On Mon, 2006-10-09 at 19:39 -0500, Moises Silva wrote:
Same problem as your other post. dtmf_put is no longer available in
newer spandsp versions, the solutions is the same as with libmfcr2,
downgrade spandsp, or upgrade chan_unicall (not always a matching
Ok, I downgraded all the
Hi
I noticed that in the normal flow of invite message, there is a
message 407 in between.
UA1 -INVITE -- AST1
407--
--ACK
--INVITE---
--Trying--
There is a problem if 407 exists
I have a macro that bridges a call or sends it to
VM based on what the called user presses. If he just hangs up the call gets
dissconnected. This is the error I get
WARNING[7598]: res_features.c:1381
ast_bridge_call: Bridge failed on channels SIP/11011-9ea135e0 and
SIP/11010-007d0780
Is
On Fri, Jul 21, 2006 at 06:13:50PM -0500, brandon kruz wrote:
in addition to russel
use
(in ubuntu)
sudo netstat
or man netstat for further, more precise methods
look for your specific port
eg
sudo netstat -a | grep 5060
and it shoudl tell you the process name, and what directory it is
MySql password for root:
Domain (realm) for the default user 'admin': localhost.localdomain
creating database openser ...
ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password:
Y ES)
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This isn't even a question... It's just an error. Worse is that it appears to be an error for OpenSER, not Asterisk. Even so, your error looks to be a simple authentication error. Make sure the root user exists for your MySQL install and that you can login to MySQL via the command line with these
El vie, 21-07-2006 a las 18:53 -0400, Russell Bryant escribió:
On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote:
Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to
bind to 10.152.58.9:5060: Address already in use
It looks like another application on your
@lists.digium.com
Subject: Re: [asterisk-users] Error in ubuntu dapper
Date: Fri, 21 Jul 2006 21:10:14 -0400
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I'm using asterisk 1.2.7.1 on ubuntu dapper. It was working.
But today, without changing nothing in the config, and without
connecting-desconnecting anything, it began to give me this error:
Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to bind
to 10.152.58.9:5060:
On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote:
Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to bind
to 10.152.58.9:5060: Address already in use
It looks like another application on your system is using port 5060.
Did you install any new software such as
in addition to russel
use
(in ubuntu)
sudo netstat
or man netstat for further, more precise methods
look for your specific port
eg
sudo netstat -a | grep 5060
and it shoudl tell you the process name, and what directory it is comming
from
shut it off
and do that
sudo netstat -a | grep 5060 again
Yeah, ekiga was using port 5060, althoug netstat -a didn't say it.
I might issue netstat -alpn
The weird thing is that I had configured ekiga so that it used port
5061, but unfortunatly if ekiga is run before asterisk it catchs port
5060 too.
El vie, 21-07-2006 a las 18:13 -0500, brandon kruz
Hello,
I get this error message when trying to route an incoming fax from a packet
based T1 to an EICON board that is connected to an external fax voice mail
server.
Voice calls route to this external server with no error. Both fax and voice
calls that come in a channelized T1 also route
__
Od: [EMAIL PROTECTED]
Komu: asterisk-users@lists.digium.com
Datum: 18.07.2006 20:42
Předmět: [asterisk-users] Error: Dropping incompatible voice frame
Hello,
I get this error message when trying to route an incoming fax from a
packet based T1 to an EICON
Hi list,
I get this message sometimes ( randomly ) when queues are calling agents:
Jul 10 11:26:46 ERROR[8856]: app_dial.c:1481 dial_exec_full: Could not
stop autoservice on calling channel
I'm trying to see where it comes from ...
Does someone has an idea ???
Thanks in advance !
My teeth are on edge after this one. A couple of perfectly good hours
of my life, and I still don't know what's going on. . . .
The extensions.conf.sample that comes with the current SVN trunk has
this line, in an example that shows how to use ChanIsAvail:
exten = s,n,GoToIf([${AVAILSTATUS}
Brian Capouch wrote:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
I couldn't get this to work unless I surrounded the first part of the
test with quotes, too, like this:
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
Ooops.
Actually, I mis-pasted one of my
. juni 2006 09:10
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Error in config sample for GoToIf?
My teeth are on edge after this one. A couple of perfectly good hours
of my life, and I still don't know what's going on. . . .
The extensions.conf.sample
Jon Schøpzinsky wrote:
Hello
As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail)
${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1
Through more testing, the double quotes I used seemed superfluous; if
you use them in both
BLAH=1BLAH=1On 6/27/06, Brian Capouch [EMAIL PROTECTED] wrote:
Jon Schøpzinsky wrote: Hello As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1Through more testing, the double quotes
hello
when I execute asterisk I have this
error
Loaded /usr/lib/asterisk/modules/chan_oss.so
= (OSS Console Channel Driver)Jun 9 11:40:44 NOTICE[5284]:
chan_oss.c:1380 load_module: Unable to load config oss.confJun 9
11:40:44 WARNING[5284]: loader.c:414 __load_resource: chan_oss.so:
Hi, all. Every now and then, some of my users get Error on their
phones. A reboot fixes it, but it's quite annoying/inconvenient. I'm
running Asterisk 1.2.4, and have the following firmware, etc.:
Bootrom: 2.6.2.0032
BootBlock: 2.5.0(11500_030)
SIP application: 1.6.2.0041
Any ideas as to why
Are you using an idle webpage? If for some reason the phone can't reach the
page it will display an error and rebooting is about the only way to fix it.
--johann
Ken D'Ambrosio wrote:
Hi, all. Every now and then, some of my users get Error on their
phones. A reboot fixes it, but it's quite
Hi Ken -
Bootrom: 2.6.2.0032
BootBlock: 2.5.0(11500_030)
SIP application: 1.6.2.0041
In any case, I'd suggest updating to a later firmware version. SIP
firmware 1.6.6 has been officially released. If you are unable to get
it, just send me a personal email (offlist).
- Noah
On 5/25/06,
Hello Im having problem building oh323 on SuSE Linux 9.3, I have build the
openH323 and pwlib and Im getting the following error:
g++ -Wall -felide-constructors -x c++ -Os -DP_USE_PRAGMA
-ffunction-sections -fd
ata-sections -D_REENTRANT -Wall -fPIC -I/usr/src/pwlib/include -DPTRACING
-I/usr
Has anyone seen this (version 1.2)?
The following dialplan should result in the voicemail
message being delivered two both mailboxes ([EMAIL PROTECTED] and
[EMAIL PROTECTED]);
exten = 0,1,SetCIDName(OPER ${CALLERIDNAME})
exten =
Hi Damon -
The following dialplan should result in the voicemail message being
delivered two both mailboxes ([EMAIL PROTECTED] and
[EMAIL PROTECTED]);
The actual result is an error copying the message to the second mailbox as
follows;
I do the same thing (with version 1.2.7.1), and it's
Hi all,
I always get this error message after I hangup a call, what does it mean ?
WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame
cheers,
hn.
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On 4/30/06, Hatami Nugraha [EMAIL PROTECTED] wrote:
Hi all,
I always get this error message after I hangup a call, what does it mean ?
WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame
This means you hungup while asterisk was trying to play a file to you.
It should be
Are there some instructions how to solve problems that produce some typical
error messages in asterisk? For example, if I don't use iax, dundi or mysql
logging, every time I start asterisk I'll get several error messages. How What
can I do to disable loading those files?
Here re some error
Tomislav Parčina wrote:
Are there some instructions how to solve problems that produce some typical
error messages in asterisk? For example, if I don't use iax, dundi or mysql
logging, every time I start asterisk I'll get several error messages. How What
can I do to disable loading those
I'm running:
OS: FreeBSD 6.0
Asterisk: 1.2.4
Installing OH323:
0.7.3
I have this error
when compiling
chan_oh323.c: In
function `reload_config':
chan_oh323.c:4677:
warning: implicit declaration of function `sscanf'
chan_oh323.c: At
top level:
chan_oh323.c:3244:
warning:
-Users] Error installing oh323
I'm
running:
OS: FreeBSD
6.0
Asterisk:
1.2.4
Installing OH323:
0.7.3
I have this error when
compiling
chan_oh323.c: In function
`reload_config':
chan_oh323.c:4677: warning:
implicit declaration of function `sscanf'
chan_oh323.c: At top
level:
chan_oh323.c
Hi all
I want to use sphinx2 with asterisk. I install
sphinx but when i type sphinx2-server, i have the
errors below:
ad_oss.c(105): Failed to open audio device(/dev/dsp):
No such device
FATAL_ERROR: server.c, line 476: ad_open() failed
Thanks for all!
