Displaying 20 results from an estimated 500 matches similar to: "Change verbose level"
2007 Jan 17
3
Asterisk 1.4 and CDR
Hi guys, I have recently installed a Asterisk Server with CDR Call Detail
Records. I have installed it over a Asterisk 1.2 , but now It do not run
. I have installed it with the following procedure:
# yum install ncurses
#yum install openh323-devel
# yum install mysql-server
# yum install mysql
# yum install php-gd
# yum install php-mysql
# yum install mysqlclient10
# yum install zlib
# yum
2007 Oct 18
4
Issues with making calls
Hi List,
I am from Peru, I have installed an asterisk server in my company with
digium card E1 TE120P, I am having issues when i make calls, here the
error from my server
[Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
channel type registered for 'Zap'
[Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full:
Unable to create channel of type 'Zap' (cause
2007 Feb 11
2
TDM02B not working
I am trying to reconfigure an asterisk box that was using an HFC-S card
with bristuff but is now using 2 analog lines therefore I want to use the
TDM02B to connect to two POTS lines. The TDM02B has 2 red modules.
I have this in /etc/zaptel.conf
loadzone=nl
defaultzone=nl
fxsks=1-2
I have /etc/asterisk/zapata.conf
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
2004 Sep 02
1
asterisk config and root
Hi
Another beginner's question:
Can I gain root if I have write access to asterisk's config files?
--
Tzafrir Cohen +---------------------------+
http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
mailto:tzafrir@technion.ac.il +---------------------------+
2007 Mar 07
1
Problem HandyTone 488 does not call transfer
Hi
I have a analog phone connected to my Gateway Handytone and registered to
Asterisk 1.4 I have configured my HandyTone 488
(in the section FXS Port) for make and receive calls, however I can
not transfer a call when it come via PSTN. But, when a call come from via IP
I can transfer it.
[phoneanalog]
type=friend
secret=XXXXXXX
context=local
nat=no
qualify=yes
host=dynamic
dtmfmode=rfc2833
2007 Aug 15
2
Disable MoH for certain phones
Hi,
Is it possible to configure asterisk so it doesn't play MoH from certain
phones?
Regards,
Jan
2007 Sep 20
1
Paging MEETME_RECORDINGFILE Variable
I am having a weird issue with setting the recording file for the
Page app. Here is some quick background info
I have a macro that pages all my phones:
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The
2008 Apr 14
2
polycom auto answer
I was trying to get my polycom phone to auto answer.
I added this to the dialplan. Used a different phone to call "22"
and the phone rang it did not auto answer.
Did I miss something?
exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0)
exten => 22,n,SipAddHeader(Alert-Info: Ring Answer)
exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0)
exten =>
2007 Oct 08
1
anyone using SIP trunks from Time Warner Telecom?
I am currently using a T1 PRI from TWTelecom for DID and outgoing
calls, but I recently discovered that they're offering call
termination/origination over SIP trunks in my area now. If they could
deliver these SIP trunks to me over a guaranteed-QoS circuit, this
would be of great interest to me. We're already using a DS3 circuit
from TW for our internet uplink, so I'd imagine it
2009 Feb 25
1
Realtime database function help
Hello Everyone!
According to voip-info.org the correcy syntax for the realtime function is:
REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read
REALTIME(family|fieldmatch|value|field) on write
It seems from the syntax that it is only possible to retrieve a full
row according to the value of only of column. This translates in SQL
language as Select * from family where fieldmath =
2007 Aug 06
1
CDR/MySQL basic config
Hi,
I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The
add-ons pack has been installed for a while, so now I'm trying to add
the Mysql config.
I've created a mysql database, added the grants for a user acces, and
can run a mysql -u asteriskcdruser -p and can connect to the database.
I've been using this as a guide:
2007 Nov 12
3
No sound from playback and voicemail
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to the outside, but I don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played, but I hear
nothing at all.
Here is as simple example:
[monkeys]
??? exten => 99,1,ANSWER()
??? exten => 99,2,PLAYBACK(tt-monkeys)
??? exten => 99,3,HANGUP()
The phone
2007 Sep 18
6
Limiting Simultaneous calls
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
Thanks
Jim
2009 Apr 27
1
Where I get free VoiP-in numbers?
Hi list,
Anyone knows how to get free VoiP-in numbers from USA or Canada, I
have found some links for example sipnumber.com but it does not run.
Also I want to know how to configure it in my asterisk server.
Thanks in advance.
Regards
2006 Jun 18
2
convert timespan to verbose description
Hi,
I always see RoR pages showing the elapsed time like that
''Published 3 days, 23 hour and 5 minutes ago''
''Published less than 3 minutes ago''
I guess this conversion can be done using a Rails function, however I
can''t find it.
Does somebody know its name??
Thanks a lot
Peter
2006 Jan 24
0
Verbose traces (debugging question)
Is there a way to coax a verbose trace out of Ruby on Rails?
For example, I just received this execution dump:
undefined method `pr'' for {"Q-SA1"=>"3"}:Answers
#{RAILS_ROOT}/app/models/interview.rb:58:in `self_eval''
(eval):6:in `self_eval''
#{RAILS_ROOT}/app/models/interview.rb:57:in `self_eval''
2009 Jan 16
4
Verbose Information from "zfs send -v <snapshot>"
What ''verbose information'' is reported by the "zfs send -v <snapshot>" contain?
Also on Solaris 10u6 I don''t get any output at all - is this a bug?
Regards,
Nick
--
This message posted from opensolaris.org
2012 Oct 16
1
core show channels verbose output
At the end of the output for "core show channels verbose" is a line that
reads "4 active calls". Does anyone know how that number is formatted
if there are more than 999 active calls? Will it have a comma or not?
--
Mitch
2008 Sep 11
0
Bug#498659: logcheck-database: amavis filter a little too verbose?
Package: logcheck-database
Version: 1.2.68
Severity: normal
Hi,
I use postfix, amavisd-new and logcheck on a lenny server, and the mails I
receive seem a little to verbose. For example, I get in the report all mails
which are received, for example:
Sep 11 08:35:11 heracles amavis[19788]: (19788-02) Passed CLEAN, [xxx.xxx.xxx.xxx] [xxx.xxx.xxx.xxx] <user at domain> -> <user at
2012 Jun 07
1
[PATCH] klcc --version is -V because -v is --verbose already (unbreak -v)
Signed-off-by: Thorsten Glaser <tg at mirbsd.org>
---
klcc/klcc.in | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/klcc/klcc.in b/klcc/klcc.in
index 43d0984..e03bf3c 100644
--- a/klcc/klcc.in
+++ b/klcc/klcc.in
@@ -136,7 +136,7 @@ while ( defined($a = shift(@ARGV)) ) {
} else {
die "$0: unknown option: $a\n";
}
- } elsif ( $a =~