From e5f1f0496a0abdf472e81f1f818a2a942635b64a Mon Sep 17 00:00:00 2001
From: Mark Michelson <mmichelson@digium.com>
Date: Mon, 21 May 2012 19:22:25 +0000
Subject: [PATCH] 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/trunk@367163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 15 ++++++++++++++-
include/asterisk/callerid.h | 1 +
main/callerid.c | 1 +
3 files changed, 16 insertions(+), 1 deletion(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a84adeb75e..d34d2718e3 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
diff --git a/include/asterisk/callerid.h b/include/asterisk/callerid.h
index c047632b99..7c4905e13d 100644
--- 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 dc3a91093a..37edd992ca 100644
--- 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* */
};
--
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