similar to: Master.csv

Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

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

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the

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, I am running Redhat, I loaded the mysql and mysql-devel RPM's and then recompiled *.. I thought it would be that simple but it looks like I have missed something becasue it doesn't look like the module has been complied.. What did I leave out? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by

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

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

When I set up my Asterisk box at home I didn't want to have to dial 9 to dial off premises, so I gave all my local phones three digit extensions with this format: 1[1,0]*. My thought is that there are no area codes that start with 0 or 1, so if I use those numbers, I can create 20 local extensions that can be dialed with 3 digits, and not have to use a timeout when dialing long distance. If

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

hi, i saw the cdr file called Master.csv and i want to know what these represent. examples "","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u8888","2003-11-07 17:43:04","2003-11-07 17:43:04","2003-11-07 17:43:22","ANSWERED","DOCUMENTATION"

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at

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, 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 All, I am installing a Zaptel Quad E1 card with ISDN PRI in Brazil and I?m with this problem: Asterisk CLI : Mar 27 01:49:57 WARNING[2612]: chan_zap.c:7143 zt_pri_error: PRI: !! No channel map, no channel, and no ds1? W hat am I supposed to identify? Mar 27 01:49:57 WARNING[2612]: chan_zap.c:7143 zt_pri_error: PRI: !! Unable to add IE 'Channel Identification' == Restart

Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't

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

Whenever I issue the reload command, asterisk crashes. Below is the output I get from (gdb) bt Any help is appreciated. *************************************************************** *CLI> reload == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Not found (No such file or

If I have ODBC logging enabled (with cdr_odbc), Asterisk logs everything to ODBC *and* to the CSV file (Master.csv). If I issue a "reload", it stops logging to the CSV file, but continues logging to ODBC.   To have it log to the CSV file again, I have to issue "unload cdr_csv.so" then "load cdr_csv.so".   Is that normal behaviour? Is it supposed to log to the CSV file

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