Adds an option to sip.conf that prevents diversion headers from being added.

send_diversion=no will prevent Diversion headers from being added to SIP
requests. This doesn't prevent Diversion from being added with dialplan
such as with SIPAddHeader.

(closes issue ASTERISK-16862)
Reported by: rsw686
Review: https://reviewboard.asterisk.org/r/1769/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -65,6 +65,8 @@ SIP Changes
    which set the force_rport and comedia options automatically if Asterisk
    detects that an incoming SIP request crossed a NAT after being sent by
    the remote endpoint.
+ * Adds an option send_diversion which can be disabled to prevent
+   diversion headers from automatically being added to invites.
 
 Chan_local changes
 ------------------

channels/chan_sip.c

@@ -12542,6 +12542,11 @@ static void add_diversion_header(struct sip_request *req, struct sip_pvt *pvt)
 	const char *reason;
 	char header_text[256];
 
+	/* We skip this entirely if the configuration doesn't allow diversion headers */
+	if (!sip_cfg.send_diversion) {
+		return;
+	}
+
 	if (!pvt->owner) {
 		return;
 	}
@@ -18827,6 +18832,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
 	ast_cli(a->fd, "  Trust RPID:             %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_TRUSTRPID)));
 	ast_cli(a->fd, "  Send RPID:              %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_SENDRPID)));
 	ast_cli(a->fd, "  Legacy userfield parse: %s\n", AST_CLI_YESNO(sip_cfg.legacy_useroption_parsing));
+	ast_cli(a->fd, "  Send Diversion:         %s\n", AST_CLI_YESNO(sip_cfg.send_diversion));
 	ast_cli(a->fd, "  Caller ID:              %s\n", default_callerid);
 	if ((default_fromdomainport) && (default_fromdomainport != STANDARD_SIP_PORT)) {
 		ast_cli(a->fd, "  From: Domain:           %s:%d\n", default_fromdomain, default_fromdomainport);
@@ -29166,6 +29172,7 @@ static int reload_config(enum channelreloadreason reason)
 	sip_set_default_format_capabilities(sip_cfg.caps);
 	sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
 	sip_cfg.legacy_useroption_parsing = DEFAULT_LEGACY_USEROPTION_PARSING;
+	sip_cfg.send_diversion = DEFAULT_SEND_DIVERSION;
 	sip_cfg.notifyringing = DEFAULT_NOTIFYRINGING;
 	sip_cfg.notifycid = DEFAULT_NOTIFYCID;
 	sip_cfg.notifyhold = FALSE;		/*!< Keep track of hold status for a peer */
@@ -29467,6 +29474,8 @@ static int reload_config(enum channelreloadreason reason)
 			sip_cfg.regextenonqualify = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "legacy_useroption_parsing")) {
 			sip_cfg.legacy_useroption_parsing = ast_true(v->value);
+		} else if (!strcasecmp(v->name, "send_diversion")) {
+			sip_cfg.send_diversion = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "callerid")) {
 			ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
 		} else if (!strcasecmp(v->name, "mwi_from")) {

channels/sip/include/sip.h

@@ -219,6 +219,7 @@
 #define DEFAULT_ACCEPT_OUTOFCALL_MESSAGE TRUE
 #define DEFAULT_REGEXTENONQUALIFY FALSE
 #define DEFAULT_LEGACY_USEROPTION_PARSING FALSE
+#define DEFAULT_SEND_DIVERSION TRUE
 #define DEFAULT_T1MIN             100   /*!< 100 MS for minimal roundtrip time */
 #define DEFAULT_MAX_CALL_BITRATE (384)  /*!< Max bitrate for video */
 #ifndef DEFAULT_USERAGENT
@@ -733,6 +734,7 @@ struct sip_settings {
 	int callevents;             /*!< Whether we send manager events or not */
 	int regextenonqualify;      /*!< Whether to add/remove regexten when qualifying peers */
 	int legacy_useroption_parsing; /*!< Whether to strip useroptions in URI via semicolons */
+	int send_diversion;	        /*!< Whether to Send SIP Diversion headers */
 	int matchexternaddrlocally;   /*!< Match externaddr/externhost setting against localnet setting */
 	char regcontext[AST_MAX_CONTEXT];  /*!< Context for auto-extensions */
 	char messagecontext[AST_MAX_CONTEXT];  /*!< Default context for out of dialog msgs. */

configs/sip.conf.sample

@@ -476,6 +476,11 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                                     ; user options for whatever reason.  The behavior is similar to
                                                     ; how SIP URI's were typically handled in 1.6.2, hence the name.
 
+;send_diversion=no              ; Default "yes"     ; Asterisk normally sends Diversion headers with certain SIP
+                                                    ; invites to relay data about forwarded calls. If this option
+                                                    ; is disabled, Asterisk won't send Diversion headers unless
+                                                    ; they are added manually.
+
 ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
 ; in square brackets.  For example, the caller id value 555.5555 becomes 5555555
 ; when this option is enabled.  Disabling this option results in no modification