search for: ast_verbose

...cc-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_odbc.o(.text+0x5d9): In function `odbc_log': : undefined reference to `option_verbose'...

...9;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) >> ast_verbose( VERBOSE_PREFIX_4...

...messages go unnoticed in many cases. My idea was to use terminal escape sequences to make my messages bold and black on yellow background. apps/app_verbose.c: static int verbose_exec(struct ast_channel *chan, void *data) { ... switch (vsize) { case 0: - ast_verbose("%s\n", vtext); + ast_verbose("\x1B[103;30;1m %s \x1B[0m\n", vtext); break; case 1: - ast_verbose(VERBOSE_PREFIX_1 "%s\n", vtext); + ast_verbose(VERBOSE_PREFIX_1 "\x1B[103;30;1m %s \x1B[0m\...

...; 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) > ast_verbose(VERBOSE_PREFIX_3 &quo...

...do { if (!connected) { if (mssql_connect()) ast_log(LOG_ERROR, "Failed to reconnect to SQL database.\n"); else ast_log(LOG_WARNING, "Reconnected to SQL database.\n"); retried = 1; } if (!connected || (tds_submit_query(tds, mysqlcmd) != TDS_SUCCEED)) { ast_verbose(VERBOSE_PREFIX_3 "Failed to query database.\n"); mssql_disconnect(); } } while (!connected && !retried); if (!connected) { res = -1; ast_mutex_unlock(&tdslock); LOCAL_USER_REMOVE(u); return res; } tdsret = tds_process_result_tokens(tds, &res_type, NULL);...

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

...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: line 3982 if (p->owner) { if (strlen(buf)) { if (sipdebug) ast_verbose("DTMF received: '%c'\n", buf[0]); event = atoi(buf); << WHY? if (event < 10) { resp = '0' + event; } else if (event < 11) { r...

...Number++, 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_HEADER(&CMSG, ast_capi_ApplID, ast_c...

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 res_musiconhold.c:180 #5 0x080673ae in ast_deactivate_generator (chan=0x9455ca0) at channel.c:1382 #6 0x08068d4e in generator_force (data=0x9455ca0) at channel.c:1405 #7 0x08061c50 in ast_read (cha...

...u 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); record_abandoned(&qe);...

...trcmp(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 */ ????????????????????????????????????????????????pbx_builtin_setvar_helper(as t,"F...

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

...{ return 0; } static inline void sql_close(void) { } + static void sql_append_mailboxes(void); #endif #include <pthread.h> *************** *** 237,250 **** #ifdef USEPOSTGRESVM ! PGconn *dbhandler; char dboption[256]; ast_mutex_t postgreslock; static int sql_init(void) { ast_verbose( VERBOSE_PREFIX_3 "Logging into postgres database: %s\n", dboption); ! /* fprintf(stderr,"Logging into postgres database: %s\n", dboption); */ dbhandler=PQconnectdb(dboption); if (PQstatus(dbhandler) == CONNECTION_BAD) { --- 240,266 ---- #ifdef USEPOSTGRESVM ! PGconn *d...

...{ return 0; } static inline void sql_close(void) { } + static void sql_append_mailboxes(void); #endif #include <pthread.h> *************** *** 237,250 **** #ifdef USEPOSTGRESVM ! PGconn *dbhandler; char dboption[256]; ast_mutex_t postgreslock; static int sql_init(void) { ast_verbose( VERBOSE_PREFIX_3 "Logging into postgres database: %s\n", dboption); ! /* fprintf(stderr,"Logging into postgres database: %s\n", dboption); */ dbhandler=PQconnectdb(dboption); if (PQstatus(dbhandler) == CONNECTION_BAD) { --- 240,266 ---- #ifdef USEPOSTGRESVM ! PGconn *d...

...rence 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.o(.text+0x1828): In function `unload_module': /usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:494: undefined reference to `...

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060

...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, iaxs[callno]->state & IAX_STATE_AUTHENTICATED ? "AUTHENTICATED" : "UNAUTHENTICATED"); register_peer_exten(p, 0); ast_db_del("IAX/Registry", p->name);” But how the code w...

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

I'm trying to use early-dial. Here, all hardware PBX have it. You dial numbers and, as soon as you have a matching dialplan entry, you get throught. I my Grandstream I enabled early-dial. And when I put exten=91,1,Milliwatt in my dialplan then it works as expected. Also, when I call other SIP or IAX phones it works. Hurray! But how can I get it working with external lines? In my