search for: option_verbose

...at I get: # cd /usr/src/asterisk/cdr # make cc -o cdr_odbc.so cdr_odbc.o -lodbc /usr/lib/gcc-lib/i386-redhat-linux/3.3.2/../../../crt1.o(.text+0x18): In function `_start': : undefined reference to `main' cdr_odbc.o(.text+0x197): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x1ae): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x251): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x268): In function `odbc_log': : undefined reference to `ast_verbose' cdr...

...r_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( VERBOSE_PREFIX_4 "cdr_odbc: Error SQLConnect %d\n", ODBC_res); SQLGetDiagRec(SQL_HANDLE_DBC, ODBC_con, 1, ODBC_stat, &ODBC_err, ODBC_msg, 100, &ODBC_mlen); >> if(option_verbose > 10) >>...

...mailbox = strsep(&stringp, "|"); > timeout = atoi(stimeout) * 1000; > } > > > if(mailbox && ast_app_has_voicemail(mailbox)) { > ts = ast_get_indication_tone(chan->zone, "dialrecall"); > if (option_verbose > 2) > ast_verbose(VERBOSE_PREFIX_3 "Dialtone: playing dialrecall for %d mili seconds\n", timeout); > } else { > ts = ast_get_indication_tone(chan->zone, "dial"); > if (option_verbose > 2) >...

...p;CMSG, ast_capi_ApplID, ast_capi_MessageNumber++, 0); DISCONNECT_B3_REQ_NCCI(&CMSG) = i->NCCI; if ((error = _capi_put_cmsg(&CMSG)) != 0) { ast_log(LOG_ERROR, "error sending DISCONNECT_B3_REQ NCCI=%#x\n",i->NCCI); } else { if (option_verbose > 5) { ast_verbose(VERBOSE_PREFIX_4 "sent DISCONNECT_B3_REQ NCCI=%#x\n",i->NCCI); } } // wait for the B3 layer to go down while (i->state != CAPI_STATE_CONNECTED) { usleep(10000); } SELECT_B_PROTOCOL_REQ_HE...

...28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose I got nothing error during installation of asterisk-addons-1.4.2 after I had change the Make file on the chan_ooh323.so.1.0.1. Tried; I tried to define noload to the chan_00h323.so and res_config_mysql.so, my asterisk start but give me others problems as bellowing... [Jun 28 17:03:12] WA...

...eal with the 'n' option. If you don't pass 'n', you should never see it. 1.4.8 code: /* exit after 'timeout' cycle if 'n' option enabled */ if (go_on) { if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3 "Exiting on time-out cycle\n"); ast_queue_log(args.queuename, chan->uniqueid, "NONE", "EXITWITHTIMEOUT", "%d", qe.pos);...

...??????????????????????p->faxhandled++; ????????????????????????????????if (strcmp(ast->exten, "fax")) { ????????????????????????????????????????if (ast_exists_extension(ast, ast->context, "fax", 1, ast->callerid)) { ????????????????????????????????????????????????if (option_verbose > 2) ????????????????????????????????????????????????????????ast_verbose(VERBOSE_ PREFIX_3 "Redirecting %s to fax extension\n", ast->name); ????????????????????????????????????????????????/* Save the DID/DNIS when we transfer the fax call to a "fax" extension */ ??????????...

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Nope ! This is the one that tries to include PRE 1.0.X header file <parking.h>. It cannot compile on * 1.0.X (I have tried also to include <features.h> instead of <parking.h> (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad > > Try this > >> Since www.bkw.org seems not to exist anymore (getting

...asterisk -cvg from /opt/asterisk/usr/sbin), I get the following error: (snip) Asterisk Dynamic Loader Starting: [res_musiconhold.so]Mar 28 09:23:48 WARNING[18299]: loader.c:325 __load_resource: ld.so.1: ./asterisk: fatal: relocation error: file /opt/asterisk/lib/modules/res_musiconhold.so: symbol option_verbose: referenced symbol not found Mar 28 09:23:48 WARNING[18299]: loader.c:499 load_modules: Loading module res_musiconhold.so failed! # I am not entirely sure what this error means. I tried adding /opt/asterisk/lib/modules to the system's dynamic linker path using crle, but that didn't seem to...

Hi List, I am currently using the H parameter in the dial command. The issue that I am having is that if the user is calling an ivr that requires him to press the * key then the call gets hung up on. How would I go about changing it so that the user will have to press say ** for the H parameter to come in to effect ? Thanks a lot. Dovid -------------- next part -------------- An HTML attachment

...30257] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 23 12:10:01] WARNING[30257] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose "" If I manually disable the modules in the modules.conf then my Asterisk 1.4.5 will run. I am using belwoing release; Asterisk 1.4.5 Zaptel 1.4.3 Asterisk-addons 1.4.2 Libpri 1.4.0 Thank you in advance if some one can help. -------------- next part ------------...

Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P

...28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose I got nothing error during installation of asterisk-addons-1.4.2 after I had change the Make file on the chan_ooh323.so.1.0.1. Tried; I tried to define noload to the chan_00h323.so and res_config_mysql.so, my asterisk start but give me others problems as bellowing... [Jun 28 17:03:12] WA...

Hi there, Besides the problem of Voicetronix dialing too early before the carrier gives a dial tone, there also appears to be issues with the audio quality when more than 1 channel is utilized.

...to do so, fd = open(fullpath, O_RDWR | O_CREAT | O_TRUNC, 0770); if (fd < 0) { ast_log(LOG_WARNING, "Failed to write '%s': %s\n", fullpath, strerror(errno)); res = -1; goto free_res; } res = SQLGetData(stmt, 1, SQL_BINARY, empty, 0, &colsize); fdlen = colsize; if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3 "COLSIZE = %d", colsize); //PRINTING COLSIZE ON CLI if (fd > -1) { char tmp[1]=""; lseek(fd, fdlen - 1, SEEK_SET); if (write(fd, tmp, 1) != 1) { close(fd); res = -1; goto free_res; } } if (fd > -1){ //Trying to fetch data in...

...ed reference to `ast_log' res_config_mysql.o(.text+0x17a6):/usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:478: undefined reference to `ast_config_engine_register' res_config_mysql.o(.text+0x17af):/usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:479: undefined reference to `option_verbose' res_config_mysql.o(.text+0x17c3):/usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:480: undefined reference to `ast_verbose' res_config_mysql.o(.text+0x17d5):/usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:482: undefined reference to `ast_cli_register' res_config_mysql....

List, Does anyone know how to limit the amount of registrations that a sip user can have? For example, I have 2 softphones that I use on my laptop & desktop, both use the same username & password. If I have both softphones up at the same time, I can make simultaneous calls with each of them. I know you can have call-limit=1 but in this case, I want to allow them to have 3 way calling

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension