similar to: CDR problems

Hi, I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've noticed that the CDR logging in MySQL (on a different computer) has stopped. I thought it wasn't logging anything at all, but I realized after a bit of searching that there were log files in /var/log/asterisk/cdr_custom and /var/log/asterisk/cdr_csv with up to date logs. My cdr_mysql.conf is set up

Hi, my Asterisk 1.2.9.1 suddenly freezed ("stop now" did not work!!) . I found the following errors inside /var/log/asterisk/message: Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many open files Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14

Hi For whatever reason we've getting 2 or 3 CDR lines logged for each call, often in different formats: as1:~# grep test-89-1e2c /var/log/asterisk/cdr-csv/*.csv

hello, im having trouble with asterisk with medium load, it seems im running out of files, here is a chunk of the logs with grep "\(file\|pipe\)": Apr 18 15:40:46 WARNING[11644] res_agi.c: Unable to create toast pipe: Too many open files Apr 18 15:40:46 WARNING[11574] channel.c: Channel allocation failed: Can't create alert pipe! Apr 18 15:40:46 WARNING[11574] channel.c: Channel

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

Hi i am upgrading an asterisk server from 1.2.4 to 1.4. i've installed libpri 1.4 i've installed zaptel 1.4 I've installed the new version of misdn with the script of beronet. i use this configure script: ./configure --with-postgres=/usr/local/pgsql then: make menuselect [*] 1. cdr_csv [*] 2. cdr_custom [*] 3. cdr_manager XXX 4. cdr_odbc [*] 5. cdr_pgsql XXX 6. cdr_radius

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

Now that I have most everything actually working I've noticed that about every 3-4 days on average..... and at worse... Once a day my asterisk box seems to lose it's registered state with our sip provider and no longer will take any incoming calls. The caller simply hears a fast busy (reorder) If I do a reload at the command prompt all is well for another few days..... What I'm

hi there :) i get this error in the asterisk CLI: Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away when I run 'cdr mysql status' i get: Connected to voiceone at localhost, port 3306 using table cdr for 18 minutes, 24 seconds. Wrote 8 records since last restart and 1 records since last reconnect. I

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

I have built Asterisk 1.4.4 on my Solaris 10 x86 box: LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib' CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw --without-oss --without-vpb --prefix=/opt/asterisk-1.4 The build and install go fine but the asterisk executable reproducibly dumps core with a segmentation violation. If I start it as: asterisk -gc and

> Message: 1 > Date: Tue, 15 May 2007 23:01:24 -0400 > From: Frank Tarczynski <ftarz at mindspring.com> > Subject: [asterisk-users] Asterisk 1.4.4 reproducibly dumps core on > Solaris 10 > To: asterisk-users at lists.digium.com > Message-ID: <464A7404.5000706 at mindspring.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I have

Hello All, It seems that this question is very stupid, but anyway. Do I need any additional configuration for cdr_csv.so? This module is loaded by default at Asterisk's startup (asterisk -fvvv): [cdr_csv.so] => (Comma Separated Values CDR Backend) But when I place call I didn't see anything in /var/log/asterisk/cdr-csv. There is also no errors or warnings regarding this

Hi all. I have some question about cdr_csv. We want to have some IVR system for routing incoming calls to multiple directions. So when user make a call, he must put his PIN and phone number. Then asterisk will Dial this call to appropriate destination. Billing system deal with cdr_csv records. But the "Billable seconds" field - all time of the call, including the IVR part (that

I have asterisk 1.4 ,the function that I am using in extensions.conf is not functioning Its was functioning on asterisk 1.2.further more cdr_addon_mysql.so cdr_csv.so cdr_custom.so cdr_manager.so cdr are loaded Is there any missing module ? Function IS for example exten => *70,n,MYSQL(Connect connid 127.0.0.1 root passw0rd asterisk) exten =>

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

Dear ALL, Any clues or tips for the following gdb messages. [root@localhost asterisk]# uname -a Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct 29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux localhost*CLI> show version Asterisk CVS-HEAD-09/22/04-11:19:09 built by root@localhost on a i686 running Linux [root@localhost asterisk]# gdb asterisk core.13089 GNU gdb Red Hat Linux

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial