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

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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
This commit is contained in:
Mark Michelson
2012-05-21 19:05:52 +00:00
parent 3ef6bb1d21
commit 65e49d0ae4
3 changed files with 17 additions and 2 deletions
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17
channels/chan_sip.c
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@@ -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