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

diff --git a/CHANGES b/CHANGES
index b0a9edd..b728626 100644 (file)
--- a/CHANGES
+++ b/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
 ------------------

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 417e7e3..657844d 100644 (file)
--- a/channels/chan_sip.c
+++ b/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")) {

diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 2f9beff..d9f99d6 100644 (file)
--- a/channels/sip/include/sip.h
+++ b/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. */

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index f64dc04..d546106 100644 (file)
--- a/configs/sip.conf.sample
+++ b/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