[asterisk-commits] rizzo: branch rizzo/astobj2 r76215 - /team/rizzo/astobj2/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: rizzo
Date: Fri Jul 20 16:52:27 2007
New Revision: 76215

URL: http://svn.digium.com/view/asterisk?view=rev&rev=76215
Log:
more diff reduction with trunk. mostly whitespace, but also
some minor bugfixes

Modified:
    team/rizzo/astobj2/channels/chan_sip.c

Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=76215&r1=76214&r2=76215
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Fri Jul 20 16:52:27 2007
@@ -2714,12 +2714,13 @@
 	\return pointer to terminated stripped string
 	\param tmp input string that will be modified
 	Examples:
-
+\verbatim
 	"foo" <bar>	valid input, returns bar
 	foo		returns the whole string
 	< "foo ... >	returns the string between brackets
 	< "foo...	bogus (missing closing bracket), returns the whole string
 			XXX maybe should still skip the opening bracket
+\endverbatim
  */
 static char *get_in_brackets(char *tmp)
 {
@@ -2768,7 +2769,9 @@
  * Init pointers to empty string so we never get NULL dereferencing.
  * Overwrites the string.
  * return 0 on success, other values on error.
+\verbatim
  * general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...] 
+\endverbatim
  */
 static int parse_uri(char *uri, char *scheme,
 	char **ret_name, char **pass, char **domain, char **port, char **options)
@@ -2783,7 +2786,7 @@
 		*port = "";
 	if (scheme) {
 		int l = strlen(scheme);
-		if (!strncmp(uri, scheme, l))
+		if (!strncasecmp(uri, scheme, l))
 			uri += l;
 		else {
 			ast_log(LOG_NOTICE, "Missing scheme '%s' in '%s'\n", scheme, uri);
@@ -2868,6 +2871,7 @@
 
 	time_t nowtime = time(NULL) + expirey;
 	const char *fc = fullcontact ? "fullcontact" : NULL;
+
 	int realtimeregs = ast_check_realtime("sipregs");
 
 	tablename = realtimeregs ? "sipregs" : "sippeers";
@@ -3480,6 +3484,7 @@
 		/* Initialize auto-congest time */
 		p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, pvt_ref(p));
 	}
+
 	return res;
 }
 
@@ -3666,9 +3671,10 @@
 
 	ast_debug(3, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
 
+
 	/* Test if we need to check call limits, in order to avoid 
 	   realtime lookups if we do not need it */
-	if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
+	if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT) && !ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD))
 		return 0;
 
 	ast_copy_string(name, fup->username, sizeof(name));
@@ -3706,7 +3712,8 @@
 		/* Decrement onhold count if applicable */
 		if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && global_notifyhold)
 			sip_peer_hold(fup, FALSE);
-		ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
+		if (sipdebug)
+			ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
 		break;
 
 	case INC_CALL_RINGING:
@@ -3929,7 +3936,7 @@
 	}
 
 	if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
-		if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
+		if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
 			if (sipdebug)
 				ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
 			update_call_counter(p, DEC_CALL_LIMIT);
@@ -3948,11 +3955,12 @@
 		ast_debug(1, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
 	else
 		ast_debug(1, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid);
+
 	if (ast_test_flag(ast, AST_FLAG_ZOMBIE)) 
 		ast_debug(1, "Hanging up zombie call. Be scared.\n");
 
 	sip_pvt_lock(p);
-	if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
+	if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
 		if (sipdebug)
 			ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
 		update_call_counter(p, DEC_CALL_LIMIT);
@@ -3964,6 +3972,7 @@
 		sip_pvt_unlock(p);
 		return 0;
 	}
+	/* If the call is not UP, we need to send CANCEL instead of BYE */
 	/* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
 	 if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) {
 		needcancel = TRUE;
@@ -4536,6 +4545,7 @@
 	ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
 	ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
 
+
 	/* Don't use ast_set_callerid() here because it will
 	 * generate an unnecessary NewCallerID event  */
 	tmp->cid.cid_num = ast_strdup(i->cid_num);
@@ -4556,15 +4566,17 @@
 		pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
 	if (i->rtp)
 		ast_jb_configure(tmp, &global_jbconf);
+
+	/* Set channel variables for this call from configuration */
+	for (v = i->chanvars ; v ; v = v->next)
+		pbx_builtin_setvar_helper(tmp,v->name,v->value);
+
 	if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
 		ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
 		tmp->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
 		ast_hangup(tmp);
 		tmp = NULL;
 	}
-	/* Set channel variables for this call from configuration */
-	for (v = i->chanvars ; v ; v = v->next)
-		pbx_builtin_setvar_helper(tmp,v->name,v->value);
 
 	append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);