Add "send to voicemail" Digium phone functionality to Asterisk.

This change accommodates two methods by which calls can be directed to
a user's voicemail.

* Incoming calls can be redirected to any user's voicemail.
* Established calls can be blind transferred to any user's voicemail.

Digium phones indicate the desire to direct a call to voicemail by using
a Diversion header with a reason parameter of "send_to_vm".

This patch adds the "send_to_vm" reason as a valid redirecting reason. In
addition, chan_sip.c has been modified to update redirecting information
on the transferred channel by reading a Diversion header on a REFER request.

(closes issue AST-871)
Reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/1925

git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/1.8.11@367161 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index dd4deccb6bbfbb6992dd9ad361b5cd1943659149..4821b856295229544e9ebb5fc7e2937ba6c56b3b 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -683,7 +683,8 @@ static const struct sip_reasons {
        { AST_REDIRECTING_REASON_FOLLOW_ME, "follow-me" },
        { AST_REDIRECTING_REASON_OUT_OF_ORDER, "out-of-service" },
        { AST_REDIRECTING_REASON_AWAY, "away" },
-       { AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"}
+       { AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"},
+       { AST_REDIRECTING_REASON_SEND_TO_VM, "send_to_vm"},
 };
 
 
@@ -15390,7 +15391,7 @@ static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq, char **name, c
                        }
                        /* Remove enclosing double-quotes */
                        if (*reason_param == '"')
-                               ast_strip_quoted(reason_param, "\"", "\"");
+                               reason_param = ast_strip_quoted(reason_param, "\"", "\"");
                        if (!ast_strlen_zero(reason_param)) {
                                sip_set_redirstr(p, reason_param);
                                if (p->owner) {
@@ -23673,6 +23674,8 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
        int localtransfer = 0;
        int attendedtransfer = 0;
        int res = 0;
+       struct ast_party_redirecting redirecting;
+       struct ast_set_party_redirecting update_redirecting;
 
        if (req->debug) {
                ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n",
@@ -23977,6 +23980,16 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
        }
        ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);  /* Delay hangup */
 
+       /* When a call is transferred to voicemail from a Digium phone, there may be
+        * a Diversion header present in the REFER with an appropriate reason parameter
+        * set. We need to update the redirecting information appropriately.
+        */
+       ast_party_redirecting_init(&redirecting);
+       memset(&update_redirecting, 0, sizeof(update_redirecting));
+       change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE);
+       ast_channel_update_redirecting(current.chan2, &redirecting, &update_redirecting);
+       ast_party_redirecting_free(&redirecting);
+
        /* Do not hold the pvt lock during the indicate and async_goto. Those functions
         * lock channels which will invalidate locking order if the pvt lock is held.*/
        /* For blind transfers, move the call to the new extensions. For attended transfers on multiple

diff --git a/include/asterisk/callerid.h b/include/asterisk/callerid.h
index e052f64781625a5a511d43cb06e3088747dedff9..eb25d27125f2cf69673af6467a5534ec364c8f13 100644 (file)
--- a/include/asterisk/callerid.h
+++ b/include/asterisk/callerid.h
@@ -400,6 +400,7 @@ enum AST_REDIRECTING_REASON {
        AST_REDIRECTING_REASON_OUT_OF_ORDER,
        AST_REDIRECTING_REASON_AWAY,
        AST_REDIRECTING_REASON_CALL_FWD_DTE,           /* This is something defined in Q.931, and no I don't know what it means */
+       AST_REDIRECTING_REASON_SEND_TO_VM,
 };
 
 /*!

diff --git a/main/callerid.c b/main/callerid.c
index 3af85b467b3cfb8dc43b73b8176910df7ba145ed..d9c65c416170804bc7def13474b52ae34c7df311 100644 (file)
--- a/main/callerid.c
+++ b/main/callerid.c
@@ -1203,6 +1203,7 @@ static const struct ast_value_translation redirecting_reason_types[] = {
        { AST_REDIRECTING_REASON_OUT_OF_ORDER,   "out_of_order", "Called DTE Out-Of-Order" },
        { AST_REDIRECTING_REASON_AWAY,           "away",         "Callee is Away" },
        { AST_REDIRECTING_REASON_CALL_FWD_DTE,   "cf_dte",       "Call Forwarding By The Called DTE" },
+       { AST_REDIRECTING_REASON_SEND_TO_VM,     "send_to_vm",   "Call is being redirected to user's voicemail"},
 /* *INDENT-ON* */
 };