Merge "res_pjsip: New endpoint option "refer_blind_progress""
[asterisk/asterisk.git] / res / res_pjsip_send_to_voicemail.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, Digium, Inc.
5  *
6  * Jonathan Rose <jrose@digium.com>
7  *
8  * See http://www.asterisk.org for more information about
9  * the Asterisk project. Please do not directly contact
10  * any of the maintainers of this project for assistance;
11  * the project provides a web site, mailing lists and IRC
12  * channels for your use.
13  *
14  * This program is free software, distributed under the terms of
15  * the GNU General Public License Version 2. See the LICENSE file
16  * at the top of the source tree.
17  */
18
19 /*! \file
20  *
21  * \brief Module for managing send to voicemail requests in SIP
22  *        REFER messages against PJSIP channels
23  *
24  * \author Jonathan Rose <jrose@digium.com>
25  */
26
27 /*** MODULEINFO
28          <depend>pjproject</depend>
29          <depend>res_pjsip</depend>
30          <depend>res_pjsip_session</depend>
31          <support_level>core</support_level>
32 ***/
33
34 #include "asterisk.h"
35
36 #include <pjsip.h>
37 #include <pjsip_ua.h>
38
39 #include "asterisk/pbx.h"
40 #include "asterisk/res_pjsip.h"
41 #include "asterisk/res_pjsip_session.h"
42 #include "asterisk/module.h"
43
44 #define DATASTORE_NAME "call_feature_send_to_vm_datastore"
45
46 #define SEND_TO_VM_HEADER "PJSIP_HEADER(add,X-Digium-Call-Feature)"
47 #define SEND_TO_VM_HEADER_VALUE "feature_send_to_vm"
48
49 #define SEND_TO_VM_REDIRECT "REDIRECTING(reason)"
50 #define SEND_TO_VM_REDIRECT_VALUE "send_to_vm"
51 #define SEND_TO_VM_REDIRECT_QUOTED_VALUE "\"" SEND_TO_VM_REDIRECT_VALUE "\""
52
53 static void send_response(struct ast_sip_session *session, int code, struct pjsip_rx_data *rdata)
54 {
55         pjsip_tx_data *tdata;
56
57         if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, code, NULL, &tdata) == PJ_SUCCESS) {
58                 struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
59
60                 pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
61         }
62 }
63
64 static void channel_cleanup_wrapper(void *data)
65 {
66         struct ast_channel *chan = data;
67         ast_channel_cleanup(chan);
68 }
69
70 static struct ast_datastore_info call_feature_info = {
71         .type = "REFER call feature info",
72         .destroy = channel_cleanup_wrapper,
73 };
74
75 static pjsip_param *get_diversion_reason(pjsip_fromto_hdr *hdr)
76 {
77         static const pj_str_t reason_str = { "reason", 6 };
78         return pjsip_param_find(&hdr->other_param, &reason_str);
79 }
80
81 static pjsip_fromto_hdr *get_diversion_header(pjsip_rx_data *rdata)
82 {
83         static const pj_str_t from_str = { "From", 4 };
84         static const pj_str_t diversion_str = { "Diversion", 9 };
85
86         pjsip_generic_string_hdr *hdr;
87         pj_str_t value;
88
89         if (!(hdr = pjsip_msg_find_hdr_by_name(
90                       rdata->msg_info.msg, &diversion_str, NULL))) {
91                 return NULL;
92         }
93
94         pj_strdup_with_null(rdata->tp_info.pool, &value, &hdr->hvalue);
95
96         /* parse as a fromto header */
97         return pjsip_parse_hdr(rdata->tp_info.pool, &from_str, value.ptr,
98                                pj_strlen(&value), NULL);
99 }
100
101 static int has_diversion_reason(pjsip_rx_data *rdata)
102 {
103         pjsip_param *reason;
104         pjsip_fromto_hdr *hdr = get_diversion_header(rdata);
105
106         if (!hdr) {
107                 return 0;
108         }
109         reason = get_diversion_reason(hdr);
110         return reason
111                 && (!pj_stricmp2(&reason->value, SEND_TO_VM_REDIRECT_QUOTED_VALUE)
112                         || !pj_stricmp2(&reason->value, SEND_TO_VM_REDIRECT_VALUE));
113 }
114
115 static int has_call_feature(pjsip_rx_data *rdata)
116 {
117         static const pj_str_t call_feature_str = { "X-Digium-Call-Feature", 21 };
118
119         pjsip_generic_string_hdr *hdr = pjsip_msg_find_hdr_by_name(
120                 rdata->msg_info.msg, &call_feature_str, NULL);
121
122         return hdr && !pj_stricmp2(&hdr->hvalue, SEND_TO_VM_HEADER_VALUE);
123 }
124
125 static int handle_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
126 {
127         struct ast_datastore *sip_session_datastore;
