similar to: Verbose() messages go unnoticed

I'm not really being too lucky in the last days. After trying to compile cdr_mysql with no success, I am switching to cdr_odbc. I have installed unixODBC, iODBC and MyODBC correctly, I am even able to make queries with isql. But when trying to "make" in the cdr directory of the latest CVS, that's what I get: # cd /usr/src/asterisk/cdr # make cc -o cdr_odbc.so cdr_odbc.o -lodbc

-----BEGIN PGP SIGNED MESSAGE----- ============================================================================= FreeBSD-SA-00:41 Security Advisory FreeBSD, Inc. Topic: Malformed ELF images can cause a system hang Category: core Module: kernel Announced:

Hi All I just completed a custom application for Asterisk (i m not a C guru so i just copy codes from other application and alter according to my needs) attached files is the source file this application is working fine but still i need you people to give suggestion to improve it Primary task of this application is to get a parameter from extensions.conf, query sql server and play a files

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key

Hi, Since Carl has kindly provided us with fax support for CAPI based cards, we have been using it with much success. Today I have modified app_capiFax so that it now supports a dynamic CSID. The following example uses the DNID created by chan_capi on an AVM Fritz! card. * Receive a fax with CAPI API. * Usage : capiAnswerFax2(path_output_file.SFF|stationID) * * This function can be

>Did you check your security log? >There is usually a wealth of info there about who, what, where when and why I also checked /var/log/asterisk/messages and it just has the same line. Nothing additional. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to add cdr_odbc.so to log my CDR data to a mysql DB. I've managed to compile everything, and seem to almost be ready to head home. I've added a small debug line to cdr_odbc.c as follows: if((ODBC_res != SQL_SUCCESS) && (ODBC_res != SQL_SUCCESS_WITH_INFO)) { if(option_verbose > 10) ast_verbose(

I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8. I do not pass the 'n' option to any call to Queue() in my dialplan. Yet since I upgraded to 1.4.9, I have occasionally seen this on my console: -- Nobody picked up in 20000 ms -- Exiting on time-out cycle That log message "Exiting on time-out cycle" is exclusive to the logic in app_queue meant to

Is this the right place to ask questions about PXE and booting linux via PXE? -- Petro at corp.vendio.com ccpetro at vtext.com [sms] Unix Administrator 2766480 at skytel.com [pager] Vendio Service Inc. (650) 793-1650 [cell]

I forgot to take out the portion that would read in the voicemail boxes from the text file. If you want to leave it in then you could have some voicemail boxes defined in the text voicemail.conf. I do not, so I have removed it. Below is the new patch: *** app_voicemail.c 2004-06-23 07:55:54.000000000 -0600 --- app_voicemail.c.new 2004-06-23 07:55:47.000000000 -0600 *************** *** 49,61 ****

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

Hello all, I am just getting going on building my system, but I thought I'd send you all a patch that I wrote so the command: show voicemail users issued from the CLI works properly when there is a postgres backend for the voicemail. The current version of the app does not display the voicemail boxes found in a database. It is called in the load_config function. I haven't done

Hello All: I'm having some difficutly getting res_config_mysql from the 1.4.1 addons package to compile ( I need it for Realtime) First of all, when I make everything appears to compile ok with no errors however the res_config_mysql doesn't get compiled. So I tried "make res_config_mysql" and a whackload of errors starting with the following: # make res_config_mysql gcc -g

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

Hi everyone, IAX2 softphone is not working with the “Unregistered” part in the asterisk (chan_iax2.c and data_pgsql.c) But with the Xlite softphone the unregistered worked properly and ast_data properly updated the IP address and port number in the database. I have seen some codes in the chan_iax2.c file: “ast_verbose(VERBOSE_PREFIX_3 "Unregistered '%s' (%s)\n", p->name,

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

I am trying to write a customized app using C that would fetch voice file from SQL Server 2000 using ODBC and FREETDS. Currently I am only able to fetch first 63 KB chunk from the DB, and not able to fetch the rest of the file, below is the code that i am using to do so, fd = open(fullpath, O_RDWR | O_CREAT | O_TRUNC, 0770); if (fd < 0) { ast_log(LOG_WARNING, "Failed to write

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps

JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen

The USA is on DST now, but Europe is not. If you are in Europe, be aware that the VoIP Users Conference conference will start one hour early today. In Paris, that translates to GMT+1 or 5PM, in the UK 4PM. Grand Central is about to be re-branded as Google Voice. http://www.google.com/voice Changes should be announced soon. I logged in but see no difference yet. FWIW, Google says it'll still