1 /*!
2 * \file ast113.c
3 * \brief SCCP PBX Asterisk Wrapper Class
4 * \author Marcello Ceshia
5 * \author Diederik de Groot <ddegroot [at] users.sourceforge.net>
6 * \note This program is free software and may be modified and distributed under the terms of the GNU Public License.
7 * See the LICENSE file at the top of the source tree.
8 */
9
10 #include "config.h"
11 #include "common.h"
12 #include "chan_sccp.h"
13 #include "sccp_pbx.h"
14 #include "sccp_channel.h"
15 #include "sccp_device.h"
16 #include "sccp_line.h"
17 #include "sccp_linedevice.h"
18 #include "sccp_cli.h"
19 #include "sccp_utils.h"
20 #include "sccp_indicate.h"
21 #include "sccp_hint.h"
22 #include "sccp_appfunctions.h"
23 #include "sccp_management.h"
24 #include "sccp_netsock.h"
25 #include "sccp_rtp.h"
26 #include "sccp_session.h" // required for sccp_session_getOurIP
27 #include "sccp_labels.h"
28 #include "sccp_enum.h"
29 #include "pbx_impl/ast113/ast113.h"
30 #include "pbx_impl/ast_announce/ast_announce.h"
31
32 SCCP_FILE_VERSION(__FILE__, "");
33
34 __BEGIN_C_EXTERN__
35 #ifdef HAVE_PBX_ACL_H
36 # include <asterisk/acl.h>
37 #endif
38 #include <asterisk/module.h>
39 #include <asterisk/callerid.h>
40 #include <asterisk/causes.h>
41 #include <asterisk/musiconhold.h>
42 #ifdef HAVE_PBX_FEATURES_H
43 # include <asterisk/features_config.h>
44 # include <asterisk/features.h>
45 # include <asterisk/parking.h>
46 # include <asterisk/pickup.h>
47 #endif
48 #ifdef HAVE_PBX_APP_H
49 # include <asterisk/app.h>
50 #endif
51 #include <asterisk/translate.h>
52 #include <asterisk/indications.h>
53 #include <asterisk/cel.h>
54 #include <asterisk/netsock2.h>
55 #include <asterisk/pickup.h>
56 #include <asterisk/stasis.h>
57 #include <asterisk/stasis_channels.h>
58 #include <asterisk/stasis_endpoints.h>
59 #include <asterisk/bridge_after.h>
60 #include <asterisk/bridge_channel.h>
61 #include <asterisk/format_cap.h> // for AST_FORMAT_CAP_NAMES_LEN
62 #include <asterisk/timing.h>
63 #include <asterisk/say.h> // PARKING
64 __END_C_EXTERN__
65 #define pbx_module_info ast_module_info
66
67 struct ast_sched_context *sched = 0;
68 struct io_context *io = 0;
69
70 static PBX_CHANNEL_TYPE *sccp_astwrap_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const PBX_CHANNEL_TYPE * requestor, const char *dest, int *cause);
71 static int sccp_astwrap_call(PBX_CHANNEL_TYPE * ast, const char *dest, int timeout);
72 static int sccp_astwrap_answer(PBX_CHANNEL_TYPE * chan);
73 static PBX_FRAME_TYPE *sccp_astwrap_rtp_read(PBX_CHANNEL_TYPE * ast);
74 static int sccp_astwrap_rtp_write(PBX_CHANNEL_TYPE * ast, PBX_FRAME_TYPE * frame);
75 static int sccp_astwrap_indicate(PBX_CHANNEL_TYPE * ast, int ind, const void *data, size_t datalen);
76 static int sccp_astwrap_fixup(PBX_CHANNEL_TYPE * oldchan, PBX_CHANNEL_TYPE * newchan);
77 static void sccp_astwrap_setDialedNumber(constChannelPtr channel, const char *number);
78 //#ifdef CS_AST_RTP_INSTANCE_BRIDGE
79 //static enum ast_bridge_result sccp_astwrap_rtpBridge(PBX_CHANNEL_TYPE * c0, PBX_CHANNEL_TYPE * c1, int flags, PBX_FRAME_TYPE ** fo, PBX_CHANNEL_TYPE ** rc, int timeoutms);
80 //#endif
81 static int sccp_pbx_sendtext(PBX_CHANNEL_TYPE * ast, const char *text);
82 static int sccp_wrapper_recvdigit_begin(PBX_CHANNEL_TYPE * ast, char digit);
83 static int sccp_wrapper_recvdigit_end(PBX_CHANNEL_TYPE * ast, char digit, unsigned int duration);
84 static int sccp_pbx_sendHTML(PBX_CHANNEL_TYPE * ast, int subclass, const char *data, int datalen);
85 int sccp_astwrap_hangup(PBX_CHANNEL_TYPE * ast_channel);
86 int sccp_astwrap_queue_control(const PBX_CHANNEL_TYPE * pbx_channel, enum ast_control_frame_type control);
87 int sccp_astwrap_queue_control_data(const PBX_CHANNEL_TYPE * pbx_channel, enum ast_control_frame_type control, const void *data, size_t datalen);
88 static int sccp_astwrap_devicestate(const char *data);
89 PBX_CHANNEL_TYPE *sccp_astwrap_findPickupChannelByExtenLocked(PBX_CHANNEL_TYPE * chan, const char *exten, const char *context);
90 PBX_CHANNEL_TYPE *sccp_astwrap_findPickupChannelByGroupLocked(PBX_CHANNEL_TYPE * chan);
91
sccp_astwrap_getSkinnyFormatSingle(struct ast_format_cap * ast_format_capability)92 static inline skinny_codec_t sccp_astwrap_getSkinnyFormatSingle(struct ast_format_cap *ast_format_capability)
93 {
94 uint formatPosition;
95 skinny_codec_t codec = SKINNY_CODEC_NONE;
96 struct ast_format *format;
97
98 for (formatPosition = 0; formatPosition < ast_format_cap_count(ast_format_capability); ++formatPosition) {
99 format = ast_format_cap_get_format(ast_format_capability, formatPosition);
100 uint64_t ast_codec = ast_format_compatibility_format2bitfield(format);
101 ao2_ref(format, -1);
102
103 if ((codec = pbx_codec2skinny_codec(ast_codec)) != SKINNY_CODEC_NONE) {
104 break;
105 }
106 }
107
108 if (codec == SKINNY_CODEC_NONE) {
109 ast_log(LOG_WARNING, "SCCP: (getSkinnyFormatSingle) No matching codec found");
110 }
111 return codec;
112 }
113
sccp_astwrap_getSkinnyFormatMultiple(struct ast_format_cap * ast_format_capability,skinny_codec_t codecs[],int length)114 static uint8_t sccp_astwrap_getSkinnyFormatMultiple(struct ast_format_cap *ast_format_capability, skinny_codec_t codecs[], int length)
115 {
116 // struct ast_format tmp_fmt;
117 uint formatPosition;
118 skinny_codec_t found = SKINNY_CODEC_NONE;
119 uint8_t position = 0;
120 struct ast_format *format;
121
122 memset(codecs, 0, length * sizeof(skinny_codec_t));
123
124 for (formatPosition = 0; formatPosition < ast_format_cap_count(ast_format_capability); ++formatPosition) {
125 format = ast_format_cap_get_format(ast_format_capability, formatPosition);
126 uint64_t ast_codec = ast_format_compatibility_format2bitfield(format);
127 ao2_ref(format, -1);
128
129 if ((found = pbx_codec2skinny_codec(ast_codec)) != SKINNY_CODEC_NONE) {
130 codecs[position++] = found;
131 }
132 }
133
134 if (codecs[0] == SKINNY_CODEC_NONE) {
135 ast_log(LOG_WARNING, "SCCP: (getSkinnyFormatSingle) No matching codecs found");
136 }
137
138 return position;
139 }
140
sccp_astwrap_skinny2ast_format(skinny_codec_t skinnycodec)141 static struct ast_format *sccp_astwrap_skinny2ast_format(skinny_codec_t skinnycodec)
142 {
143 switch (skinnycodec) {
144 case SKINNY_CODEC_WIDEBAND_256K:
145 return ast_format_slin16;
146 case SKINNY_CODEC_G711_ALAW_64K:
147 case SKINNY_CODEC_G711_ALAW_56K:
148 return ast_format_alaw;
149 case SKINNY_CODEC_G711_ULAW_64K:
150 case SKINNY_CODEC_G711_ULAW_56K:
151 return ast_format_ulaw;
152 case SKINNY_CODEC_G722_64K:
153 case SKINNY_CODEC_G722_56K:
154 case SKINNY_CODEC_G722_48K:
155 return ast_format_g722;
156 case SKINNY_CODEC_G723_1:
157 return ast_format_g723;
158 case SKINNY_CODEC_G729:
159 return ast_format_g729;
160 case SKINNY_CODEC_G729_A:
161 return ast_format_g729;
162 case SKINNY_CODEC_G729_B_LOW:
163 return ast_format_ilbc;
164 // case SKINNY_CODEC_G729_B:
165 // case SKINNY_CODEC_G729_ANNEX_B:
166 // case SKINNY_CODEC_G729_AB:
167 // return ast_format_g729b;
168 case SKINNY_CODEC_G726_32K:
169 return ast_format_g726;
170 case SKINNY_CODEC_GSM:
171 return ast_format_gsm;
172 #ifdef AST_FORMAT_OPUS
173 case SKINNY_CODEC_OPUS:
174 return ast_format_opus;
175 #endif
176 #ifdef AST_FORMAT_SIREN7
177 case SKINNY_CODEC_G722_1_24K:
178 return ast_format_siren7;
179 #endif
180 #ifdef AST_FORMAT_SIREN7
181 case SKINNY_CODEC_G722_1_32K:
182 return ast_format_siren14;
183 #endif
184 case SKINNY_CODEC_H261:
185 return ast_format_h261;
186 case SKINNY_CODEC_H263:
187 return ast_format_h263;
188 case SKINNY_CODEC_H263P:
189 return ast_format_h263p;
190 case SKINNY_CODEC_H264:
191 return ast_format_h264;
192 case SKINNY_CODEC_T120:
193 return ast_format_t140;
194 default:
195 return ast_format_none;
196 }
197 }
198
pbx_format_cap_append_skinny(struct ast_format_cap * caps,skinny_codec_t codecs[SKINNY_MAX_CAPABILITIES])199 static void pbx_format_cap_append_skinny(struct ast_format_cap *caps, skinny_codec_t codecs[SKINNY_MAX_CAPABILITIES]) {
200 int i;
201 for (i=0; i<SKINNY_MAX_CAPABILITIES; i++) {
202 if (codecs[i] == SKINNY_CODEC_NONE) {
203 break;
204 }
205 struct ast_format *format = sccp_astwrap_skinny2ast_format(codecs[i]);
206 if (format != ast_format_none && ast_format_cap_iscompatible_format(caps, format) == AST_FORMAT_CMP_NOT_EQUAL) {
207 unsigned int framing = ast_format_get_default_ms(format);
208 ast_format_cap_append(caps, format, framing);
209 //sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "Added codec:%s\n", codec2str(codecs[i]));
210 }
211 }
212 }
213
214 #if defined(__cplusplus) || defined(c_plusplus)
215 /*!
216 * \brief NULL Tech Structure
217 */
218 static const struct ast_channel_tech null_tech = {
219 type: "NULL",
220 description:"Null channel (should not see this)",
221 };
222
223 /*!
224 * \brief SCCP Tech Structure
225 */
226 static struct ast_channel_tech sccp_tech = {
227 /* *INDENT-OFF* */
228 type: SCCP_TECHTYPE_STR,
229 description: "Skinny Client Control Protocol (SCCP)",
230 properties: AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
231 requester: sccp_astwrap_request,
232 devicestate: sccp_astwrap_devicestate,
233 send_digit_begin: sccp_wrapper_recvdigit_begin,
234 send_digit_end: sccp_wrapper_recvdigit_end,
235 call: sccp_astwrap_call,
236 hangup: sccp_astwrap_hangup,
237 answer: sccp_astwrap_answer,
238 read: sccp_astwrap_rtp_read,
239 write: sccp_astwrap_rtp_write,
240 send_text: sccp_pbx_sendtext,
241 send_image: NULL,
242 send_html: sccp_pbx_sendHTML,
243 exception: NULL,
244 // bridge: sccp_astwrap_rtpBridge,
245 bridge: ast_rtp_instance_bridge,
246 early_bridge: ast_rtp_instance_early_bridge,
247 indicate: sccp_astwrap_indicate,
248 fixup: sccp_astwrap_fixup,
249 setoption: NULL,
250 queryoption: NULL,
251 transfer: NULL,
252 write_video: sccp_astwrap_rtp_write,
253 write_text: NULL,
254 bridged_channel: NULL,
255 func_channel_read: sccp_astgenwrap_channel_read,
256 func_channel_write: sccp_astgenwrap_channel_write,
257 get_base_channel: NULL,
258 set_base_channel: NULL,
259 /* *INDENT-ON* */
260 };
261
262 #else
263 /*!
