Asterisk - The Open Source Telephony Project  18.5.0
res_pjsip_one_touch_record_info.c
Go to the documentation of this file.
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2013, malleable, llc.
5  *
6  * Sean Bright <[email protected]>
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 /*** MODULEINFO
20  <depend>pjproject</depend>
21  <depend>res_pjsip</depend>
22  <depend>res_pjsip_session</depend>
23  <support_level>core</support_level>
24 ***/
25 
26 #include "asterisk.h"
27 
28 #include <pjsip.h>
29 #include <pjsip_ua.h>
30 
31 #include "asterisk/features.h"
32 #include "asterisk/res_pjsip.h"
34 #include "asterisk/module.h"
36 
37 static void send_response(struct ast_sip_session *session, int code, struct pjsip_rx_data *rdata)
38 {
39  pjsip_tx_data *tdata;
40 
41  if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, code, NULL, &tdata) == PJ_SUCCESS) {
42  struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
43 
44  pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
45  }
46 }
47 
48 static int handle_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
49 {
50  static const pj_str_t rec_str = { "Record", 6 };
51  pjsip_generic_string_hdr *record;
52  int feature_res;
53  char feature_code[AST_FEATURE_MAX_LEN];
54  const char *feature;
55  char *digit;
56 
57  record = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &rec_str, NULL);
58 
59  /* If we don't have Record header, we have nothing to do */
60  if (!record) {
61  return 0;
62  }
63 
64  if (!pj_stricmp2(&record->hvalue, "on")) {
65  feature = session->endpoint->info.recording.onfeature;
66  } else if (!pj_stricmp2(&record->hvalue, "off")) {
67  feature = session->endpoint->info.recording.offfeature;
68  } else {
69  /* Don't send response because another module may handle this */
70  return 0;
71  }
72 
73  if (!session->channel) {
74  send_response(session, 481, rdata);
75  return 1;
76  }
77 
78  /* Is this endpoint configured with One Touch Recording? */
79  if (!session->endpoint->info.recording.enabled || ast_strlen_zero(feature)) {
80  send_response(session, 403, rdata);
81  return 1;
82  }
83 
84  ast_channel_lock(session->channel);
85  feature_res = ast_get_feature(session->channel, feature, feature_code, sizeof(feature_code));
86  ast_channel_unlock(session->channel);
87 
88  if (feature_res || ast_strlen_zero(feature_code)) {
89  send_response(session, 403, rdata);
90  return 1;
91  }
92 
93  for (digit = feature_code; *digit; ++digit) {
94  struct ast_frame f = { AST_FRAME_DTMF, .subclass.integer = *digit, .len = 100 };
95  ast_queue_frame(session->channel, &f);
96  }
97 
98  send_response(session, 200, rdata);
99 
100  return 1;
101 }
102 
104  .method = "INFO",
106  .incoming_request = handle_incoming_request,
107 };
108 
109 static int load_module(void)
110 {
111  ast_sip_session_register_supplement(&info_supplement);
112 
114 }
115 
116 static int unload_module(void)
117 {
118  ast_sip_session_unregister_supplement(&info_supplement);
119  return 0;
120 }
121 
122 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP INFO One Touch Recording Support",
123  .support_level = AST_MODULE_SUPPORT_CORE,
124  .load = load_module,
125  .unload = unload_module,
126  .load_pri = AST_MODPRI_APP_DEPEND,
127  .requires = "res_pjsip,res_pjsip_session",
128 );
char digit
#define ast_channel_lock(chan)
Definition: channel.h:2945
struct ast_sip_endpoint * endpoint
static struct ast_sip_session_supplement info_supplement
Asterisk main include file. File version handling, generic pbx functions.
struct ast_sip_info_recording_configuration recording
Definition: res_pjsip.h:683
void ast_sip_session_unregister_supplement(struct ast_sip_session_supplement *supplement)
Unregister a an supplement to SIP session processing.
Definition: pjsip_session.c:63
#define NULL
Definition: resample.c:96
#define AST_FRAME_DTMF
struct pjsip_inv_session * inv_session
A structure describing a SIP session.
#define ast_strlen_zero(foo)
Definition: strings.h:52
static int handle_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
static struct ast_mansession session
static void send_response(struct ast_sip_session *session, int code, struct pjsip_rx_data *rdata)
const ast_string_field offfeature
Definition: res_pjsip.h:673
struct ast_channel * channel
int ast_get_feature(struct ast_channel *chan, const char *feature, char *buf, size_t len)
Get the DTMF code for a call feature.
int ast_queue_frame(struct ast_channel *chan, struct ast_frame *f)
Queue one or more frames to a channel&#39;s frame queue.
Definition: channel.c:1139
static int unload_module(void)
struct ast_sip_endpoint_info_configuration info
Definition: res_pjsip.h:849
static int load_module(void)
#define ast_channel_unlock(chan)
Definition: channel.h:2946
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS|AST_MODFLAG_LOAD_ORDER, "HTTP Phone Provisioning",.support_level=AST_MODULE_SUPPORT_EXTENDED,.load=load_module,.unload=unload_module,.reload=reload,.load_pri=AST_MODPRI_CHANNEL_DEPEND,.requires="http",)
A supplement to SIP message processing.
const ast_string_field onfeature
Definition: res_pjsip.h:671
#define AST_FEATURE_MAX_LEN
Data structure associated with a single frame of data.
Call Parking and Pickup API Includes code and algorithms from the Zapata library. ...
#define ASTERISK_GPL_KEY
The text the key() function should return.
Definition: module.h:46
Asterisk module definitions.
#define ast_sip_session_register_supplement(supplement)