I am instaling asterisk on Fedora core 3. I have instaled zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error:_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_zapscan.o app_zapscan.cgcc
yum install libidn-develOn 4/19/06, Luis Fernando Ramírez Cueva [EMAIL PROTECTED] wrote:
I am instaling asterisk on Fedora core 3. I have instaled
zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error:_GNU_SOURCE -O6 -march=i686
Someone know the meaing of this error?chan_sip.c:3853 copy_via_headers: No header field 'Via' present to copy
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On Sat, 2006-03-25 at 09:41 +0100, Dave Cotton wrote:
On Sat, 2006-03-25 at 11:52 +0330, Paradise Dove wrote:
hi,
i've just upgraded to latest trunk. everything compiles fine but when
starting this message appears and fails to start.
WARNING[3990] loader.c: module chan_zap.so error
hi,
i've just upgraded to latest trunk. everything compiles fine but when
starting this message appears and fails to start.
WARNING[3990] loader.c: module chan_zap.so error
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
ast_pickup_call
thanks,
paradise dove
On Sat, 2006-03-25 at 11:52 +0330, Paradise Dove wrote:
hi,
i've just upgraded to latest trunk. everything compiles fine but when
starting this message appears and fails to start.
WARNING[3990] loader.c: module chan_zap.so error
/usr/lib/asterisk/modules/chan_zap.so: undefined symbol:
Hi,
I am using 1.2.3, and sometimes I can see the
following message:
Mar 2 08:42:42 WARNING[25937] ast_expr2.fl:
ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting
TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:+
1^Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: If
On 09:44, Thu 02 Mar 06, Dov Bigio wrote:
Hi,
I am using 1.2.3, and sometimes I can see the following message:
Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN; Input:
+ 1
Michiel van Baak wrote:
On 09:44, Thu 02 Mar 06, Dov Bigio wrote
Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN; Input:
+ 1
^
Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: If you
On 08:32, Thu 02 Mar 06, Doug Lytle wrote:
From what has been discussed in the last month, it would indicates that
it's a variable that hasn't been defined before doing a math function.
(i.e. not setting a=0 before doing an a=a+1)
Hhmm, Only thing in my setup that looks like it can go wrong
At 06:00 AM 03/02/2006, you wrote:
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3)
Looks like the Dial statement is not setting the $DIALEDTIME
in some cases.
This is the general solution to that.
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3)
Ira
--
No virus found in this
hi All,
Why do i get this error when I click on outbound routing?
Warning: Missing argument 5 for addroute() in
/var/www/html/admin/functions.php on line 1313
Warning: Missing argument 5 for addroute() in
/var/www/html/admin/functions.php on line 1313
has anyone encountered this error before?
On Sat, Feb 11, 2006 at 10:17:37AM +0500, ast guy wrote:
Hi,
I have downloaded iaxcomm version iaxcomm-lin-1.0rc3, when I try to
execute it it gives following error.
# ./iaxcomm
Error wxWindows Fatal Error : Couldn't Initialize IAX Client .
any idea what's going wrong ?
No. But this is
Hi,
I have downloaded iaxcomm version iaxcomm-lin-1.0rc3, when I try to
execute it it gives following error.
# ./iaxcomm
Error wxWindows Fatal Error : Couldn't Initialize IAX Client .
any idea what's going wrong ?
-ag
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Hi,
After installing mysql, mysql-devel mysql cdr add
on, I get the following error when I start Asterisk:
[res_config_mysql.so]2006-02-03 18:41:16
WARNING[24786]: loader.c:325 __load_resource:
/usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol:
_intel_fast_memcpy
My server
It's been about 2 months since I have updated my asterisk box.
I was running CVS HEAD and I notice a whole lot has changed since
my last update!
I'm running Debian Sarge up to date on a 2.4 Kernel.
I was updating about every 2 or 3 weeks and never had any problems
compiling
Bummer - Possibly a bug
The stable stuff compiles and runs fine :(
Steve
-
It's been about 2 months since I have updated my asterisk box.