264 * \brief NULL Tech Structure
265 */
266 static const struct ast_channel_tech null_tech = {
267 .type = "NULL",
268 .description = "Null channel (should not see this)",
269 };
270
271 /*!
272 * \brief SCCP Tech Structure
273 */
274 struct ast_channel_tech sccp_tech = {
275 /* *INDENT-OFF* */
276 .type = SCCP_TECHTYPE_STR,
277 .description = "Skinny Client Control Protocol (SCCP)",
278 .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
279 .requester = sccp_astwrap_request,
280 .devicestate = sccp_astwrap_devicestate,
281 .call = sccp_astwrap_call,
282 .hangup = sccp_astwrap_hangup,
283 .answer = sccp_astwrap_answer,
284 .read = sccp_astwrap_rtp_read,
285 .write = sccp_astwrap_rtp_write,
286 .write_video = sccp_astwrap_rtp_write,
287 .indicate = sccp_astwrap_indicate,
288 .fixup = sccp_astwrap_fixup,
289 //.transfer = sccp_pbx_transfer,
290 #ifdef CS_AST_RTP_INSTANCE_BRIDGE
291 .bridge = ast_rtp_instance_bridge,
292 // .bridge = sccp_astwrap_rtpBridge,
293 #endif
294 // asterisk-13 rtp_engine.h implementation of ast_rtp_instance_early_bridge is actually not fully c++ compatible to their own definition, so a cast is required
295 .early_bridge = (enum ast_bridge_result (*)(struct ast_channel *, struct ast_channel *))&ast_rtp_instance_early_bridge,
296 //.bridged_channel =
297
298 .send_text = sccp_pbx_sendtext,
299 .send_html = sccp_pbx_sendHTML,
300 //.send_image =
301
302 .func_channel_read = sccp_astgenwrap_channel_read,
303 .func_channel_write = sccp_astgenwrap_channel_write,
304
305 .send_digit_begin = sccp_wrapper_recvdigit_begin,
306 .send_digit_end = sccp_wrapper_recvdigit_end,
307
308 //.write_text =
309 //.write_video =
310 //.cc_callback = // ccss, new >1.6.0
311 //.exception = // new >1.6.0
312 //.setoption = sccp_astwrap_setOption,
313 //.queryoption = // new >1.6.0
314 //.get_pvt_uniqueid = sccp_pbx_get_callid, // new >1.6.0
315 //.get_base_channel =
316 //.set_base_channel =
317 /* *INDENT-ON* */
318 };
319
320 #endif
321
sccp_astwrap_devicestate(const char * data)322 static int sccp_astwrap_devicestate(const char *data)
323 {
324 enum ast_device_state res = AST_DEVICE_UNKNOWN;
325 char *lineName = (char *) data;
326 char *deviceId = NULL;
327 sccp_channelstate_t state;
328
329 if ((deviceId = strchr(lineName, '@'))) {
330 *deviceId = '\0';
331 deviceId++;
332 }
333
334 state = sccp_hint_getLinestate(lineName, deviceId);
335 sccp_log((DEBUGCAT_HINT)) (VERBOSE_PREFIX_4 "SCCP: (devicestate) sccp_hint returned state:%s for '%s'\n", sccp_channelstate2str(state), lineName);
336 switch (state) {
337 case SCCP_CHANNELSTATE_DOWN:
338 case SCCP_CHANNELSTATE_ONHOOK:
339 res = AST_DEVICE_NOT_INUSE;
340 break;
341 case SCCP_CHANNELSTATE_RINGING:
342 res = AST_DEVICE_RINGING;
343 break;
344 case SCCP_CHANNELSTATE_HOLD:
345 res = AST_DEVICE_ONHOLD;
346 break;
347 case SCCP_CHANNELSTATE_INVALIDNUMBER:
348 case SCCP_CHANNELSTATE_INVALIDCONFERENCE:
349 res = AST_DEVICE_INVALID;
350 break;
351 case SCCP_CHANNELSTATE_BUSY:
352 case SCCP_CHANNELSTATE_DND:
353 res = AST_DEVICE_BUSY;
354 break;
355 case SCCP_CHANNELSTATE_CONGESTION:
356 res = AST_DEVICE_UNAVAILABLE;
357 break;
358 case SCCP_CHANNELSTATE_RINGOUT:
359 case SCCP_CHANNELSTATE_RINGOUT_ALERTING:
360 res = AST_DEVICE_RINGINUSE;
361 break;
362 case SCCP_CHANNELSTATE_DIALING:
363 case SCCP_CHANNELSTATE_DIGITSFOLL:
364 case SCCP_CHANNELSTATE_PROGRESS:
365 case SCCP_CHANNELSTATE_CALLWAITING:
366 case SCCP_CHANNELSTATE_CONNECTEDCONFERENCE:
367 case SCCP_CHANNELSTATE_OFFHOOK:
368 case SCCP_CHANNELSTATE_GETDIGITS:
369 case SCCP_CHANNELSTATE_CONNECTED:
370 case SCCP_CHANNELSTATE_PROCEED:
371 case SCCP_CHANNELSTATE_BLINDTRANSFER:
372 case SCCP_CHANNELSTATE_CALLTRANSFER:
373 case SCCP_CHANNELSTATE_CALLCONFERENCE:
374 case SCCP_CHANNELSTATE_CALLPARK:
375 case SCCP_CHANNELSTATE_CALLREMOTEMULTILINE:
376 case SCCP_CHANNELSTATE_SPEEDDIAL:
377 res = AST_DEVICE_INUSE;
378 break;
379 case SCCP_CHANNELSTATE_SENTINEL:
380 case SCCP_CHANNELSTATE_ZOMBIE:
381 res = AST_DEVICE_UNKNOWN;
382 break;
383 }
384
385 sccp_log((DEBUGCAT_HINT)) (VERBOSE_PREFIX_4 "SCCP: (devicestate) PBX requests state for '%s' - state %s\n", lineName, ast_devstate2str(res));
386 return (int)res;
387 }
388
389 #define RTP_NEW_SOURCE(_c,_log) \
390 if(c->rtp.audio.instance) { \
391 ast_rtp_new_source(c->rtp.audio.instance); \
392 sccp_log((DEBUGCAT_PBX | DEBUGCAT_INDICATE))(VERBOSE_PREFIX_3 "SCCP: " #_log "\n"); \
393 }
394
395 #define RTP_CHANGE_SOURCE(_c,_log) \
396 if(c->rtp.audio.instance) { \
397 ast_rtp_change_source(c->rtp.audio.instance); \
398 sccp_log((DEBUGCAT_PBX | DEBUGCAT_INDICATE))(VERBOSE_PREFIX_3 "SCCP: " #_log "\n"); \
399 }
400
401 /*! \brief Get the name of a format
402 * \note replacement for ast_getformatname
403 * \param format id of format
404 * \return A static string containing the name of the format or "unknown" if unknown.
405 */
pbx_getformatname(const struct ast_format * format)406 const char *pbx_getformatname(const struct ast_format *format)
407 {
408 return ast_format_get_codec_name(format);
409 }
410
411 /*!
412 * \brief Get the names of a set of formats
413 * \note replacement for ast_getformatname_multiple
414 * \param buf a buffer for the output string
415 * \param size size of buf (bytes)
416 * \param format the format (combined IDs of codecs)
417 * Prints a list of readable codec names corresponding to "format".
418 * ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)"
419 * \return The return value is buf.
420 */
pbx_getformatname_multiple(char * buf,size_t size,struct ast_format_cap * format)421 const char *pbx_getformatname_multiple(char *buf, size_t size, struct ast_format_cap *format)
422 {
423 pbx_str_t *codec_buf = pbx_str_alloca(64);
424 ast_format_cap_get_names(format, &codec_buf);
425 snprintf(buf, size, "%s", ast_format_cap_get_names(format, &codec_buf));
426 return buf;
427 }
428
429 /*!
430 * \brief Find Asterisk/PBX channel by linkid
431 *
432 * \param ast pbx channel
433 * \param data linkId as void *
434 *
435 * \return int
436 */
pbx_find_channel_by_linkid(PBX_CHANNEL_TYPE * ast,PBX_CHANNEL_TYPE * remotePeer,const void * data)437 static int pbx_find_channel_by_linkid(PBX_CHANNEL_TYPE * ast, PBX_CHANNEL_TYPE * remotePeer, const void *data)
438 {
439 const char *linkedId = (char *) data;
440 const char *remoteLinkedId = ast_channel_linkedid(remotePeer);
441
442 if (!linkedId) {
443 return 0;
444 }
445
446 return remotePeer != ast && remoteLinkedId && (!strcasecmp(linkedId, remoteLinkedId)) && !pbx_channel_masq(remotePeer);
447 }
448
449 /*! Generic helper function to find the best common codec to use */
__find_joint_capabilities(sccp_channel_t * c,PBX_CHANNEL_TYPE * peer,enum ast_media_type media_type,skinny_codec_t remoteCapabilities[])450 static void __find_joint_capabilities(sccp_channel_t *c, PBX_CHANNEL_TYPE* peer, enum ast_media_type media_type, skinny_codec_t remoteCapabilities[])
451 {
452 char buf[512];
453 pbx_str_t *codec_buf = pbx_str_alloca(64);
454 PBX_CHANNEL_TYPE *owner = c->owner;
455 RAII(struct ast_format_cap *, caps, ao2_t_bump(c->caps, "sccp_channel_caps"), ao2_cleanup);
456 RAII(struct ast_format_cap *, payload_caps, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT), ao2_cleanup);
457 RAII(struct ast_format_cap *, remote_caps, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT), ao2_cleanup);
458 RAII(struct ast_format_cap *, joint, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT), ao2_cleanup);
459 struct ast_format *best_fmt_cap = NULL;
460 struct ast_format *best_fmt_native = NULL;
461 if (caps && payload_caps && remote_caps && joint && !ast_format_cap_empty(caps)) {
462 ast_format_cap_append_from_cap(payload_caps, caps, media_type);
463 ast_channel_lock(peer);
464 struct ast_format_cap * cap_native_peer = ast_channel_nativeformats(peer);
465 if (!ast_format_cap_empty(cap_native_peer) && ast_format_cap_has_type(cap_native_peer, media_type)) {
466 ast_format_cap_append_from_cap(remote_caps, cap_native_peer, media_type);
467 }
468 ast_channel_unlock(peer);
469 if (!ast_format_cap_empty(payload_caps) && !ast_format_cap_empty(remote_caps)) {
470 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "%s: (find_joint_capabilities) native caps:%s (%d)\n", c->designator, ast_format_cap_get_names(caps, &codec_buf), (int)ast_format_cap_count(caps));
471 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "%s: (find_joint_capabilities) %s payload:%s (%d)\n", c->designator, ast_codec_media_type2str(media_type), ast_format_cap_get_names(payload_caps, &codec_buf),
472 (int)ast_format_cap_count(payload_caps));
473 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "%s: (find_joint_capabilities) %s remote:%s (%d)\n", c->designator, ast_codec_media_type2str(media_type), ast_format_cap_get_names(remote_caps, &codec_buf),
474 (int)ast_format_cap_count(remote_caps));
475
476 // fill c->remoteCapabilities
477 sccp_astwrap_getSkinnyFormatMultiple(remote_caps, remoteCapabilities, SKINNY_MAX_CAPABILITIES);
478
479 sccp_codec_multiple2str(buf, sizeof(buf) - 1, remoteCapabilities, SKINNY_MAX_CAPABILITIES);
480 sccp_log(DEBUGCAT_CODEC) (VERBOSE_PREFIX_4 "skinny set remote %s caps: %s\n", ast_codec_media_type2str(media_type), buf);
481
482 ast_format_cap_get_compatible(payload_caps, remote_caps, joint);
483
484 if (!ast_format_cap_empty(joint)) {
485 best_fmt_native = ast_format_cap_get_best_by_type(joint, media_type);
486 sccp_log(DEBUGCAT_CODEC) (VERBOSE_PREFIX_4 "%s: (find_joint_capabilities) NOT transcoding, best_native:%s\n", c->designator, ast_format_get_codec_name(best_fmt_native));
487 if (best_fmt_native) {
488 ast_format_cap_remove_by_type(caps, media_type);
489 ast_format_cap_append(caps, best_fmt_native, media_type);
490 }
491 ast_format_cap_append_from_cap(caps, joint, media_type);
492 } else {
493 ast_translator_best_choice(remote_caps, caps, &best_fmt_cap, &best_fmt_native);
494 if (best_fmt_native) {
495 ao2_ref(best_fmt_native, +1);
496 ast_format_cap_remove_by_type(caps, media_type);
497 ast_format_cap_append(caps, best_fmt_cap, media_type);
498 } else {
499 best_fmt_native = ast_format_cap_get_best_by_type(caps, media_type);
500 }
501 ast_format_cap_append_from_cap(caps, payload_caps, media_type);
502 sccp_log(DEBUGCAT_CODEC) (VERBOSE_PREFIX_3 "%s: pbx_retrieve_remote_capabilities: transcoding, best_cap:%s, best_native:%s\n",
503 c->designator, ast_format_get_codec_name(best_fmt_cap), best_fmt_native ? ast_format_get_codec_name(best_fmt_native) : "");
504 }
505
506 ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(c->owner), AST_MEDIA_TYPE_UNKNOWN);
507 if (best_fmt_native) {
508 if (best_fmt_native != ast_format_none) {
509 if (!ast_format_cap_empty(caps)) {
510 ast_channel_lock(owner);
511 ast_channel_nativeformats_set(c->owner, caps);
512 ast_channel_unlock(owner);
513 sccp_log(DEBUGCAT_CODEC) (VERBOSE_PREFIX_4 "%s: (find_joint_capabilities) using: caps:%s, codec:%s\n", c->designator, ast_format_cap_get_names(caps, &codec_buf), ast_format_get_codec_name(best_fmt_native));
514 }
515 if (AST_MEDIA_TYPE_AUDIO == media_type) {
516 ast_channel_set_writeformat(c->owner, best_fmt_native);
517 ast_channel_set_readformat(c->owner, best_fmt_native);
518 if (c->rtp.audio.instance) {
519 ast_channel_set_rawwriteformat(c->owner, best_fmt_native);
520 ast_channel_set_rawreadformat(c->owner, best_fmt_native);
521 }
522 if (c->rtp.audio.instance) {
523 ast_rtp_instance_set_write_format(c->rtp.audio.instance, best_fmt_native);
524 ast_rtp_instance_set_read_format(c->rtp.audio.instance, best_fmt_native);
525 }
526 } else if (AST_MEDIA_TYPE_VIDEO == media_type && c->rtp.video.instance) {
527 ast_rtp_instance_set_write_format(c->rtp.video.instance, best_fmt_native);
528 ast_rtp_instance_set_read_format(c->rtp.video.instance, best_fmt_native);
529 }
530 }
531 ao2_ref(best_fmt_native, -1);
532 }
533 }
534 return;
535 }
536 }
537
pbx_retrieve_remote_capabilities(sccp_channel_t * c)538 static void pbx_retrieve_remote_capabilities(sccp_channel_t *c)
539 {
540 pbx_assert(c != NULL);
541 PBX_CHANNEL_TYPE *ast = c->owner;
542 PBX_CHANNEL_TYPE *remotePeer;
543
544 struct ast_channel_iterator *iterator = ast_channel_iterator_all_new();
545 ((struct ao2_iterator *) iterator)->flags |= AO2_ITERATOR_DONTLOCK;
546
547 //! \todo handle multiple remotePeers i.e. DIAL(SCCP/400&SIP/300), find smallest common codecs, what order to use ?
