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authorMark Michelson <mmichelson@digium.com>2012-05-21 19:22:25 +0000
committerMark Michelson <mmichelson@digium.com>2012-05-21 19:22:25 +0000
commite5f1f0496a0abdf472e81f1f818a2a942635b64a (patch)
tree3fbeabb7bd1bffd54ba43dc8a8d5afcc003f7ad4 /channels
parent45149bfdf8e7116e43f2de967a256439872f9e25 (diff)
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
Diffstat (limited to 'channels')
-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