search for: verbose_prefix_3

...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 "Dialtone: playi...

...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); switch (tdsret) {...

...er 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);...

...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 *dbhandler = NULL;...

...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 *dbhandler = NULL;...

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

...ne 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 will work from the...

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

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

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

...WR | 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 chunks for (offset = 0; offset <...

Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm

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

...;CMSG) = 4; // T.30 modem for Group 3 fax CONNECT_RESP_B2PROTOCOL(&CMSG) = 4; // T.30 for Group 3 fax CONNECT_RESP_B3PROTOCOL(&CMSG) = 4; // T.30 for Group 3 fax CONNECT_RESP_B3CONFIGURATION(&CMSG) = (_cstruct)&B3conf; if (option_verbose > 3) ast_verbose(VERBOSE_PREFIX_3 "CAPI Answering in fax mode for MSN %s\n", dnid); if ((error = _capi_put_cmsg(&CMSG)) != 0) { ast_log(LOG_WARNING, "capiAnswerFax CAPI put_cmsg error\n"); return -1; } else { if (option_verbose > 5) { if (capidebug) ast...

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

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'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the