548 for (; (remotePeer = ast_channel_iterator_next(iterator)); pbx_channel_unref(remotePeer)) {
549 if (pbx_find_channel_by_linkid(ast, remotePeer, (void *) ast_channel_linkedid(ast))) {
550 __find_joint_capabilities(c, remotePeer, AST_MEDIA_TYPE_AUDIO, c->remoteCapabilities.audio);
551 #if CS_SCCP_VIDEO
552 __find_joint_capabilities(c, remotePeer, AST_MEDIA_TYPE_VIDEO, c->remoteCapabilities.video);
553 #endif
554 break;
555 }
556 }
557 ast_channel_iterator_destroy(iterator);
558 }
559
560
asterisk_indication2str(int ind)561 static const char *asterisk_indication2str(int ind)
562 {
563 switch (ind) {
564 case AST_CONTROL_HANGUP: return "AST_CONTROL_HANGUP: Other end has hungup";
565 case AST_CONTROL_RING: return "AST_CONTROL_RING: Local ring";
566 case AST_CONTROL_RINGING: return "AST_CONTROL_RINGING: Remote end is ringing";
567 case AST_CONTROL_ANSWER: return "AST_CONTROL_ANSWER: Remote end has answered";
568 case AST_CONTROL_BUSY: return "AST_CONTROL_BUSY: Remote end is busy";
569 case AST_CONTROL_TAKEOFFHOOK: return "AST_CONTROL_TAKEOFFHOOK: Make it go off hook";
570 case AST_CONTROL_OFFHOOK: return "AST_CONTROL_OFFHOOK: Line is off hook";
571 case AST_CONTROL_CONGESTION: return "AST_CONTROL_CONGESTION: Congestion (circuits busy)";
572 case AST_CONTROL_FLASH: return "AST_CONTROL_FLASH: Flash hook";
573 case AST_CONTROL_WINK: return "AST_CONTROL_WINK: Wink";
574 case AST_CONTROL_OPTION: return "AST_CONTROL_OPTION: Set a low-level option";
575 case AST_CONTROL_RADIO_KEY: return "AST_CONTROL_RADIO_KEY: Key Radio";
576 case AST_CONTROL_RADIO_UNKEY: return "AST_CONTROL_RADIO_UNKEY: Un-Key Radio";
577 case AST_CONTROL_PROGRESS: return "AST_CONTROL_PROGRESS: Indicate PROGRESS";
578 case AST_CONTROL_PROCEEDING: return "AST_CONTROL_PROCEEDING: Indicate CALL PROCEEDING";
579 case AST_CONTROL_HOLD: return "AST_CONTROL_HOLD: Indicate call is placed on hold";
580 case AST_CONTROL_UNHOLD: return "AST_CONTROL_UNHOLD: Indicate call left hold";
581 case AST_CONTROL_VIDUPDATE: return "AST_CONTROL_VIDUPDATE: Indicate video frame update";
582 case _XXX_AST_CONTROL_T38: return "_XXX_AST_CONTROL_T38: T38 state change request/notification. Deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead.";
583 case AST_CONTROL_SRCUPDATE: return "AST_CONTROL_SRCUPDATE: Indicate source of media has changed";
584 case AST_CONTROL_TRANSFER: return "AST_CONTROL_TRANSFER: Indicate status of a transfer request";
585 case AST_CONTROL_CONNECTED_LINE: return "AST_CONTROL_CONNECTED_LINE: Indicate connected line has changed";
586 case AST_CONTROL_REDIRECTING: return "AST_CONTROL_REDIRECTING: Indicate redirecting id has changed";
587 case AST_CONTROL_T38_PARAMETERS: return "AST_CONTROL_T38_PARAMETERS: T38 state change request/notification with parameters";
588 case AST_CONTROL_CC: return "AST_CONTROL_CC: Indication that Call completion service is possible";
589 case AST_CONTROL_SRCCHANGE: return "AST_CONTROL_SRCCHANGE: Media source has changed and requires a new RTP SSRC";
590 case AST_CONTROL_READ_ACTION: return "AST_CONTROL_READ_ACTION: Tell ast_read to take a specific action";
591 case AST_CONTROL_AOC: return "AST_CONTROL_AOC: Advice of Charge with encoded generic AOC payload";
592 case AST_CONTROL_END_OF_Q: return "AST_CONTROL_END_OF_Q: Indicate that this position was the end of the channel queue for a softhangup.";
593 case AST_CONTROL_INCOMPLETE: return "AST_CONTROL_INCOMPLETE: Indication that the extension dialed is incomplete";
594 case AST_CONTROL_MCID: return "AST_CONTROL_MCID: Indicate that the caller is being malicious.";
595 case AST_CONTROL_UPDATE_RTP_PEER: return "AST_CONTROL_UPDATE_RTP_PEER: Interrupt the bridge and have it update the peer";
596 case AST_CONTROL_PVT_CAUSE_CODE: return "AST_CONTROL_PVT_CAUSE_CODE: Contains an update to the protocol-specific cause-code stored for branching dials";
597 case AST_CONTROL_MASQUERADE_NOTIFY: return "AST_CONTROL_MASQUERADE_NOTIFY: A masquerade is about to begin/end."; // Never sent as a frame but directly with ast_indicate_data()
598 case -1: return "AST_CONTROL_PROD: Kick remote channel";
599 }
600 return "Unknown/Unhandled/IAX Indication";
601 }
602
sccp_astwrap_indicate(PBX_CHANNEL_TYPE * ast,int ind,const void * data,size_t datalen)603 static int sccp_astwrap_indicate(PBX_CHANNEL_TYPE * ast, int ind, const void *data, size_t datalen)
604 {
605 int res = 0; /* indication supported */
606
607 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast));
608 if (!c) {
609 sccp_log((DEBUGCAT_PBX | DEBUGCAT_CHANNEL | DEBUGCAT_INDICATE)) (VERBOSE_PREFIX_3 "SCCP: (pbx_indicate) no sccp channel yet\n");
610 return -1;
611 }
612 sccp_log((DEBUGCAT_PBX | DEBUGCAT_CHANNEL | DEBUGCAT_INDICATE)) (VERBOSE_PREFIX_3 "%s: (pbx_indicate) start indicate '%s'\n", c->designator, asterisk_indication2str(ind));
613
614 AUTO_RELEASE(sccp_device_t, d , sccp_channel_getDevice(c));
615 if (!d || c->state == SCCP_CHANNELSTATE_DOWN) {
616 //sccp_log((DEBUGCAT_PBX | DEBUGCAT_CHANNEL | DEBUGCAT_INDICATE)) (VERBOSE_PREFIX_3 "SCCP: (pbx_indicate) no sccp device yet\n");
617 switch (ind) {
618 case AST_CONTROL_CONNECTED_LINE:
619 sccp_astwrap_connectedline(c, data, datalen);
620 break;
621 case AST_CONTROL_REDIRECTING:
622 sccp_astwrap_redirectedUpdate(c, data, datalen);
623 break;
624 default:
625 res = -1; /* this indication is not supported */
626 break;
627 }
628 return res;
629 }
630
631 sccp_log((DEBUGCAT_PBX | DEBUGCAT_CHANNEL
632 | DEBUGCAT_INDICATE))(VERBOSE_PREFIX_1 "%s: (pbx_indicate) start indicate '%s' (%d) condition on channel %s (rtp_instance:%s, reception.state:%d/%s, transmission.state:%d/%s)\n", DEV_ID_LOG(d),
633 asterisk_indication2str(ind), ind, pbx_channel_name(ast), c->rtp.audio.instance ? "yes" : "no", sccp_rtp_getState(&c->rtp.audio, SCCP_RTP_RECEPTION),
634 codec2str(c->rtp.audio.reception.format), sccp_rtp_getState(&c->rtp.audio, SCCP_RTP_TRANSMISSION), codec2str(c->rtp.audio.transmission.format));
635
636 boolean_t inband_if_receivechannel = FALSE;
637 switch (ind) {
638 case AST_CONTROL_RINGING:
639 if (SKINNY_CALLTYPE_OUTBOUND == c->calltype) {
640 if (pbx_channel_state (c->owner) != AST_STATE_UP) {
641 if (c->remoteCapabilities.audio[0] == SKINNY_CODEC_NONE) {
642 pbx_retrieve_remote_capabilities (c);
643 }
644 // Allow signalling of RINGOUT only on outbound calls.
645 // Otherwise, there are some issues with late arrival of ringing
646 // indications on ISDN calls (chan_lcr, chan_dahdi) (-DD).