I was running CVS HEAD and I notice a whole lot has changed since
my last update!
I'm running Debian Sarge up to date on a 2.4 Kernel.
I was
Here is the issue:
[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# lsmod | grep zaptel
zaptel206724 0
crc_ccitt 2113 1 zaptel
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is
I am compiling a app file writting in C for asterisk and I am getting the following errors: ../include/asterisk/file.h:27:2: #error You must include stdio.h before file.h!In file included from app_akEventsProxy.c:17:../include/asterisk/file.h:56: error: syntax error before '*'
jourdan lemieux wrote:
I am compiling a app file writting in C for asterisk and I am getting the
following errors:
../include/asterisk/file.h:27:2: #error You must include stdio.h before
file.h!
Any help please on this!!
How much clearer can that be? Your source file is out of date
Iam using Fedora 3 and gcc is installed Please let me knowMark Quitoriano [EMAIL PROTECTED] a écrit: Jourdan,What Distro are you using? do you have gcc installed? On 12/6/05, jourdan lemieux [EMAIL PROTECTED] wrote: Any help on this please Hi, I am getting this error when comp
please follow this instruction
http://www.voip-info.org/wiki/view/Asterisk+Fedora+Core+3
let me know what happened.On 12/6/05, jourdan lemieux [EMAIL PROTECTED] wrote:
Iam using Fedora 3 and gcc is installed Please let me knowMark Quitoriano
[EMAIL PROTECTED] a écrit: Jourdan,What Distro are
Any help on this pleaseHi, I am getting this error when compiling asterisk `ls *.c`: unrecognized optionh -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ...GNU long options: --debug --dump-po-strings
Jourdan,
What Distro are you using? do you have gcc installed?On 12/6/05, jourdan lemieux [EMAIL PROTECTED] wrote:
Any help on this pleaseHi, I am getting this error when compiling asterisk
`ls *.c`: unrecognized optionh -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long
Hi, I am getting this error when compiling asterisk `ls *.c`: unrecognized optionh -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ...GNU long options: --debug --dump-po-strings --dump-strings --help --login
:33 AM
Subject: Re: [Asterisk-Users] Error on using queue.
If you are using 1.2, it might be the joinempty and leavewhenempty
parameters.
Their default are different than the 1.0.x releases
- Original Message -
From: gc
To: Asterisk Users Mailing List - Non-Commercial
Subject: Re: [Asterisk-Users] Error on
using queue.
Thanks. I made change to joinempty=yes. And now
I can hear the music on hold. But it would not ring the agent even if I
login agent in. When I run show queue command under CLI, I got these
messages:
queue1 has
1
I am trying to use * as ACD server for our sip
proxy.
I first dial 55 to login 98 as ACD
agent it worked fine and then when I dialed 98, I got these messages from * CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing
Subject: [Asterisk-Users] Error on using
queue.
I am trying to use * as ACD server for our sip
proxy.
I first dial 55 to login 98 as
ACD agent it worked fine and then when I dialed 98, I got these messages from * CLI:
-- Executing
Answer(&quo
List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 8:33
AM
Subject: Re: [Asterisk-Users] Error on
using queue.
If you are using 1.2, it might be the joinempty
and leavewhenempty parameters.
Their default are different than the 1.0.x
releases
How is your agents.conf ? How is your login in
extensions.conf?
- Original Message -
From:
gc
To: Dov Bigio ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 2:53
PM
Subject: Re: [Asterisk-Users] Error on
using queue
These errors just started showing on asterisk cli. Me setup is a
pri/e1 card, connected to a philips PBX.
google gave no answers.
Any ideas???
2005-11-15 11:25:58 ERROR[23932]: utils.c:509 tvfix: warning negative
timestamp -267121.-532000
2005-11-15 11:25:58 ERROR[23932]: utils.c:509
I was trying to install the new RC1 version of 1.2 and I get the
following error:
In file included from app_rxfax.c:15:
../include/asterisk/file.h:27:2: error: #error You must include stdio.h before
file.h!
