Displaying 20 results from an estimated 400 matches similar to: "compiling asterisk-addons for Mysql-cdr"
2005 Feb 02
6
problem in compiling asterisk-addons
there is a problem in compiling asterisk-addons
any one have fixed this problem. i want
res_config_mysql.so any one help me
-----------------------------------------------------
[root@localhost asterisk-addons]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/include/mysql -c -o res_config_mysql.o
res_config_mysql.c
res_config_mysql.c: In function `realtime_mysql':
2005 Mar 18
2
Asterisk 1.0.7 Released
Hello everyone,
Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now
been released. Libpri and -addons have not changed, but have been
updated anyway to keep the version numbers consistent. All of the
tarballs are available on the ftp site.
ftp://ftp.asterisk.org/pub/asterisk/
I have posted the ChangeLogs for easy viewing at the following address.
2005 Jan 30
1
Slackware + Asterisk + asterisk-addons
Hello
I am trying to get asterisk-addons installed so that I can use the mysql
cdr feature. OK, I have the MySQL server (mysqld) installed, but I
noticed that mysql-devel is also required. I tried to compile
asterisk-addons and got a:
--CUT---
res_config_mysql.c:422: error: unknown field `realtime_multi_func'
specified in initializer
res_config_mysql.c:422: warning: excess elements in
2007 Nov 20
1
Realtime - mysql query gives wrong results??
Hi,
I am using Realtime for sip configuration.
When there is an INVITE which arrives at asterisk
asterisk makes the following selects:
Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect:
MySQL RealTime: Everything is fine.
[Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql:
MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name =
'tzl'
[Nov
2009 Mar 11
4
Are .call files working with extensions.ael ?
Hello,
With an extensions.ael enabled system, I keep getting whatever I change into
my "astup.call" file :
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least
one of app or extension (or keyword message/pdu) must be specified, along
with tech and dest in file /var/spool/asterisk/outgoing/astup.call
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:
2009 Mar 16
1
Bristuff bug or feature ? (Was: Are .call files working with extensions.ael ? bristuff problem)
Hi,
Is the following behaviour a bug or a feature ?
Using bristuff-0.4.0-RC3d.tar.gz, the call file thereafter produces :
[Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:267 apply_outgoing: At least
one of app or extension (or keyword message/pdu) must be specified, along
with tech and dest in file /var/spool/asterisk/outgoing/astup.call
[Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:457
2008 Dec 29
3
Manager API
Hi
I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial
out from manager's console and with Asterisk 1.4.X this settings were OK.
Action: Originate
Channel: SIP/384
Context: main
Exten: 102
Priority: 1
Callerid: 384
I could dial out, but with asterisk 1.6 I get this error.
Response: Error
Message: Channel not specified
I have originate and system privilege in
2010 Nov 05
2
How to append custom option to Contact: header on outgoing SIP INVITE msgs?
Hi list,
My need is to append a site specific parameter to the
Contact: header on all INVITEs exiting * via a SIP trunk.
I'd like it to look something like this:
Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here>
where SITE-ID=us.here is set in a config file that * parses on
startup. Or in a Dial() command option? Or I don't care exactly
how. :-)
It is possible to
2009 Oct 05
3
Questions about app_jack.c
Hello,
My configuration is :
Card 0 - kernel dummy sound card
Card 1 - my soundcard
I have a jackd running in background. My jackd launch command is :
jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
--capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
--outchannels 2 --dither triangular &
1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to
2005 Mar 10
5
asterisk and Broadvoice Outgoing Again :(
Hi,
I can't make outgoing calls via Broadvoice. I have tried each and every
configuration that was posted to list previously.
I am able to receive incoming calls fine.
I get the following in asterisk console:
=====================================================
asterisk*CLI> show version
Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running
Linux
2010 Oct 12
0
rtpip patch
Hello *,
is the rtpip patch still valid for asterisk 1.6 (with some code
changes, obviously)?
https://issues.asterisk.org/view.php?id=8161
Or, in asterisk 1.6 there is an alternative to using it?
This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11
--- chan_sip.c 2010-10-12 13:47:49.000000000 +0200
+++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200
@@ -987,9 +987,6 @@
2004 Jun 26
2
Newbie needs help
I've been banging my head on a brick wall for about an hour now trying
to understand why the following doesn't work (which is even provided as
an example in the distribution!).
The goal is to create a voicemail-only extension not associated with a
phone. I'd rather not have an extension dedicated to VoicemailMain(),
so I would like the user to be able to hit '*' during
2008 Jul 07
1
cdr_addon_mysql - additional fields
Hi,
I need help with modifying cdr_addon_mysql.c I want to have more
fields in cdr table in asterisk. I've tried to modify cdr_addon_mysql.c
and replace userfield with ex team (sed -e 's/userfield/team/g' ). When
I try to recomplie
menuselect/menuselect --check-deps menuselect.makeopts
Generating embedded module rules ...
make[1]: Nothing to be done for `all'.
2007 Dec 19
3
Realtime logic in Asterisk 1.4.16.1
Hello,
I have configured one provider in Asterisk Realtime DB without username and password, only host=<providers_IP> and ipaddress=<providers_IP>
Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/NUMBER at Provider)
In Asterisk 1.4.15 debug I see that Realtime engine is using query:
[Dec 20 00:02:15] DEBUG[14634]:
2007 Jun 28
2
fail to load modules
Hi all,
I am a bit out with the Asterisk 1.4.4, after I complied and installed the
Asterisk and I got such error messages
[Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are
available to listen on, not starting SDMI listener.
[Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module
'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined
symbol:
2010 Sep 08
1
Upgrade from 1.4 to 1.6 : problems with realtime mysql
Hello,
in asterisk 1.4.30 all realtime configurations go well.
In asterisk 1.6.2.11 the following appears on CLI :
[Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
[Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL RealTime: Invalid database specified: MyDBase (check
2004 Jul 21
0
Voicemal error
Hi, i've a proble using voicemail. when i make a call and start voicemail
asterisk tell me mail address is missing even if i used it as written
mailbox => name,pwd,mail@mail
I saw that modifying in app_voicemail.c line 836 in this manner: if (vmu
&& ast_strlen_zero(vmu->email)), so replacing !(ast_strlen_zero(vmu->email)),
it works.
did anyone have the same problem? or is
2006 Dec 06
1
0002475: [patch] Allow app_directory to work with REALTIME
Hi All,
I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to
stable release or is it still only in CVS. Will this file patch apply
correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing
app_directory_realtime_1.6.1.patch
<http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and
config.h.patch
2007 Jun 21
7
asterisk 1.4.1 app_addon_sql_mysql
when I enter asterisk-addons-1.4.1 and make menuselect
*************************************
Asterisk-addons Module
Selection
*************************************
Press 'h' for help.
XXX 1.
app_addon_sql_mysql
2006 Jun 14
0
Directory - First Name/Last Name - How to, use both? a@h?
We wrote and submitted a patch to do this. Just modify app_directory.c
and recompile. It adds a new flag "b" to the directory( ) app where you
can have it use both first and last name.
-= Info about application 'Directory' =-
[Synopsis]
Provide directory of voicemail extensions
[Description]
Directory(vm-context[|dial-context[|options]]): This application will
present