks cause the system to run "uname -r" and
place the output there. The "uname -r" command returns the version
number of the currently running kernel.
> 2) Why I have to create such symbolic link to do pointing for the kernel? For what exctly will be used with asterisk?
Asterisk doesn't need it, but the Zaptel kernel drivers need to know
where the kernel sources are, at least for the Linux 2.4 kernel.
-Jared
------------------------------
Message: 6
Date: Fri, 12 Jan 2007 09:15:49 +0800
From: "serva" <serva at yeah.net>
Subject: Re: [asterisk-dev] rate_engine
To: "Asterisk Developers Mailing List"
<asterisk-dev at lists.digium.com>, "Asterisk Developers Mailing List"
<asterisk-dev at lists.digium.com>
Message-ID: <45A6E150.00538E.21350>
Content-Type: text/plain; charset="gb2312"
HI,Ian Esper£¡
You can define a AST_MODULE yourself that you can compile asteisk-1.4.Like
#define AST_MODULE "ael".
======= 2007-01-11 23:44:02 FROM YOUR LAST LETTER£º=======
>Hi,
>
>I'm trying to port rate_engine to aserisk-1.4 and I getting this error
>
>error: 'AST_MODULE' undeclared here (not in a function)
>
>I'm compiling it external. with it's own Makefile.
>
>What should I do?
>--
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¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡ÖÂ
Àñ£¡
¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡serva
¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡serva at yeah.net
¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡2007-01-12
------------------------------
Message: 7
Date: Thu, 11 Jan 2007 22:35:19 -0500
From: Russell Bryant <russell at digium.com>
Subject: [asterisk-dev] Re: [svn-commits] file: trunk r1807 -
/trunk/zaptel.c
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID: <45A701F7.9010608 at digium.com>
Content-Type: text/plain; charset="utf-8"
svn-commits at lists.digium.com wrote:
> Author: file
> Date: Thu Jan 11 11:07:38 2007
> New Revision: 1807
>
> URL: http://svn.digium.com/view/zaptel?view=rev&rev=1807
> Log:
> Return what I took away (fcstab) since it's actually required for some instances... (issue #8792 reported by tootai)
If we make the zaptel configure script check for it to make sure that it is
supported, we can use the "unused" attribute on this to eliminate the compiler
warning for the cases where it isn't needed.
static __u16 fcstab[256] __attribute__ ((unused)) =
...
--
Russell Bryant
Software Engineer
Digium, Inc.
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Message: 8
Date: Thu, 11 Jan 2007 22:39:47 -0800
From: John Todd <jtodd at loligo.com>
Subject: [asterisk-dev] Re: [svn-commits] kpfleming: trunk r50538 -
/trunk/main/channel.c
To: asterisk-dev at lists.digium.com
Message-ID: <p0624052ac1ccdbd08094@[192.168.210.110]>
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While I understand the sentiment here, I'm not sure this is a good
idea. This builds in a 500ms post-dial delay issue into every call.
I've been building systems for three years now, and everywhere there
is an "Answer" (which, I believe, should be the only method that
picks up a line and sets up a media channel locally, but that's a
discussion for another thread) there is a "Wait(.5)" or even a
"Wait(1)".
Building this in as a non-optional default seems a bit on the drastic
side to deal with people who don't know about how to manage their
dialplans. Perhaps making this a selectable option, or removing it
and better educating folks how to write a dialplan that has better
audio performance?
There are instances where instant audio path access is useful. If
I'm using DTMF to page through ChanSpy sessions, as an example, is
this called? How about during large test jig configurations where
thousands of audio channels are being set up/torn down in short
order? (which, BTW, will be happening next week on a grand scale
hopefully.)
It just seems "wrong" to me to insert a mandatory delay. I know the
intentions are good, but hard-coded things like this make me
uncomfortable. Discussion?
JT
>Author: kpfleming
>Date: Thu Jan 11 17:42:14 2007
>New Revision: 50538
>
>URL: http://svn.digium.com/view/asterisk?view=rev&rev=50538
>Log:
>when a channel gets automatically answered by an application, sleep
>a bit to give the audio path (for VOIP channels) time to be setup
>
>Modified:
> trunk/main/channel.c
>
>Modified: trunk/main/channel.c
>URL:
>http://svn.digium.com/view/asterisk/trunk/main/channel.c?view=diff&rev=50538&r1=50537&r2=50538
>==============================================================================
>--- trunk/main/channel.c (original)
>+++ trunk/main/channel.c Thu Jan 11 17:42:14 2007
>@@ -1615,17 +1615,21 @@
> int ast_answer(struct ast_channel *chan)
> {
> int res = 0;
>+
> ast_channel_lock(chan);
>+
> /* You can't answer an outbound call */
> if (ast_test_flag(chan, AST_FLAG_OUTGOING)) {
> ast_channel_unlock(chan);
> return 0;
> }
>+
> /* Stop if we're a zombie or need a soft hangup */
> if (ast_test_flag(chan, AST_FLAG_ZOMBIE) || ast_check_hangup(chan)) {
> ast_channel_unlock(chan);
> return -1;
> }
>+
> switch(chan->_state) {
> case AST_STATE_RINGING:
> case AST_STATE_RING:
>@@ -1633,6 +1637,7 @@
> res = chan->tech->answer(chan);
> ast_setstate(chan, AST_STATE_UP);
> ast_cdr_answer(chan->cdr);
>+ ast_safe_sleep(chan, 500);
> break;
> case AST_STATE_UP:
> ast_cdr_answer(chan->cdr);
>@@ -1640,7 +1645,9 @@
> default:
> break;
> }
>+
> ast_channel_unlock(chan);
>+
> return res;
> }
>
>
>_______________________________________________
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