647 iPbx.set_callstate (c, AST_STATE_RING);
648 } else {
649 iPbx.set_callstate (c, AST_STATE_RINGING);
650 }
651 sccp_indicate(d, c, SCCP_CHANNELSTATE_RINGOUT);
652 }
653 inband_if_receivechannel = TRUE;
654 break;
655
656 case AST_CONTROL_BUSY:
657 sccp_indicate(d, c, SCCP_CHANNELSTATE_BUSY);
658 iPbx.set_callstate(c, AST_STATE_BUSY);
659 inband_if_receivechannel = TRUE;
660 break;
661
662 case AST_CONTROL_CONGESTION:
663 sccp_indicate(d, c, SCCP_CHANNELSTATE_CONGESTION);
664 inband_if_receivechannel = TRUE;
665 break;
666
667 case AST_CONTROL_PROGRESS:
668 if (c->remoteCapabilities.audio[0] == SKINNY_CODEC_NONE) {
669 pbx_retrieve_remote_capabilities(c);
670 }
671 sccp_indicate(d, c, SCCP_CHANNELSTATE_PROGRESS);
672 /*
673 if (c->state != SCCP_CHANNELSTATE_CONNECTED && c->previousChannelState != SCCP_CHANNELSTATE_CONNECTED) {
674 sccp_indicate(d, c, SCCP_CHANNELSTATE_PROGRESS);
675 } else {
676 // ORIGINATE() to SIP indicates PROGRESS after CONNECTED, causing issues with transfer
677 sccp_indicate(d, c, SCCP_CHANNELSTATE_CONNECTED);
678 }*/
679 inband_if_receivechannel = TRUE;
680 break;
681
682 case AST_CONTROL_PROCEEDING:
683 sccp_indicate(d, c, SCCP_CHANNELSTATE_PROCEED);
684 //inband_if_receivechannel = TRUE;
685 break;
686
687 case AST_CONTROL_SRCCHANGE: /* ask our channel's remote source address to update */
688 if (c->rtp.audio.instance) {
689 ast_rtp_instance_change_source(c->rtp.audio.instance);
690 }
691 break;
692
693 case AST_CONTROL_SRCUPDATE: /* semd control bit to force other side to update, their source address */
694 /* Source media has changed. */
695 sccp_log((DEBUGCAT_PBX | DEBUGCAT_INDICATE)) (VERBOSE_PREFIX_3 "SCCP: Source UPDATE request\n");
696
697 if (c->rtp.audio.instance) {
698 ast_rtp_instance_update_source(c->rtp.audio.instance);
699 }
700 break;
701
702 case AST_CONTROL_HOLD: /* when the bridged channel hold/unhold the call we are notified here */
703 if (c->rtp.audio.instance) {
704 ast_rtp_instance_update_source(c->rtp.audio.instance);
705 }
706 #ifdef CS_SCCP_VIDEO
707 if (c->rtp.video.instance && d && sccp_device_isVideoSupported(d) && sccp_channel_getVideoMode(c) != SCCP_VIDEO_MODE_OFF) {
708 d->protocol->sendMultiMediaCommand(d, c, SKINNY_MISCCOMMANDTYPE_VIDEOFREEZEPICTURE);
709 if(sccp_rtp_getState(&c->rtp.video, SCCP_RTP_TRANSMISSION)) {
710 sccp_channel_stopMultiMediaTransmission(c, TRUE);
711 }
712 ast_rtp_instance_update_source(c->rtp.video.instance);
713 }
714 #endif
715 sccp_astwrap_moh_start(ast, (const char *) data, c->musicclass);
716 sccp_dev_set_message(d, SKINNY_DISP_CALL_ON_HOLD, SCCP_MESSAGE_PRIORITY_TIMEOUT, FALSE, FALSE);
717 break;
718
719 case AST_CONTROL_UNHOLD:
720 if (c->rtp.audio.instance) {
721 ast_rtp_instance_update_source(c->rtp.audio.instance);
722 }
723 #ifdef CS_SCCP_VIDEO
724 if(c->rtp.video.instance && d && sccp_device_isVideoSupported(d) && sccp_channel_getVideoMode(c) != SCCP_VIDEO_MODE_OFF) {
725 ast_rtp_instance_update_source(c->rtp.video.instance);
726 if(!sccp_rtp_getState(&c->rtp.video, SCCP_RTP_RECEPTION)) {
727 sccp_channel_openMultiMediaReceiveChannel(c);
728 } else if((sccp_rtp_getState(&c->rtp.video, SCCP_RTP_RECEPTION) & SCCP_RTP_STATUS_ACTIVE) && !sccp_rtp_getState(&c->rtp.video, SCCP_RTP_TRANSMISSION)) {
729 sccp_channel_startMultiMediaTransmission(c);
730 }
731 ast_rtp_instance_update_source(c->rtp.video.instance);
732 }
733 #endif
734 sccp_astwrap_moh_stop(ast);
735 sccp_dev_clear_message(d, FALSE);
736 break;
737
738 case AST_CONTROL_CONNECTED_LINE:
739 /* remarking out this code, as it is causing issues with callforward + FREEPBX, the calling party will not hear the remote end ringing
740 this patch was added to suppress 'double callwaiting tone', but channel PROD(-1) below is taking care of that already
741 */
742 //if (c->calltype == SKINNY_CALLTYPE_OUTBOUND && c->rtp.audio.reception.state == SCCP_RTP_STATUS_INACTIVE && c->state > SCCP_CHANNELSTATE_DIALING) {
743 // sccp_channel_openReceiveChannel(c);
744 //}
745 sccp_astwrap_connectedline(c, data, datalen);
746 inband_if_receivechannel = TRUE;
747 break;
748
749 case AST_CONTROL_TRANSFER:
750 ast_log(LOG_NOTICE, "%s: Ast Control Transfer: %d", c->designator, *(int *)data);
751 //sccp_astwrap_connectedline(c, data, datalen);
752 res = -1;
753 break;
754
755 case AST_CONTROL_REDIRECTING:
756 sccp_astwrap_redirectedUpdate(c, data, datalen);
757 sccp_indicate(d, c, c->state);
758 break;
759
760 case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
761 #ifdef CS_SCCP_VIDEO
762 if (c->rtp.video.instance && d && sccp_device_isVideoSupported(d) && sccp_channel_getVideoMode(c) != SCCP_VIDEO_MODE_OFF) {
763 d->protocol->sendMultiMediaCommand(d, c, SKINNY_MISCCOMMANDTYPE_VIDEOFASTUPDATEPICTURE);
764 } else
765 #endif
766 {
767 res = -1; /* this indication is not supported */
768 }
769 break;
770
771 #ifdef CS_AST_CONTROL_INCOMPLETE
772 case AST_CONTROL_INCOMPLETE: /*!< Indication that the extension dialed is incomplete */
773 /* \todo implement dial continuation by:
774 * - display message incomplete number
775 * - adding time to channel->scheduler.digittimeout
776 * - rescheduling sccp_pbx_sched_dial
777 */
778 inband_if_receivechannel = TRUE;
779 res = -1;
780 break;
781 #endif
782
783 case AST_CONTROL_PVT_CAUSE_CODE:
784 {
785 /*! \todo This would also be a good moment to update the c->requestHangup to requestQueueHangup */
786 // int hangupcause = ast_channel_hangupcause(ast);
787 // sccp_log((DEBUGCAT_PBX | DEBUGCAT_INDICATE)) (VERBOSE_PREFIX_3 "%s: hangup cause set: %d\n", c->designator, hangupcause);
788 // res = -1;
789 inband_if_receivechannel = TRUE;
790 }
791 break;
792
793 case AST_CONTROL_MASQUERADE_NOTIFY:
794 res = -1; // Return -1 so that asterisk core will correctly set up hangupcauses.
795 break;
796
797 case -1: // Asterisk prod the channel /* STOP_TONE */
798 if ( c->line &&
799 c->state > SCCP_GROUPED_CHANNELSTATE_DIALING &&
800 c->calltype == SKINNY_CALLTYPE_OUTBOUND &&
801 !ast_channel_hangupcause(ast)
802 ) {
803 if(!sccp_rtp_getState(&c->rtp.audio, SCCP_RTP_RECEPTION)) {
804 sccp_channel_openReceiveChannel(c);
805 }
806 }
807 inband_if_receivechannel = TRUE;
808 break;
809
810 default:
811 sccp_log((DEBUGCAT_PBX | DEBUGCAT_INDICATE)) (VERBOSE_PREFIX_3 "%s: (pbx_indicate) Don't know how to indicate condition '%s' (%d)\n", DEV_ID_LOG(d), asterisk_indication2str(ind), ind);
812 res = -1; /* this indication is not supported */
813 break;
814 }
815 if(inband_if_receivechannel && sccp_rtp_getState(&c->rtp.audio, SCCP_RTP_RECEPTION)) {
816 res = -1;
817 }
818 sccp_log((DEBUGCAT_PBX | DEBUGCAT_INDICATE)) (VERBOSE_PREFIX_2 "%s: (pbx_indicate) finish: send indication (res:%d)\n", DEV_ID_LOG(d), res);
819 return res;
820 }
821
822 /*!
823 * \brief Read from an Asterisk Channel
824 * \param ast Asterisk Channel as ast_channel
825 *
826 * \called_from_asterisk
827 *
828 * \note not following the refcount rules... channel is already retained
829 */
sccp_astwrap_rtp_read(PBX_CHANNEL_TYPE * ast)830 static PBX_FRAME_TYPE *sccp_astwrap_rtp_read(PBX_CHANNEL_TYPE * ast)
831 {
832 //AUTO_RELEASE(sccp_channel_t, c , NULL); // not following the refcount rules... channel is already retained
833 sccp_channel_t *c = NULL;
834 PBX_FRAME_TYPE *frame = &ast_null_frame;
835
836 if (!(c = CS_AST_CHANNEL_PVT(ast))) { // not following the refcount rules... channel is already retained
837 pbx_log(LOG_ERROR, "SCCP: (rtp_read) no channel pvt\n");
838 goto EXIT_FUNC;
839 }
840
841 if (!c->rtp.audio.instance) {
842 pbx_log(LOG_NOTICE, "SCCP: (rtp_read) no rtp stream yet. skip\n");
843 goto EXIT_FUNC;
844 }
845
846 switch (ast_channel_fdno(ast)) {
847
848 case 0:
849 frame = ast_rtp_instance_read(c->rtp.audio.instance, 0); /* RTP Audio */
850 break;
851 case 1:
852 frame = ast_rtp_instance_read(c->rtp.audio.instance, 1); /* RTCP Control Channel */
853 break;
854 case 2:
855 #ifdef CS_SCCP_VIDEO
856 frame = ast_rtp_instance_read(c->rtp.video.instance, 0); /* RTP Video */
857 #else
858 pbx_log(LOG_NOTICE, "SCCP: (rtp_read) Cannot handle video rtp stream.\n");
859 #endif
860 break;
861 case 3:
862 #ifdef CS_SCCP_VIDEO
863 frame = ast_rtp_instance_read(c->rtp.video.instance, 1); /* RTCP Control Channel for video */
864 #else
865 pbx_log(LOG_NOTICE, "SCCP: (rtp_read) Cannot handle video rtcp stream.\n");
866 #endif
867 break;
868 default:
869 // pbx_log(LOG_NOTICE, "%s: (rtp_read) Unknown Frame Type (%d). Skipping\n", c->designator, ast_channel_fdno(ast));
870 goto EXIT_FUNC;
871 }
872 //sccp_log((DEBUGCAT_CORE))(VERBOSE_PREFIX_3 "%s: read format: ast->fdno: %d, frametype: %d, %s(%d)\n", DEV_ID_LOG(c->device), ast_channel_fdno(ast), frame->frametype, pbx_getformatname(frame->subclass), frame->subclass);
873 if(frame && frame != &ast_null_frame && frame->frametype == AST_FRAME_VOICE) {
874 #ifdef CS_SCCP_CONFERENCE
875 if (c->conference && (!ast_format_cache_is_slinear(ast_channel_readformat(ast)))) {
876 ast_set_read_format(ast, ast_format_slin96);
877 } else
878 #endif
879 {
880 if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
881 struct ast_format_cap *caps;
882 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "%s: (rtp_read) Format changed to %s\n", c->designator, ast_format_get_name(frame->subclass.format));
883 caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
884 if (caps) {
885 ast_format_cap_append(caps, frame->subclass.format, 0);
886 ast_channel_nativeformats_set(ast, caps);
887 ao2_cleanup(caps);
888 }
889 ast_set_read_format(ast, ast_channel_readformat(ast));
890 ast_set_write_format(ast, ast_channel_writeformat(ast));
891 }
892 }
893 }
894
895 /* Only allow audio through if they sent progress, or if the channel is actually answered */
896 /* removed causing one way audio trouble, needs more research */
897 if(c->calltype != SKINNY_CALLTYPE_INBOUND && pbx_channel_state(ast) != AST_STATE_UP && c->wantsEarlyRTP() && c->progressSent() && (frame == &ast_null_frame || !sccp_channel_finishHolePunch(c))) {
898 // if hole punch is not active and the channel is not active either, we transmit null packets in the meantime
899 // Only allow audio through if they sent progress
900 ast_frfree(frame);
901 frame = &ast_null_frame;
902 }
903 EXIT_FUNC:
904 return frame;
905 }
906
907 /*!
