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-rw-r--r--channels/chan_sip.c15
1 files changed, 14 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a84adeb75..d34d2718e 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -671,7 +671,8 @@ 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"},
};
@@ -24257,6 +24258,8 @@ 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",
@@ -24561,6 +24564,16 @@ 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