diff options
author | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2003-01-17 05:10:52 +0000 |
---|---|---|
committer | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2003-01-17 05:10:52 +0000 |
commit | 47ff9c54029a3012c7112e808a1c435e811c9823 (patch) | |
tree | 97bb6b30f6a8ce0e16ac2498a38afefdac76049e | |
parent | 1ef4f09fcfea211e2c5db9106d2270cf4a6a75c4 (diff) |
Version 0.3.0 from FTP
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@585 f38db490-d61c-443f-a65b-d21fe96a405b
-rwxr-xr-x | apps/app_dial.c | 122 |
1 files changed, 109 insertions, 13 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index ee49dc2bd..754895f4d 100755 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -1,7 +1,7 @@ /* * Asterisk -- A telephony toolkit for Linux. * - * Trivial application to dial a channel + * Trivial application to dial a channel and send an URL on answer * * Copyright (C) 1999, Mark Spencer * @@ -22,6 +22,7 @@ #include <asterisk/say.h> #include <asterisk/parking.h> #include <asterisk/musiconhold.h> +#include <asterisk/callerid.h> #include <stdlib.h> #include <errno.h> #include <unistd.h> @@ -41,7 +42,7 @@ static char *app = "Dial"; static char *synopsis = "Place an call and connect to the current channel"; static char *descrip = -" Dial(Technology/resource[&Technology2/resource2...][|timeout][|options]):\n" +" Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):\n" "Requests one or more channels and places specified outgoing calls on them.\n" "As soon as a channel answers, the Dial app will answer the originating\n" "channel (if it needs to be answered) and will bridge a call with the channel\n" @@ -63,8 +64,11 @@ static char *descrip = " 'm' -- provide hold music to the calling party until answered.\n" " 'd' -- data-quality (modem) call (minimum delay).\n" " 'H' -- allow caller to hang up by hitting *.\n" +" 'P[(x)]' -- privacy mode, using 'x' as database if provided.\n" " In addition to transferring the call, a call may be parked and then picked\n" -"up by another user.\n"; +"up by another user.\n" +" The optionnal URL will be sent to the called party if the channel supports\n" +"it.\n"; /* We define a customer "local user" structure because we use it not only for keeping track of what is in use but @@ -160,6 +164,12 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu ast_verbose( VERBOSE_PREFIX_2 "No one is available to answer at this time\n"); } *to = 0; + /* if no one available we'd better stop MOH/ringing to */ + if (moh) { + ast_moh_stop(in); + } else if (ringind) { + ast_indicate(in, -1); + } return NULL; } winner = ast_waitfor_n(watchers, pos, to); @@ -210,6 +220,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu if (!sentringing) { ast_indicate(in, AST_CONTROL_RINGING); sentringing++; + ringind++; } break; case AST_CONTROL_OFFHOOK: @@ -277,15 +288,22 @@ static int dial_exec(struct ast_channel *chan, void *data) int res=-1; struct localuser *u; char info[256], *peers, *timeout, *tech, *number, *rest, *cur; + char privdb[256] = "", *s; struct localuser *outgoing=NULL, *tmp; struct ast_channel *peer; int to; int allowredir=0; int allowdisconnect=0; + int privacy=0; char numsubst[AST_MAX_EXTENSION]; char restofit[AST_MAX_EXTENSION]; char *transfer = NULL; char *newnum; + char callerid[256], *l, *n; + char *url=NULL; /* JDG */ + struct ast_var_t *current; + struct varshead *headp, *newheadp; + struct ast_var_t *newvar; if (!data) { ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout)\n"); @@ -306,6 +324,15 @@ static int dial_exec(struct ast_channel *chan, void *data) if (transfer) { *transfer = '\0'; transfer++; + /* JDG */ + url = strchr(transfer, '|'); + if (url) { + *url = '\0'; + url++; + ast_log(LOG_DEBUG, "DIAL WITH URL=%s_\n", url); + } else + ast_log(LOG_DEBUG, "SIMPLE DIAL (NO URL)\n"); + /* /JDG */ } } } else @@ -315,6 +342,46 @@ static int dial_exec(struct ast_channel *chan, void *data) goto out; } + + if (transfer) { + /* Extract privacy info from transfer */ + if ((s = strstr(transfer, "P("))) { + privacy = 1; + strncpy(privdb, s + 2, sizeof(privdb) - 1); + /* Overwrite with X's what was the privacy info */ + while(*s && (*s != ')')) + *(s++) = 'X'; + if (*s) + *s = 'X'; + /* Now find the end of the privdb */ + s = strchr(privdb, ')'); + if (s) + *s = '\0'; + else { + ast_log(LOG_WARNING, "Transfer with privacy lacking trailing '('\n"); + privacy = 0; + } + } else if (strchr(transfer, 'P')) { + /* No specified privdb */ + privacy = 1; + } + } + if (!strlen(privdb) && privacy) { + /* If privdb is not specified and we are using privacy, copy from extension */ + strncpy(privdb, chan->exten, sizeof(privdb) - 1); + } + if (privacy) { + if (chan->callerid) + strncpy(callerid, chan->callerid, sizeof(callerid)); + else + strcpy(callerid, ""); + ast_callerid_parse(callerid, &n, &l); + if (l) { + ast_shrink_phone_number(l); + } else + l = ""; + ast_log(LOG_NOTICE, "Privacy DB is '%s', privacy is %d, clid is '%s'\n", privdb, privacy, l); + } cur = peers; do { /* Remember where to start next time */ @@ -341,14 +408,19 @@ static int dial_exec(struct ast_channel *chan, void *data) if (transfer) { if (strchr(transfer, 't')) tmp->allowredirect = 1; + else tmp->allowredirect = 0; if (strchr(transfer, 'r')) tmp->ringbackonly = 1; + else tmp->ringbackonly = 0; if (strchr(transfer, 'm')) tmp->musiconhold = 1; + else tmp->musiconhold = 0; if (strchr(transfer, 'd')) tmp->dataquality = 1; + else tmp->dataquality = 0; if (strchr(transfer, 'H')) tmp->allowdisconnect = 1; + else tmp->allowdisconnect = 0; } strncpy(numsubst, number, sizeof(numsubst)-1); /* If we're dialing by extension, look at the extension to know what to dial */ @@ -382,6 +454,23 @@ static int dial_exec(struct ast_channel *chan, void *data) cur = rest; break; } + /* If creating a SIP channel, look for a variable called */ + /* VXML_URL in the calling channel and copy it to the */ + /* new channel. */ + if (strcasecmp(tech,"SIP")==0) + { + headp=&chan->varshead; + AST_LIST_TRAVERSE(headp,current,entries) { + if (strcasecmp(ast_var_name(current),"VXML_URL")==0) + { + newvar=ast_var_assign(ast_var_name(current),ast_var_value(current)); + newheadp=&tmp->chan->varshead; + AST_LIST_INSERT_HEAD(newheadp,newvar,entries); + break; + } + } + } + tmp->chan->appl = "AppDial"; tmp->chan->data = "(Outgoing Line)"; tmp->chan->whentohangup = 0; @@ -445,6 +534,18 @@ static int dial_exec(struct ast_channel *chan, void *data) /* Ah ha! Someone answered within the desired timeframe. Of course after this we will always return with -1 so that it is hung up properly after the conversation. */ + if (!strcmp(chan->type,"Zap")) + { + int x = 2; + if (tmp->dataquality) x = 0; + ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0); + } + if (!strcmp(peer->type,"Zap")) + { + int x = 2; + if (tmp->dataquality) x = 0; + ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0); + } hanguptree(outgoing, peer); outgoing = NULL; /* If appropriate, log that we have a destination channel */ @@ -457,16 +558,11 @@ static int dial_exec(struct ast_channel *chan, void *data) ast_hangup(peer); return -1; } - if (!strcmp(chan->type,"Zap")) { - int x = 2; - if (tmp->dataquality) x = 0; - ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0); - } - if (!strcmp(peer->type,"Zap")) { - int x = 2; - if (tmp->dataquality) x = 0; - ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0); - } + /* JDG: sendurl */ + if( url && strlen(url) && ast_channel_supports_html(peer) ) { + ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url); + ast_channel_sendurl( peer, url ); + } /* /JDG */ res = ast_bridge_call(chan, peer, allowredir, allowdisconnect); ast_hangup(peer); } |