908 * \brief Write to an Asterisk Channel
909 * \param ast Channel as ast_channel
910 * \param frame Frame as ast_frame
911 *
912 * \called_from_asterisk
913 *
914 * \note not following the refcount rules... channel is already retained
915 */
sccp_astwrap_rtp_write(PBX_CHANNEL_TYPE * ast,PBX_FRAME_TYPE * frame)916 static int sccp_astwrap_rtp_write(PBX_CHANNEL_TYPE * ast, PBX_FRAME_TYPE * frame)
917 {
918 //AUTO_RELEASE(sccp_channel_t, c , NULL); // not following the refcount rules... channel is already retained
919 sccp_channel_t *c = NULL;
920
921 int res = 0;
922
923 if (!(c = CS_AST_CHANNEL_PVT(ast))) { // not following the refcount rules... channel is already retained
924 return -1;
925 }
926
927 switch (frame->frametype) {
928 case AST_FRAME_VOICE:
929 // checking for samples to transmit
930 if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
931 pbx_str_t *codec_buf = pbx_str_alloca(64);
932 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "%s: (rtp_write) Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
933 c->designator,
934 ast_format_get_name(frame->subclass.format),
935 ast_format_cap_get_names(ast_channel_nativeformats(ast), &codec_buf),
936 ast_channel_readformat(ast) ? ast_format_get_name(ast_channel_readformat(ast)) : "",
937 ast_channel_writeformat(ast) ? ast_format_get_name(ast_channel_writeformat(ast)) : ""
938 );
939 //return -1;
940 }
941 if(pbx_channel_state(ast) != AST_STATE_UP && c->state > SCCP_GROUPED_CHANNELSTATE_DIALING && c->wantsEarlyRTP() && !c->progressSent()) {
942 sccp_log(DEBUGCAT_RTP)(VERBOSE_PREFIX_3 "%s: (rtp_write) device requested earlyRtp and we received an incoming audio packet calling makeProgress\n", c->designator);
943 c->makeProgress(c);
944 }
945 if (!frame->samples) {
946 if(!strcasecmp(frame->src, "ast_prod")) {
947 sccp_log((DEBUGCAT_PBX | DEBUGCAT_CHANNEL)) (VERBOSE_PREFIX_3 "%s: Asterisk prodded channel %s.\n", c->currentDeviceId, pbx_channel_name(ast));
948 } else {
949 pbx_log(LOG_NOTICE, "%s: Asked to transmit frame type %d ('%s') with no samples.\n", c->currentDeviceId, (int)frame->frametype, frame->src);
950 }
951 break;
952 }
953 if(c->rtp.audio.instance && (sccp_rtp_getState(&c->rtp.audio, SCCP_RTP_RECEPTION) & SCCP_RTP_STATUS_ACTIVE) != 0) {
954 res = ast_rtp_instance_write(c->rtp.audio.instance, frame);
955 }
956 break;
957 case AST_FRAME_IMAGE:
958 case AST_FRAME_VIDEO:
959 #ifdef CS_SCCP_VIDEO
960 if(sccp_channel_getVideoMode(c) != SCCP_VIDEO_MODE_OFF && c->state != SCCP_CHANNELSTATE_HOLD) {
961 if(ast_format_cmp(ast_format_h264, frame->subclass.format) != AST_FORMAT_CMP_EQUAL) {
962 sccp_channel_closeMultiMediaReceiveChannel(c, TRUE);
963 sccp_channel_stopMultiMediaTransmission(c, TRUE);
964 sccp_channel_setVideoMode(c, "off");
965 return -1;
966 }
967 sccp_rtp_status_t receptionstate = sccp_rtp_getState(&c->rtp.video, SCCP_RTP_RECEPTION);
968 if(pbx_channel_state(ast) != AST_STATE_UP && c->state > SCCP_GROUPED_CHANNELSTATE_DIALING) {
969 if(c->wantsEarlyRTP() && !c->progressSent()) {
970 sccp_log(DEBUGCAT_RTP)(VERBOSE_PREFIX_3 "%s: (rtp_write) device requested earlyRtp and we received an incoming video packet calling makeProgress\n", c->designator);
971 c->makeProgress(c);
972 }
973 } else if(!receptionstate) {
974 sccp_log((DEBUGCAT_RTP))(VERBOSE_PREFIX_3 "%s: got first video frame %s\n", c->currentDeviceId, "H264");
975 sccp_channel_openMultiMediaReceiveChannel(c);
976 }
977 if(receptionstate & SCCP_RTP_STATUS_ACTIVE) {
978 res = ast_rtp_instance_write(c->rtp.video.instance, frame);
979 }
980 }
981 #endif
982 break;
983 case AST_FRAME_TEXT:
984 case AST_FRAME_MODEM:
985 default:
986 pbx_log(LOG_WARNING, "%s: Can't send %d type frames with SCCP write on channel %s\n", c->currentDeviceId, frame->frametype, pbx_channel_name(ast));
987 break;
988 }
989
990 return res;
991 }
992
sccp_astwrap_setDialedNumber(const sccp_channel_t * channel,const char * number)993 static void sccp_astwrap_setDialedNumber(const sccp_channel_t *channel, const char *number)
994 {
995 PBX_CHANNEL_TYPE *pbx_channel = channel->owner;
996 if (pbx_channel && number) {
997 struct ast_party_dialed dialed;
998 ast_party_dialed_init(&dialed);
999 dialed.number.str = pbx_strdupa(number);
1000 ast_trim_blanks(dialed.number.str);
1001 ast_party_dialed_set(ast_channel_dialed(pbx_channel), &dialed);
1002 }
1003 }
1004
sccp_astwrap_setCalleridPresentation(PBX_CHANNEL_TYPE * pbx_channel,sccp_callerid_presentation_t presentation)1005 static void sccp_astwrap_setCalleridPresentation(PBX_CHANNEL_TYPE *pbx_channel, sccp_callerid_presentation_t presentation)
1006 {
1007 if (pbx_channel && CALLERID_PRESENTATION_FORBIDDEN == presentation) {
1008 ast_channel_caller(pbx_channel)->id.name.presentation |= AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
1009 ast_channel_caller(pbx_channel)->id.number.presentation |= AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
1010 }
1011 }
1012
sccp_astwrap_setNativeAudioFormats(constChannelPtr channel,skinny_codec_t codecs[])1013 static int sccp_astwrap_setNativeAudioFormats(constChannelPtr channel, skinny_codec_t codecs[])
1014 {
1015 if (!channel || !channel->owner || !ast_channel_nativeformats(channel->owner)) {
1016 pbx_log(LOG_ERROR, "SCCP: (setNativeAudioFormats) no channel provided!\n");
1017 return 0;
1018 }
1019 PBX_CHANNEL_TYPE *ast = channel->owner;
1020
1021 struct ast_format_cap *caps = ao2_t_bump((struct ast_format_cap *)channel->caps, "sccp_channel_caps");
1022 if (!caps) {
1023 return 0;
1024 }
1025 ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN);
1026 ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO);
1027 pbx_format_cap_append_skinny(caps, codecs);
1028
1029 ast_channel_lock(ast);
1030 ast_channel_nativeformats_set(ast, caps);
1031 ast_channel_unlock(ast);
1032 ao2_t_ref(caps, -1, "sccp_channel_caps");
1033
1034 pbx_str_t *codec_buf = pbx_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
1035 sccp_log((DEBUGCAT_RTP | DEBUGCAT_CODEC)) (VERBOSE_PREFIX_3 "%s: (setNativeAudioFormats) %s\n", channel->designator, ast_format_cap_get_names(ast_channel_nativeformats(ast), &codec_buf));
1036
1037 return 1;
1038 }
1039
sccp_astwrap_setNativeVideoFormats(constChannelPtr channel,skinny_codec_t codecs[])1040 static int sccp_astwrap_setNativeVideoFormats(constChannelPtr channel, skinny_codec_t codecs[])
1041 {
1042 if (!channel || !channel->owner || !ast_channel_nativeformats(channel->owner)) {
1043 pbx_log(LOG_ERROR, "SCCP: (setNativeVideoFormats) no channel provided!\n");
1044 return 0;
1045 }
1046 PBX_CHANNEL_TYPE *ast = channel->owner;
1047
1048 struct ast_format_cap *caps = ao2_t_bump((struct ast_format_cap *)channel->caps, "sccp_channel_caps");
1049 if (!caps) {
1050 return 0;
1051 }
1052 ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN);
1053 ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_VIDEO);
1054 pbx_format_cap_append_skinny(caps, codecs);
1055
1056 ast_channel_lock(ast);
1057 ast_channel_nativeformats_set(ast, caps);
1058 ast_channel_unlock(ast);
1059 ao2_t_ref(caps, -1, "sccp_channel_caps");
1060
1061 pbx_str_t *codec_buf = pbx_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
1062 sccp_log((DEBUGCAT_RTP | DEBUGCAT_CODEC)) (VERBOSE_PREFIX_3 "%s: (setNativeVideoFormats) %s\n", channel->designator, ast_format_cap_get_names(ast_channel_nativeformats(ast), &codec_buf));
1063
1064 return 1;
1065 }
1066
sccp_astwrap_setWriteFormat(constChannelPtr channel,skinny_codec_t codec)1067 static boolean_t sccp_astwrap_setWriteFormat(constChannelPtr channel, skinny_codec_t codec)
1068 {
1069 if (!channel) {
1070 return FALSE;
1071 }
1072 sccp_log((DEBUGCAT_RTP | DEBUGCAT_CODEC)) (VERBOSE_PREFIX_3 "%s: (setWriteFormat) %s)\n", channel->designator, codec2str(codec));
1073
1074 struct ast_format *ast_format = sccp_astwrap_skinny2ast_format(codec);
1075 if (ast_format == ast_format_none)
1076 return FALSE;
1077
1078 if (channel->rtp.audio.instance) {
1079 ast_channel_set_rawwriteformat(channel->owner, ast_format);
1080 ast_rtp_instance_set_write_format(channel->rtp.audio.instance, ast_format);
1081 }
1082 ast_channel_set_writeformat(channel->owner, ast_format);
1083 return TRUE;
1084 }
1085
sccp_astwrap_setReadFormat(constChannelPtr channel,skinny_codec_t codec)1086 static boolean_t sccp_astwrap_setReadFormat(constChannelPtr channel, skinny_codec_t codec)
1087 {
1088 if (!channel) {
1089 return FALSE;
1090 }
1091
1092 sccp_log((DEBUGCAT_RTP | DEBUGCAT_CODEC)) (VERBOSE_PREFIX_3 "%s: (setReadFormat) %s)\n", channel->designator, codec2str(codec));
1093
1094 struct ast_format *ast_format = sccp_astwrap_skinny2ast_format(codec);
1095 if (ast_format == ast_format_none)
1096 return FALSE;
1097
1098 if (channel->rtp.audio.instance) {
1099 ast_channel_set_rawreadformat(channel->owner, ast_format);
1100 ast_rtp_instance_set_read_format(channel->rtp.audio.instance, ast_format);
1101 }
1102 ast_channel_set_readformat(channel->owner, ast_format);
1103 return TRUE;
1104 }
1105
sccp_astwrap_removeTimingFD(PBX_CHANNEL_TYPE * ast)1106 static void sccp_astwrap_removeTimingFD(PBX_CHANNEL_TYPE *ast)
1107 {
1108 if (ast) {
1109 struct ast_timer *timer=ast_channel_timer(ast);
1110 if (timer) {
1111 //ast_log(LOG_NOTICE, "%s: (clean_timer_fds) timername: %s, fd:%d\n", ast_channel_name(ast), ast_timer_get_name(timer), ast_timer_fd(timer));
1112 ast_timer_disable_continuous(timer);