128         struct ast_channel *other_party;
129         int has_feature;
130         int has_reason;
131
132         if (!session->channel) {
133                 return 0;
134         }
135
136         has_feature = has_call_feature(rdata);
137         has_reason = has_diversion_reason(rdata);
138         if (!has_feature && !has_reason) {
139                 /* If we don't have a call feature or diversion reason or if
140                    it's not a feature this module is related to then there
141                    is nothing to do. */
142                 return 0;
143         }
144
145         /* Check bridge status... */
146         other_party = ast_channel_bridge_peer(session->channel);
147         if (!other_party) {
148                 /* The channel wasn't in a two party bridge */
149                 ast_log(LOG_WARNING, "%s (%s) attempted to transfer to voicemail, "
150                         "but was not in a two party bridge.\n",
151                         ast_sorcery_object_get_id(session->endpoint),
152                         ast_channel_name(session->channel));
153                 send_response(session, 400, rdata);
154                 return -1;
155         }
156
157         sip_session_datastore = ast_sip_session_alloc_datastore(
158                 &call_feature_info, DATASTORE_NAME);
159         if (!sip_session_datastore) {
160                 ast_channel_unref(other_party);
161                 send_response(session, 500, rdata);
162                 return -1;
163         }
164
165         sip_session_datastore->data = other_party;
166
167         if (ast_sip_session_add_datastore(session, sip_session_datastore)) {
168                 ao2_ref(sip_session_datastore, -1);
169                 send_response(session, 500, rdata);
170                 return -1;
171         }
172
173         if (has_feature) {
174                 pbx_builtin_setvar_helper(other_party, SEND_TO_VM_HEADER,
175                                           SEND_TO_VM_HEADER_VALUE);
176         }
177
178         if (has_reason) {
179                 pbx_builtin_setvar_helper(other_party, SEND_TO_VM_REDIRECT,
180                                           SEND_TO_VM_REDIRECT_VALUE);
181         }
182
183         ao2_ref(sip_session_datastore, -1);
184         return 0;
185 }
186
187 static void handle_outgoing_response(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
188 {
189         pjsip_status_line status = tdata->msg->line.status;
190         struct ast_datastore *feature_datastore =
191                 ast_sip_session_get_datastore(session, DATASTORE_NAME);
192         struct ast_channel *target_chan;
193
194         if (!feature_datastore) {
195                 return;
196         }
197
198         /* Since we are handling the response, there is no need to keep the datastore in the session anymore. */
199         ast_sip_session_remove_datastore(session, DATASTORE_NAME);
200
201         /* If the response >= 300, the refer failed and we need to clear the feature. */
202         if (status.code >= 300) {
203                 target_chan = feature_datastore->data;
204                 pbx_builtin_setvar_helper(target_chan, SEND_TO_VM_HEADER, NULL);
205                 pbx_builtin_setvar_helper(target_chan, SEND_TO_VM_REDIRECT, NULL);
206         }
207         ao2_ref(feature_datastore, -1);
208 }
209
210 static struct ast_sip_session_supplement refer_supplement = {
211         .method = "REFER",
212         .incoming_request = handle_incoming_request,
213         .outgoing_response = handle_outgoing_response,
214 };
215
216 static int load_module(void)
217 {
218         CHECK_PJSIP_SESSION_MODULE_LOADED();
219
220         if (ast_sip_session_register_supplement(&refer_supplement)) {
221                 ast_log(LOG_ERROR, "Unable to register Send to Voicemail supplement\n");
222                 return AST_MODULE_LOAD_DECLINE;
223         }
224
225         return AST_MODULE_LOAD_SUCCESS;
226 }
227
228 static int unload_module(void)
229 {
230         ast_sip_session_unregister_supplement(&refer_supplement);
231         return 0;
232 }
233
234 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP REFER Send to Voicemail Support",
235         .support_level = AST_MODULE_SUPPORT_CORE,
236         .load = load_module,
237         .unload = unload_module,
238         .load_pri = AST_MODPRI_APP_DEPEND,
239 );