- Implementing the new SIP transfer data structure

- Changing SIP call to support sending INVITEs as part of call transfers

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20902 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d06d436c374eb69f7f607f2c72bd466259440c3b..26a7809c8dee5023e4afc41480f0ac6239ec96cc 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -708,6 +708,24 @@ static const struct c_referstatusstring {
        { REFER_NOAUTH,         "Failed - auth failure" }
 } ;
 
+/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed  */
+/* OEJ: Should be moved to string fields */
+struct sip_refer {
+       char refer_to[AST_MAX_EXTENSION];               /*!< Place to store REFER-TO extension */
+       char refer_to_domain[AST_MAX_EXTENSION];        /*!< Place to store REFER-TO domain */
+       char refer_to_urioption[AST_MAX_EXTENSION];     /*!< Place to store REFER-TO uri options */
+       char refer_to_context[AST_MAX_EXTENSION];       /*!< Place to store REFER-TO context */
+       char referred_by[AST_MAX_EXTENSION];            /*!< Place to store REFERRED-BY extension */
+       char referred_by_name[AST_MAX_EXTENSION];       /*!< Place to store REFERRED-BY extension */
+       char refer_contact[AST_MAX_EXTENSION];          /*!< Place to store Contact info from a REFER extension */
+       char replaces_callid[BUFSIZ];                   /*!< Replace info */
+       char replaces_callid_totag[BUFSIZ/2];           /*!< Replace info */
+       char replaces_callid_fromtag[BUFSIZ/2];         /*!< Replace info */
+       struct sip_pvt *refer_call;                     /*!< Call we are referring */
+       int attendedtransfer;                           /*!< Attended or blind transfer? */
+       int localtransfer;                              /*!< Transfer to local domain? */
+       enum referstatus status;                        /*!< REFER status */
+};
 
 /*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe  */
 static struct sip_pvt {
@@ -806,6 +824,7 @@ static struct sip_pvt {
        int laststate;                          /*!< SUBSCRIBE: Last known extension state */
        int dialogver;                          /*!< SUBSCRIBE: Version for subscription dialog-info */
        
+       struct sip_refer *refer;                /*!< REFER: SIP transfer data structure */
        struct ast_dsp *vad;                    /*!< Voice Activation Detection dsp */
        
        struct sip_peer *relatedpeer;           /*!< If this dialog is related to a peer, which one 
@@ -2184,8 +2203,6 @@ static int auto_congest(void *nothing)
 }
 
 
-
-
 /*! \brief Initiate SIP call from PBX 
  *      used from the dial() application      */
 static int sip_call(struct ast_channel *ast, char *dest, int timeout)
@@ -2194,7 +2211,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
        struct sip_pvt *p;
        struct varshead *headp;
        struct ast_var_t *current;
-       
+       const char *referer = NULL;   /* SIP refererer */       
+
        p = ast->tech_pvt;
        if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
                ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
@@ -2215,13 +2233,36 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
                } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
                        /* Check whether there is a variable with a name starting with SIPADDHEADER */
                        p->options->addsipheaders = 1;
+               } else if (!strcasecmp(ast_var_name(current),"SIPTRANSFER")) {
+                       /* This is a transfered call */
+                       p->options->transfer = 1;
+               } else if (!strcasecmp(ast_var_name(current),"SIPTRANSFER_REFERER")) {
+                       /* This is the referer */
+                       referer = ast_var_value(current);
+               } else if (!strcasecmp(ast_var_name(current),"SIPTRANSFER_REPLACES")) {
+                       /* We're replacing a call. */
+                       p->options->replaces = ast_var_value(current);
                }
        }
        
        res = 0;
        ast_set_flag(&p->flags[0], SIP_OUTGOING);
+
+       if (p->options->transfer) {
+               char buf[BUFSIZ/2];
+
+               if (referer) {
+                       if (sipdebug && option_debug > 2)
+                               ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer);
+                       snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
+               } else {
+                       snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
+               }
+               ast_string_field_set(p, cid_name, buf);
+       } 
        if (option_debug)
                ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
+
        res = update_call_counter(p, INC_CALL_LIMIT);
        if ( res != -1 ) {
                p->callingpres = ast->cid.cid_pres;
@@ -5753,6 +5794,12 @@ static int transmit_message_with_text(struct sip_pvt *p, const char *text)
        return send_request(p, &req, 1, p->ocseq);
 }
 
+/*! \brief Allocate SIP refer structure */
+int sip_refer_allocate(struct sip_pvt *p) {
+   p->refer = ast_calloc(1, sizeof(struct sip_refer)); 
+   return p->refer ? 1 : 0;
+}
+
 /*! \brief Transmit SIP REFER message */
 static int transmit_refer(struct sip_pvt *p, const char *dest)
 {