1113 ast_timer_close(timer);
1114 ast_channel_set_fd(ast, AST_TIMING_FD, -1);
1115 ast_channel_timingfd_set(ast, -1);
1116 ast_channel_timer_set(ast, NULL);
1117 }
1118 }
1119 }
1120
sccp_astwrap_setOwner(sccp_channel_t * channel,PBX_CHANNEL_TYPE * pbx_channel)1121 static void sccp_astwrap_setOwner(sccp_channel_t * channel, PBX_CHANNEL_TYPE * pbx_channel)
1122 {
1123 PBX_CHANNEL_TYPE *prev_owner = channel->owner;
1124
1125 if (pbx_channel) {
1126 channel->owner = pbx_channel_ref(pbx_channel);
1127 ast_module_ref(ast_module_info->self);
1128 } else {
1129 channel->owner = NULL;
1130 }
1131 if (prev_owner) {
1132 pbx_channel_unref(prev_owner);
1133 ast_module_unref(ast_module_info->self);
1134 }
1135 if (channel->rtp.audio.instance) {
1136 ast_rtp_instance_set_channel_id(channel->rtp.audio.instance, pbx_channel ? ast_channel_uniqueid(pbx_channel) : "");
1137 }
1138 if (channel->rtp.video.instance) {
1139 ast_rtp_instance_set_channel_id(channel->rtp.video.instance, pbx_channel ? ast_channel_uniqueid(pbx_channel) : "");
1140 }
1141 }
1142
__sccp_astwrap_updateConnectedLine(PBX_CHANNEL_TYPE * pbx_channel,const char * number,const char * name,uint8_t reason)1143 static void __sccp_astwrap_updateConnectedLine(PBX_CHANNEL_TYPE *pbx_channel, const char *number, const char *name, uint8_t reason)
1144 {
1145 if (!pbx_channel) {
1146 return;
1147 }
1148
1149 struct ast_party_connected_line connected;
1150 struct ast_set_party_connected_line update_connected = {{0}};
1151
1152 ast_party_connected_line_init(&connected);
1153 if (number) {
1154 update_connected.id.number = 1;
1155 connected.id.number.valid = 1;
1156 connected.id.number.str = pbx_strdupa(number);
1157 connected.id.number.presentation = AST_PRES_ALLOWED_NETWORK_NUMBER;
1158 }
1159 if (name) {
1160 update_connected.id.name = 1;
1161 connected.id.name.valid = 1;
1162 connected.id.name.str = pbx_strdupa(name);
1163 connected.id.name.presentation = AST_PRES_ALLOWED_NETWORK_NUMBER;
1164 }
1165 if (update_connected.id.number || update_connected.id.name) {
1166 ast_set_party_id_all(&update_connected.priv);
1167 // connected.id.tag = NULL;
1168 connected.source = reason;
1169 ast_channel_queue_connected_line_update(pbx_channel, &connected, &update_connected);
1170 sccp_log((DEBUGCAT_PBX)) (VERBOSE_PREFIX_3 "SCCP: do connected line for line '%s', name: %s ,num: %s\n", pbx_channel_name(pbx_channel), name ? name : "(NULL)", number ? number : "(NULL)");
1171 }
1172 }
1173
sccp_astwrap_allocPBXChannel(sccp_channel_t * channel,const void * ids,const PBX_CHANNEL_TYPE * pbxSrcChannel,PBX_CHANNEL_TYPE ** _pbxDstChannel)1174 static boolean_t sccp_astwrap_allocPBXChannel(sccp_channel_t * channel, const void *ids, const PBX_CHANNEL_TYPE * pbxSrcChannel, PBX_CHANNEL_TYPE ** _pbxDstChannel)
1175 {
1176 const struct ast_assigned_ids *assignedids = NULL;
1177 PBX_CHANNEL_TYPE *pbxDstChannel = NULL;
1178 if (!channel || !channel->line) {
1179 return FALSE;
1180 }
1181 AUTO_RELEASE(sccp_line_t, line , sccp_line_retain(channel->line));
1182 if (!line) {
1183 return FALSE;
1184 }
1185
1186 channel->caps = ast_t_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT, "sccp_channel_caps"); /* cleanup in channel destructor */
1187 if (!channel->caps) {
1188 ao2_cleanup(channel->caps);
1189 return FALSE;
1190 }
1191 struct ast_format_cap *caps = ao2_t_bump(channel->caps, "sccp_channel_caps");
1192
1193 if (ids) {
1194 assignedids = (const struct ast_assigned_ids *)ids;
1195 }
1196
1197 sccp_log(DEBUGCAT_CHANNEL)(VERBOSE_PREFIX_3 "SCCP: (allocPBXChannel) Create New Channel with name: SCCP/%s-%08X\n", line->name, channel->callid);
1198 pbxDstChannel = ast_channel_alloc(0, AST_STATE_DOWN, line->cid_num, line->cid_name, line->accountcode, line->name, line->context, assignedids, pbxSrcChannel, line->amaflags, "%s", channel->designator);
1199 if (pbxDstChannel == NULL) {
1200 pbx_log(LOG_ERROR, "SCCP: (allocPBXChannel) ast_channel_alloc failed\n");
1201 ao2_cleanup(caps);
1202 return FALSE;
1203 }
1204
1205 ast_channel_stage_snapshot(pbxDstChannel);
1206 if (channel->callid) {
1207 pbx_channel_callid_set(pbxDstChannel, channel->pbx_callid);
1208 }
1209
1210 ast_channel_tech_set(pbxDstChannel, &sccp_tech);
1211 ast_channel_tech_pvt_set(pbxDstChannel, sccp_channel_retain(channel));
1212
1213 pbx_str_t *codec_buf = pbx_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
1214 //pbx_log(LOG_NOTICE, "allocPBXChannel: tmp->nativeformats=%s\n", ast_format_cap_get_names(ast_channel_nativeformats(pbxDstChannel), &codec_buf));
1215
1216 pbx_format_cap_append_skinny(caps, channel->preferences.audio);
1217 #ifdef CS_SCCP_VIDEO
1218 pbx_format_cap_append_skinny(caps, channel->preferences.video);
1219 #endif
1220 struct ast_format_cap *joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1221 if (!joint) {
1222 ao2_cleanup(pbxDstChannel);
1223 ao2_cleanup(caps);
1224 return FALSE;
1225 }
1226 struct ast_format_cap *what = NULL;
1227 if (pbxSrcChannel && ast_format_cap_count(ast_channel_nativeformats(pbxSrcChannel)) > 0) {
1228 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "allocPBXChannel: new->nativeformats=%s based on audio preferences\n", ast_format_cap_get_names(caps, &codec_buf));
1229
1230 ast_format_cap_get_compatible(caps, ast_channel_nativeformats(pbxSrcChannel /*requestor*/), joint);
1231 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "allocPBXChannel: requestor->nativeformats=%s\n", ast_format_cap_get_names( ast_channel_nativeformats(pbxSrcChannel /*requestor*/), &codec_buf));
1232 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "allocPBXChannel: joint formats=%s\n", ast_format_cap_get_names(joint, &codec_buf));
1233 }
1234 if (ast_format_cap_count(joint) > 0) {
1235 what = joint;
1236 } else if (ast_format_cap_count(caps)) {
1237 what = caps;
1238 } else {
1239 what = (&sccp_tech)->capabilities;
1240 }
1241 ast_format_cap_append_from_cap(caps, what, AST_MEDIA_TYPE_UNKNOWN);
1242
1243 struct ast_format *fmt = ast_format_cap_get_best_by_type(what, AST_MEDIA_TYPE_AUDIO); /* get the best audio format */
1244 if (fmt) {
1245 ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO); /* remove only the other audio formats */
1246 ast_format_cap_append(caps, fmt, 0); /* add our best choice back */
1247 } else {
1248 /* If we don't have an audio format, try to get something */
1249 fmt = ast_format_cap_get_format(caps, 0);
1250 if (!fmt) {
1251 ast_log(LOG_WARNING, "No compatible formats could be found for %s\n", ast_channel_name(pbxDstChannel));
1252 ao2_cleanup(caps);
1253 ast_channel_stage_snapshot_done(pbxDstChannel);
1254 ast_hangup(pbxDstChannel);
1255 return FALSE;
1256 }
1257 }
1258 ast_channel_nativeformats_set(pbxDstChannel, caps);
1259 ao2_cleanup(joint);
1260 ao2_t_ref(caps, -1, "sccp_channel_caps"); /* see:bump */
1261
1262 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_3 "allocPBXChannel: using nativeformats=%s fmt=%s\n", ast_format_cap_get_names(ast_channel_nativeformats(pbxDstChannel), &codec_buf), ast_format_get_name(fmt));
1263
1264 ast_channel_set_writeformat(pbxDstChannel, fmt);
1265 ast_channel_set_rawwriteformat(pbxDstChannel, fmt);
1266 ast_channel_set_readformat(pbxDstChannel, fmt);
1267 ast_channel_set_rawreadformat(pbxDstChannel, fmt);
1268 ao2_ref(fmt, -1);
1269 /* EndCodec */
1270
1271 sccp_astwrap_setOwner(channel, pbxDstChannel);
1272
1273 ast_channel_context_set(pbxDstChannel, line->context);
1274 ast_channel_exten_set(pbxDstChannel, line->name);
1275 ast_channel_priority_set(pbxDstChannel, 1);
1276 ast_channel_adsicpe_set(pbxDstChannel, AST_ADSI_UNAVAILABLE);
1277
1278 if (!sccp_strlen_zero(line->language)) {
1279 ast_channel_language_set(pbxDstChannel, line->language);
1280 }
1281
1282 if (!sccp_strlen_zero(line->accountcode)) {
1283 ast_channel_accountcode_set(pbxDstChannel, line->accountcode);
1284 }
1285
1286 if (!sccp_strlen_zero(line->musicclass)) {
1287 ast_channel_musicclass_set(pbxDstChannel, line->musicclass);
1288 }
1289
1290 if (line->amaflags) {
1291 ast_channel_amaflags_set(pbxDstChannel, line->amaflags);
1292 }
1293 if (line->callgroup) {
1294 ast_channel_callgroup_set(pbxDstChannel, line->callgroup);
1295 }
1296
1297 ast_channel_callgroup_set(pbxDstChannel, line->callgroup); // needed for ast_pickup_call
1298 #if CS_SCCP_PICKUP
1299 if (line->pickupgroup) {
1300 ast_channel_pickupgroup_set(pbxDstChannel, line->pickupgroup);
1301 }
1302 #if CS_AST_HAS_NAMEDGROUP
1303 if (!sccp_strlen_zero(line->namedcallgroup)) {
1304 ast_channel_named_callgroups_set(pbxDstChannel, ast_get_namedgroups(line->namedcallgroup));
1305 }
1306
1307 if (!sccp_strlen_zero(line->namedpickupgroup)) {
1308 ast_channel_named_pickupgroups_set(pbxDstChannel, ast_get_namedgroups(line->namedpickupgroup));
1309 }
1310 # endif
1311 #endif
1312 if (!sccp_strlen_zero(line->parkinglot)) {
1313 ast_channel_parkinglot_set(pbxDstChannel, line->parkinglot);
1314 }
1315
1316 /** the the tonezone using language information */
1317 if (!sccp_strlen_zero(line->language) && ast_get_indication_zone(line->language)) {
1318 ast_channel_zone_set(pbxDstChannel, ast_get_indication_zone(line->language)); /* this will core asterisk on hangup */
1319 }
1320
1321 ast_channel_stage_snapshot_done(pbxDstChannel);
1322 ast_channel_unlock(pbxDstChannel);
1323
1324 (*_pbxDstChannel) = pbxDstChannel;
1325
1326 return TRUE;
1327 }
1328
1329 /*!