In file included from app_rxfax.c:15:
../include/asterisk/file.h:56: error: syntax
Dear All,
I am facing a problem in compiling the add-ons for the mysql, though the files
are downloaded correctly and checked and I tried different mirrors even the cvs
but yet I get those errors :
app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory
cdr_addon_mysql.c:38:19: mysql.h:
You need to install the libmysqlclient-dev package. Search the
rpm/deb/src available for your distrib and install it.
Regards,
Le mar 08/11/2005 à 10:44, Mohamed A. Gombolaty a écrit :
Dear All,
I am facing a problem in compiling the add-ons for the mysql, though the files
are downloaded
Ever since I started playing with Beta versions of Asterisk, I've had a
problem. It might just be coincidence though, since before that I didn't
touch the Asterisk PC for a good 2 weeks and I had done alot playing around
with config files.
I have a 4 port FXS/FXO card (with 2 of each
Thanks Rich for your reply.
If you modprobe zaptel and wctdm, then run ztcfg -vvv, you
shoud see the four modules like this:
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02:
FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS
Kewlstart (Default) (Slaves:
Hello,
Ever since I started playing with Beta versions of Asterisk, I've had a
problem. It might just be coincidence though, since before that I didn't
touch the Asterisk PC for a good 2 weeks and I had done alot playing around
with config files.
I have a 4 port FXS/FXO card (with 2 of
Dear Asterisk developers,
I run the same asterisk version on the home machine and on the work. On
the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work
machine I have Mandrake 10.1 (kernel 2.6.8.1).
When I run asterisk on the work machine, these warnings and error appear
Corrado wrote:
Dear Asterisk developers,
I run the same asterisk version on the home machine and on the work. On
the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work
machine I have Mandrake 10.1 (kernel 2.6.8.1 http://2.6.8.1).
When I run asterisk on the work machine, these
Hi,
I have installed AAH beta 4 and I am getting this error. I have installed it
from aahbeta.tar.gz so I can make the server dual boot.
this is what I am getting in error, any clue how I can fix this?
Thanks
Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream: Permission
Hi,
I have installed AAH beta 4 and I am getting this error. I have installed it
from aahbeta.tar.gz so I can make the server dual boot.
this is what I am getting in error, any clue how I can fix this?
Thanks
Warning: fopen(/etc/asterisk/vm_general.inc): failed to open stream:
Permission
They are chown to asterisk:asterisk and chmod 777 . But I am still getting
those error.
Any other suggestion?
Thanks
Quoting asterisk [EMAIL PROTECTED]:
Hi,
I have installed AAH beta 4 and I am getting this error. I have installed it
from aahbeta.tar.gz so I can make the server dual
On Wed, Oct 12, 2005 at 01:44:34AM -0400, Andy Goss wrote:
If anyone could tell me what this error is all about, I would be very
grateful.
Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not
of Tzafrir Cohen
Sent: Wed 10/12/2005 2:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] error message when accessing voicemail
On Wed, Oct 12, 2005 at 01:44:34AM -0400, Andy Goss wrote:
If anyone could tell me what this error is all about, I would be very
grateful.
Oct
If anyone could tell me what this error is all about, I would be very
grateful.
Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not
permitted
Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path:
Hi,
I am running asterisk on Fedora Core 3, Configured few extension, I receive
frequent error message on * console as
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back from
xxx.xxx.xxx.xxx.
This only comes from two extensions which I configured
Any idea what does this error
i get this error on dmesg:
zaptel: Unknown symbol __stack_smash_handler
zaptel: Unknown symbol __guard
paradise dove
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On Sat, Oct 01, 2005 at 05:06:13PM +0330, Paradise Dove wrote:
i get this error on dmesg:
zaptel: Unknown symbol __stack_smash_handler
zaptel: Unknown symbol __guard
Seems you built zaptel with different kernel headers/config than the
one you're currently running.
Care to give more
when I searched for _**6XX the site returned:
An error occured in a database query!
Context:
File/tiki-searchresults.php
Url
/tiki-searchresults.php?words=_**6XXwhere=pagessearch=go
Query:
SELECT COUNT(*) FROM tiki_comments c, tiki_pages p
WHERE c.objectType = wiki page AND
[app_rxfax.so]Sep 9 13:26:01 WARNING[99807]:
loader.c:258 ast_load_resource: /usr/local/lib/libspandsp.so.0: Undefined symbol
lrint
Jul 9 23:49:01 WARNING[99807]: loader.c:440 load_modules: Loading module
app_rxfax.so failed!