1330 * \brief Uses ast_channel_move instead of the old masquerade procedure to force a channel into a bridge and hangup the peer channel
1331 */
sccp_astwrap_masqueradeHelper(PBX_CHANNEL_TYPE * pbxChannel,PBX_CHANNEL_TYPE * pbxTmpChannel)1332 static boolean_t sccp_astwrap_masqueradeHelper(PBX_CHANNEL_TYPE * pbxChannel, PBX_CHANNEL_TYPE * pbxTmpChannel)
1333 {
1334 boolean_t res = FALSE;
1335 pbx_log(LOG_NOTICE, "SCCP: (masqueradeHelper) answer temp: %s\n", ast_channel_name(pbxTmpChannel));
1336
1337 ast_channel_ref(pbxChannel);
1338 if(ast_answer(pbxChannel) == 0) {
1339 pbx_log(LOG_NOTICE, "SCCP: (masqueradeHelper) replace pbxTmpChannel: %s with %s (move)\n", ast_channel_name(pbxTmpChannel), ast_channel_name(pbxChannel));
1340 if(ast_channel_move(pbxTmpChannel, pbxChannel)) {
1341 pbx_log(LOG_ERROR, "SCCP: (masqueradeHelper) move failed. Hanging up tmp channel: %s\n", ast_channel_name(pbxTmpChannel));
1342 ast_hangup(pbxTmpChannel);
1343 } else {
1344 pbx_log(LOG_NOTICE, "SCCP: (masqueradeHelper) move succeeded. Hanging up orphan: %s\n", ast_channel_name(pbxChannel));
1345 ast_hangup(pbxChannel);
1346 pbx_channel_set_hangupcause(pbxTmpChannel, AST_CAUSE_REDIRECTED_TO_NEW_DESTINATION);
1347 res = TRUE;
1348 }
1349 pbx_channel_unref(pbxTmpChannel);
1350 }
1351 pbx_channel_unref(pbxChannel);
1352 return res;
1353 }
1354
sccp_astwrap_allocTempPBXChannel(PBX_CHANNEL_TYPE * pbxSrcChannel,PBX_CHANNEL_TYPE ** _pbxDstChannel)1355 static boolean_t sccp_astwrap_allocTempPBXChannel(PBX_CHANNEL_TYPE * pbxSrcChannel, PBX_CHANNEL_TYPE ** _pbxDstChannel)
1356 {
1357 PBX_CHANNEL_TYPE *pbxDstChannel = NULL;
1358 struct ast_assigned_ids assignedids = {NULL, NULL};
1359
1360 if (!pbxSrcChannel) {
1361 pbx_log(LOG_ERROR, "SCCP: (allocTempPBXChannel) no pbx channel provided\n");
1362 return FALSE;
1363 }
1364 struct ast_format_cap *caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1365 if (!caps) {
1366 return FALSE;
1367 }
1368
1369 /*
1370 assignedids.uniqueid = ast_channel_uniqueid(pbxSrcChannel);
1371 {
1372 char *uniqueid2;
1373 uniqueid2 = ast_alloca(strlen(assignedids.uniqueid) + 3);
1374 strcpy(uniqueid2, assignedids.uniqueid);
1375 strcat(uniqueid2, ";2");
1376 assignedids.uniqueid2 = uniqueid2;
1377 }
1378 */
1379
1380 ast_channel_lock(pbxSrcChannel);
1381 pbxDstChannel = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, ast_channel_accountcode(pbxSrcChannel), pbx_channel_exten(pbxSrcChannel), pbx_channel_context(pbxSrcChannel), &assignedids, pbxSrcChannel, ast_channel_amaflags(pbxSrcChannel), "%s-TMP", ast_channel_name(pbxSrcChannel));
1382 if (pbxDstChannel == NULL) {
1383 pbx_log(LOG_ERROR, "SCCP: (allocTempPBXChannel) ast_channel_alloc failed\n");
1384 ao2_cleanup(caps);
1385 return FALSE;
1386 }
1387
1388 ast_channel_stage_snapshot(pbxDstChannel);
1389 //sccp_astwrap_setOwner(channel, pbxDstChannel);
1390
1391 ast_channel_tech_set(pbxDstChannel, &null_tech); // USE null_tech to prevent fixup issues. Channel is only used to masquerade a channel out of a running bridge.
1392 //ast_channel_tech_pvt_set(pbxDstChannel, sccp_channel_retain(channel));
1393
1394 // if (ast_format_cap_is_empty(pbx_channel_nativeformats(pbxSrcChannel))) {
1395 if (ast_format_cap_count(ast_channel_nativeformats(pbxSrcChannel)) > 0) {
1396 ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(pbxSrcChannel), AST_MEDIA_TYPE_UNKNOWN);
1397 } else {
1398 struct ast_format *ast_format = ast_format_slin16;
1399 unsigned int framing = ast_format_get_default_ms(ast_format);
1400 ast_format_cap_append(caps, ast_format, framing);
1401 }
1402 ast_channel_nativeformats_set(pbxDstChannel, caps);
1403 struct ast_format *tmpfmt = ast_format_cap_get_format(caps, 0);
1404 ast_channel_set_writeformat(pbxDstChannel, tmpfmt);
1405 ast_channel_set_rawwriteformat(pbxDstChannel, tmpfmt);
1406 ast_channel_set_readformat(pbxDstChannel, tmpfmt);
1407 ast_channel_set_rawreadformat(pbxDstChannel, tmpfmt);
1408
1409 //pbx_str_t *codec_buf = pbx_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
1410 //pbx_log(LOG_NOTICE, "allocPBXChannel: tmp->nativeformats=%s fmt=%s\n", ast_format_cap_get_names(ast_channel_nativeformats(pbxDstChannel), &codec_buf), ast_format_get_name(tmpfmt));
1411
1412 ao2_ref(tmpfmt, -1);
1413 ao2_ref(caps, -1);
1414 //ao2_cleanup(caps);
1415 /* EndCodec */
1416
1417 ast_channel_context_set(pbxDstChannel, ast_channel_context(pbxSrcChannel));
1418 ast_channel_exten_set(pbxDstChannel, ast_channel_exten(pbxSrcChannel));
1419 ast_channel_priority_set(pbxDstChannel, ast_channel_priority(pbxSrcChannel));
1420 ast_channel_adsicpe_set(pbxDstChannel, AST_ADSI_UNAVAILABLE);
1421 ast_channel_stage_snapshot_done(pbxDstChannel);
1422 ast_channel_unlock(pbxSrcChannel);
1423 ast_channel_unlock(pbxDstChannel);
1424
1425 (*_pbxDstChannel) = pbxDstChannel;
1426 //pbx_channel_ref(pbxDstChannel);
1427
1428 return TRUE;
1429 }
1430
sccp_astwrap_requestAnnouncementChannel(pbx_format_type format_type,const PBX_CHANNEL_TYPE * requestor,void * data)1431 static PBX_CHANNEL_TYPE *sccp_astwrap_requestAnnouncementChannel(pbx_format_type format_type, const PBX_CHANNEL_TYPE * requestor, void *data)
1432 {
1433 PBX_CHANNEL_TYPE *chan;
1434 int cause;
1435 struct ast_format_cap *cap;
1436 struct ast_format *ast_format;
1437 unsigned int framing;
1438
1439 cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1440 if (!cap) {
1441 return NULL;
1442 }
1443 // TODO(dkgroot): convert format_type to ast_format
1444 ast_format = ast_format_alaw;
1445 framing = ast_format_get_default_ms(ast_format);
1446
1447 ast_format_cap_append(cap, ast_format, framing);
1448 chan = ast_request("SCCPCBAnn", cap, NULL, NULL, (char *) data, &cause);
1449 ao2_ref(cap, -1);
1450
1451 if (!chan) {
1452 pbx_log(LOG_ERROR, "SCCP: Requested Unreal channel could not be created, cause: %d\n", cause);
1453 return NULL;
1454 }
1455 /* To make sure playback_chan has the same language of that profile */
1456 if (requestor) {
1457 ast_channel_lock(chan);
1458 ast_channel_language_set(chan, ast_channel_language(requestor));
1459 ast_channel_unlock(chan);
1460 }
1461
1462 ast_debug(1, "Created Unreal channel '%s' related to '%s'\n", ast_channel_name(chan), (char *) data);
1463 return chan;
1464 }
1465
1466
sccp_astwrap_hangup(PBX_CHANNEL_TYPE * ast_channel)1467 int sccp_astwrap_hangup(PBX_CHANNEL_TYPE * ast_channel)
1468 {
1469 // ast_channel_stage_snapshot(ast_channel);
1470 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast_channel));
1471 int res = -1;
1472
1473 if (c) {
1474 sccp_mutex_lock(&c->lock);
1475 if (pbx_channel_hangupcause(ast_channel) == AST_CAUSE_ANSWERED_ELSEWHERE) {
1476 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: This call was answered elsewhere\n");
1477 c->answered_elsewhere = TRUE;
1478 }
1479 /* postponing pbx_channel_unref to sccp_channel destructor */
1480 AUTO_RELEASE(sccp_channel_t, channel , sccp_pbx_hangup(c)); /* explicit release from unretained channel returned by sccp_pbx_hangup */
1481 (void) channel; // suppress unused variable warning
1482 sccp_mutex_unlock(&c->lock);
1483 ast_channel_tech_pvt_set(ast_channel, NULL);
1484 } else { // after this moment c might have gone already
1485 ast_channel_tech_pvt_set(ast_channel, NULL);
1486 pbx_channel_unref(ast_channel); // strange unknown channel, why did we get called to hang it up ?
1487 }
1488 return res;
1489 }
1490
parking_event_cb(void * data,struct stasis_subscription * sub,struct stasis_message * message)1491 static void parking_event_cb (void * data, struct stasis_subscription * sub, struct stasis_message * message)
1492 {
1493 struct ast_parked_call_payload * parked_payload = stasis_message_data (message);
1494 AUTO_RELEASE (sccp_channel_t, parker, sccp_channel_retain ((sccp_channel_t *)data));
1495 if (!parker) {
1496 pbx_log (LOG_ERROR, "No Parker provided\n");
1497 return;
1498 }
1499
1500 RAII (struct ast_channel *, parker_chan, ast_channel_ref (parker->owner), ao2_cleanup);
1501 if (!parker_chan) {
1502 pbx_log (LOG_ERROR, "%s: No Park Owner set\n", parker->designator);
1503 return;
1504 }
1505
1506 if (stasis_subscription_final_message (sub, message)) {
1507 // pbx_log(LOG_NOTICE, "%s: (parking_event_cb) Final Message", parker->designator);
1508 return;
1509 }
1510
1511 switch (parked_payload->event_type) {
1512 case PARKED_CALL: {
1513 sccp_log (DEBUGCAT_CHANNEL) (VERBOSE_PREFIX_3 "%s Parked call to parkingspace:%d@%s\n", ast_channel_name (parker_chan), parked_payload->parkingspace, parked_payload->parkinglot);
1514 char parkingspace[16];
1515 snprintf (parkingspace, sizeof (parkingspace), "%d", parked_payload->parkingspace);
1516 AUTO_RELEASE (sccp_device_t, d, sccp_channel_getDevice (parker));
1517 if (d) {
1518 char extstr[20] = "";
1519 snprintf (extstr, sizeof (extstr), "%c%c %s", 128, SKINNY_LBL_CALL_PARK_AT, parkingspace);
1520 sccp_dev_displayprinotify (d, extstr, SCCP_MESSAGE_PRIORITY_TIMEOUT, 20);
1521 }
1522 RAII (struct ast_bridge *, bridge, ast_channel_get_bridge (parker_chan), ao2_cleanup);
1523 if (bridge) {
1524 ast_bridge_suspend (bridge, parker_chan);
1525 ast_channel_lock (parker_chan);
1526 ast_channel_ref (parker_chan);
1527 ast_channel_unlock (parker_chan);
1528 ast_say_digit_str (parker_chan, parkingspace, "", ast_channel_language (parker_chan));
1529 ast_bridge_unsuspend (bridge, parker_chan);
1530 ast_channel_unref (parker_chan);
1531 } else {
1532 parker->setTone (parker, SKINNY_TONE_CONFIRMATIONTONE, SKINNY_TONEDIRECTION_USER);
1533 }
1534 sccp_astgenwrap_requestHangup (parker);
1535 } break;
1536 case PARKED_CALL_FAILED: {
1537 pbx_log (LOG_ERROR, "%s Parked failed\n", ast_channel_name (parker_chan));
1538 AUTO_RELEASE (sccp_device_t, d, sccp_channel_getDevice (parker));
1539 if (d) {
1540 sccp_dev_displayprinotify (d, SKINNY_DISP_TEMP_FAIL, SCCP_MESSAGE_PRIORITY_TIMEOUT, GLOB (digittimeout));
1541 }
1542 parker->setTone (parker, SKINNY_TONE_REORDERTONE, SKINNY_TONEDIRECTION_USER);
1543 sccp_astgenwrap_requestHangup (parker);
1544 } break;
1545 default:
1546 break;
1547 }
1548 }
1549
parking_subscriptionCleanup(void * data)1550 static void * parking_subscriptionCleanup (void * data)
1551 {
1552 AUTO_RELEASE (sccp_channel_t, hostChannel, (sccp_channel_t *)data);
1553 if (hostChannel && hostChannel->parking_sub) {
1554 hostChannel->parking_sub = stasis_unsubscribe_and_join (hostChannel->parking_sub);
1555 }
1556 return NULL;
1557 }
1558
1559 /*!