How do I fix this?
: Tuesday, August 30, 2005 9:43 AM
Subject: Re: [Asterisk-Users] error compiling on solaris 10
Message: 26
Date: Mon, 29 Aug 2005 15:26:31 +0800
From: chris [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] error compiling on solaris 10
To: Asterisk Users Mailing List - Non-Commercial
i can fix this.
thnks
- Original Message -
From: Frank Tarczynski [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 29, 2005 3:08 AM
Subject: RE: [Asterisk-Users] error compiling on solaris 10
Message: 11
Date: Sun, 28 Aug 2005 11:46:29 +0800
From: chris
Message: 26
Date: Mon, 29 Aug 2005 15:26:31 +0800
From: chris [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] error compiling on solaris 10
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
hello,
any advise? :)
thnks
- Original Message -
From: chris [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, August 29, 2005 3:26 PM
Subject: Re: [Asterisk-Users] error compiling on solaris 10
hi frank,
i
Message: 11
Date: Sun, 28 Aug 2005 11:46:29 +0800
From: chris [EMAIL PROTECTED]
Subject: [Asterisk-Users] error compiling on solaris 10
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
Hey,
does anyone know why i'd be receiving:
Aug 28 19:40:04 DEBUG[1875]: # Testing 66.27.233.241 with 10.0.10.0
Aug 28 19:40:04 DEBUG[1875]: Target address 66.27.233.241 is not local,
substituting externip
I get tons of them, usually when the phone is registering/calling/receiving
Anyone know what this means:
Aug 27 12:55:02 WARNING[7799]: chan_sip.c:959 __sip_xmit: sip_xmit of
0x94c5c80 (len 734) to 192.168.2.29 returned -1: Invalid argument
___
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Asterisk-Users mailing
hello,
i change my OS from solaris 9 to solaris 10, tried
running make to install asterisk but i'm getting the error below:
make -C editline libedit.amake[1]: Entering
directory `/export/home/fst/ice/cvs/asterisk/editline'/bin/sh makelist -h
common.c common.h/bin/sh makelist -h emacs.c
On Wed, 24 Aug 2005, Humberto Aicardi wrote:
Hi,
I've a Fritz card which was working fine, recently I changed hardware and
my nightmare started. Now when I call someone through the chan_capi (0.3.5 or
0.4.0) it works fine but when I receive calls I always get hungup. Can someone
please
On 8/25/05, Humberto Aicardi [EMAIL PROTECTED] wrote:
I've a Fritz card which was working fine, recently I changed
AVM Fritz Passive card? If so, then it doesn't work very well in
Pointto Point. Ask your Telco for a Point to Multipoint and change the
setting in capi.conf./
Note:
I am not
Hi,
I've a Fritz card which was working fine, recently I changed
hardware and my nightmare started. Now when I call someone through the
chan_capi (0.3.5 or 0.4.0) it works fine but when I receive calls I
always get hungup. Can someone please give some help? Here are the logs:
*CLI
--
-users@lists.digium.com'
Subject: [Asterisk-Users] Error compiling meetme2
app_meetme2.c:646: error:
'__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
undeclared here (not in a function)
make[1]: *** [app_meetme2.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/apps'
make
hello,i was able to install openssl by hand successfuly, then i
tied make.. theerror is till there,any more
ideas?thnks
- Original Message -From: "Derek Whitten" [EMAIL PROTECTED]To: "chris" [EMAIL PROTECTED]Sent:
Saturday, August 13, 2005 12:29 AMSubject: Re: [A
app_meetme2.c:646: error:
'__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared
here (not in a function)make[1]: *** [app_meetme2.o] Error 1make[1]:
Leaving directory `/usr/src/asterisk/apps'make: *** [subdirs] Error
1
I need a
copy of the app_meet2.c file that
Dear all,
I am getting the below errors when using asterisk.
I am using sjphone for testing purpose.
Below are the setting for sip.conf and
extension.conf
When i dial the number it rings on the remote
telephone. but after ringing 1 time it will disconnect.
Can anybody tell me what does
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