1560 * \brief Parking Thread Arguments Structure
1561 */
1562
1563 /*!
1564 * \brief Park the bridge channel of hostChannel
1565 * This function prepares the host and the bridged channel to be ready for parking.
1566 * It clones the pbx channel of both sides forward them to the park_thread
1567 *
1568 * \param hostChannel initial channel that request the parking
1569 * \todo we have a codec issue after unpark a call
1570 * \todo copy connected line info
1571 *
1572 */
sccp_astwrap_park(constChannelPtr hostChannel)1573 static sccp_parkresult_t sccp_astwrap_park(constChannelPtr hostChannel)
1574 {
1575 sccp_parkresult_t res = PARK_RESULT_FAIL;
1576 RAII (struct ast_channel *, parker_chan, ast_channel_ref (hostChannel->owner), ast_channel_cleanup);
1577 RAII(struct ast_bridge_channel *, bridge_channel, NULL, ao2_cleanup);
1578
1579 if (!ast_parking_provider_registered()) {
1580 return res;
1581 }
1582
1583 do {
1584 AUTO_RELEASE (sccp_device_t, device, sccp_channel_getDevice (hostChannel));
1585 sccp_log (DEBUGCAT_CHANNEL) (VERBOSE_PREFIX_3 "%s: Handling Park\n", hostChannel->designator);
1586 if (device && (ast_channel_state (parker_chan) == AST_STATE_UP)) {
1587 ast_channel_lock (parker_chan); /* we have to lock our channel, otherwise asterisk crashes internally */
1588 bridge_channel = ast_channel_get_bridge_channel (parker_chan);
1589 ast_channel_unlock (parker_chan);
1590 if (!bridge_channel) {
1591 pbx_log (LOG_ERROR, "Park action failed\n");
1592 break;
1593 }
1594 if (!hostChannel->parking_sub) {
1595 // pbx_log(LOG_NOTICE, "%s: Subscribing to park topic\n", hostChannel->designator);
1596 channelPtr c = (channelPtr)hostChannel; // casting away const
1597 c->parking_sub = stasis_subscribe (ast_parking_topic(), parking_event_cb, (void *)c);
1598 #if CS_AST_HAS_STASIS_SUBSCRIPTION_SET_FILTER
1599 stasis_subscription_accept_message_type (c->parking_sub, ast_parked_call_type());
1600 stasis_subscription_set_filter (c->parking_sub, STASIS_SUBSCRIPTION_FILTER_SELECTIVE);
1601 #endif
1602 // pbx_log(LOG_NOTICE, "%s: Added cleaning job\n", hostChannel->designator);
1603 sccp_channel_addCleanupJob (c, &parking_subscriptionCleanup, (void *)sccp_channel_retain (c));
1604 }
1605 if (bridge_channel) {
1606 RAII (struct ast_channel *, other_chan, NULL, ast_channel_cleanup);
1607 ast_bridge_channel_lock_bridge (bridge_channel);
1608 uint8_t peer_count = bridge_channel->bridge->num_channels;
1609 if (peer_count == 2) {
1610 struct ast_bridge_channel * other = ast_bridge_channel_peer (bridge_channel);
1611 other_chan = other->chan;
1612 ast_channel_ref (other_chan);
1613 }
1614 ast_bridge_unlock (bridge_channel->bridge);
1615 if (!other_chan) {
1616 pbx_log (LOG_WARNING, "%s: Remote channel is missing, giving up (connected to application?)\n", hostChannel->designator);
1617 break;
1618 }
1619 pbx_builtin_setvar_helper (other_chan, "_PARKED_BY_CHANNEL", ast_channel_name (parker_chan));
1620 pbx_builtin_setvar_helper (other_chan, "_PARKED_BY_UNIQUEID", ast_channel_uniqueid (parker_chan));
1621
1622 sccp_log (DEBUGCAT_CHANNEL) (VERBOSE_PREFIX_3 "%s: Parking %s\n", hostChannel->designator, ast_channel_name (other_chan));
1623 const char * parkinglot = ast_channel_parkinglot (bridge_channel->chan);
1624 char app_data[256];
1625 snprintf (app_data, 256, "%s,%s", parkinglot, "s");
1626 if (ast_bridge_channel_write_park (bridge_channel, ast_channel_uniqueid (other_chan), ast_channel_uniqueid (bridge_channel->chan), app_data) != 0) {
1627 pbx_log (LOG_ERROR, "%s: Parking bridge_channel failed\n", hostChannel->designator);
1628 break;
1629 }
1630
1631 AUTO_RELEASE (sccp_channel_t, remote, get_sccp_channel_from_pbx_channel (other_chan));
1632 if (remote) {
1633 sccp_indicate (NULL, remote, SCCP_CHANNELSTATE_CALLPARK);
1634 }
1635
1636 pbx_channel_set_hangupcause (parker_chan, AST_CAUSE_REDIRECTED_TO_NEW_DESTINATION);
1637 sccp_channel_schedule_hangup (hostChannel, SCCP_HANGUP_TIMEOUT);
1638
1639 res = PARK_RESULT_SUCCESS;
1640 }
1641 }
1642 } while (0);
1643 return res;
1644 }
1645
sccp_astwrap_getFeatureExtension(constChannelPtr channel,const char * featureName,char extension[SCCP_MAX_EXTENSION])1646 static boolean_t sccp_astwrap_getFeatureExtension(constChannelPtr channel, const char *featureName, char extension[SCCP_MAX_EXTENSION])
1647 {
1648 int feat_res = ast_get_feature(channel->owner, featureName, extension, SCCP_MAX_EXTENSION);
1649 return feat_res ? FALSE : TRUE;
1650 }
1651
sccp_astwrap_getPickupExtension(constChannelPtr channel,char extension[SCCP_MAX_EXTENSION])1652 static boolean_t sccp_astwrap_getPickupExtension(constChannelPtr channel, char extension[SCCP_MAX_EXTENSION])
1653 {
1654 boolean_t res = FALSE;
1655 struct ast_features_pickup_config *pickup_cfg = NULL;
1656
1657 if (channel->owner) {
1658 pbx_channel_lock(channel->owner);
1659 pickup_cfg = ast_get_chan_features_pickup_config(channel->owner);
1660 if (!pickup_cfg) {
1661 ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
1662 } else {
1663 if (!sccp_strlen_zero(pickup_cfg->pickupexten)) {
1664 sccp_copy_string(extension, pickup_cfg->pickupexten, SCCP_MAX_EXTENSION);
1665 res = TRUE;
1666 }
1667 pbx_channel_unref(pickup_cfg);
1668 }
1669 pbx_channel_unlock(channel->owner);
1670 }
1671 return res;
1672 }
1673
sccp_astwrap_get_payloadType(const struct sccp_rtp * rtp,skinny_codec_t codec)1674 static uint8_t sccp_astwrap_get_payloadType(const struct sccp_rtp *rtp, skinny_codec_t codec)
1675 {
1676 struct ast_format *astCodec = sccp_astwrap_skinny2ast_format(codec);
1677 if (astCodec != ast_format_none) {
1678 return ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(rtp->instance), 1, astCodec, 0);
1679 }
1680 return 0;
1681 }
1682
sccp_astwrap_get_sampleRate(skinny_codec_t codec)1683 static int sccp_astwrap_get_sampleRate(skinny_codec_t codec)
1684 {
1685 struct ast_format *astCodec = sccp_astwrap_skinny2ast_format(codec);
1686 if (astCodec != ast_format_none) {
1687 return ast_rtp_lookup_sample_rate2(1, astCodec, 0);
1688 }
1689 return 0;
1690 }
1691
sccp_astwrap_extensionStatus(constChannelPtr channel)1692 static sccp_extension_status_t sccp_astwrap_extensionStatus(constChannelPtr channel)
1693 {
1694 PBX_CHANNEL_TYPE *pbx_channel = channel->owner;
1695
1696 if (!pbx_channel || !pbx_channel_context(pbx_channel)) {
1697 pbx_log(LOG_ERROR, "%s: (extension_status) Either no pbx_channel or no valid context provided to lookup number\n", channel->designator);
1698 return SCCP_EXTENSION_NOTEXISTS;
1699 }
1700 int ignore_pat = ast_ignore_pattern(pbx_channel_context(pbx_channel), channel->dialedNumber);
1701 int ext_exist = ast_exists_extension(pbx_channel, pbx_channel_context(pbx_channel), channel->dialedNumber, 1, channel->line->cid_num);
1702 int ext_canmatch = ast_canmatch_extension(pbx_channel, pbx_channel_context(pbx_channel), channel->dialedNumber, 1, channel->line->cid_num);
1703 int ext_matchmore = ast_matchmore_extension(pbx_channel, pbx_channel_context(pbx_channel), channel->dialedNumber, 1, channel->line->cid_num);
1704
1705 // RAII(struct ast_features_pickup_config *, pickup_cfg, ast_get_chan_features_pickup_config(pbx_channel), ao2_cleanup);
1706 // const char *pickupexten = (pickup_cfg) ? pickup_cfg->pickupexten : "-";
1707
1708 /* if we dialed the pickup extention, mark this as exact match */
1709 const char *pickupexten = "";
1710 struct ast_features_pickup_config *pickup_cfg = NULL;
1711
1712 if (channel->owner) {
1713 pbx_channel_lock(channel->owner);
1714 pickup_cfg = ast_get_chan_features_pickup_config(channel->owner);
1715 if (!pickup_cfg) {
1716 ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
1717 pickupexten = "";
1718 } else {
1719 pickupexten = pbx_strdupa(pickup_cfg->pickupexten);
1720 pbx_channel_unref(pickup_cfg);
1721 }
1722 pbx_channel_unlock(channel->owner);
1723 }
1724 if (!sccp_strlen_zero(pickupexten) && sccp_strcaseequals(pickupexten, channel->dialedNumber)) {
1725 sccp_log(DEBUGCAT_CORE) (VERBOSE_PREFIX_3 "SCCP: pbx extension matcher found pickup extension %s matches dialed number %s\n", channel->dialedNumber, pickupexten);
1726 ext_exist = 1;
1727 ext_canmatch = 1;
1728 ext_matchmore = 0;
1729 }
1730
1731 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "+- pbx extension matcher (%-15s): ---+\n" VERBOSE_PREFIX_3 "|ignore |exists |can match |match more|\n" VERBOSE_PREFIX_3 "|%3s |%3s |%3s |%3s |\n" VERBOSE_PREFIX_3 "+----------------------------------------------+\n", channel->dialedNumber, ignore_pat ? "yes" : "no", ext_exist ? "yes" : "no", ext_canmatch ? "yes" : "no", ext_matchmore ? "yes" : "no");
1732
1733 if (ignore_pat) {
1734 return SCCP_EXTENSION_NOTEXISTS;
1735 }
1736 if (ext_exist) {
1737 if (ext_canmatch && !ext_matchmore) {
1738 return SCCP_EXTENSION_EXACTMATCH;
1739 }
1740 return SCCP_EXTENSION_MATCHMORE;
1741 }
1742
1743 return SCCP_EXTENSION_NOTEXISTS;
1744 }
1745
sccp_astwrap_request(const char * type,struct ast_format_cap * cap,const struct ast_assigned_ids * assignedids,const struct ast_channel * requestor,const char * dest,int * cause)1746 static PBX_CHANNEL_TYPE *sccp_astwrap_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *dest, int *cause)
1747 {
1748 sccp_channel_request_status_t requestStatus;
1749 PBX_CHANNEL_TYPE *result_ast_channel = NULL;
1750 pbx_str_t *codec_buf = pbx_str_alloca(64);
1751 #if !defined(CS_AST_CHANNEL_CALLID_TYPEDEF)
1752 struct ast_callid *callid = NULL;
1753 #else
1754 ast_callid callid = 0;
1755 #endif
1756 skinny_codec_t audioCapabilities[SKINNY_MAX_CAPABILITIES] = {SKINNY_CODEC_NONE};
1757 skinny_codec_t videoCapabilities[SKINNY_MAX_CAPABILITIES] = {SKINNY_CODEC_NONE};
1758
1759 //! \todo parse request
1760 char *lineName;
1761 skinny_codec_t audio_codec = SKINNY_CODEC_WIDEBAND_256K;
1762 #if CS_SCCP_VIDEO
1763 skinny_codec_t video_codec = SKINNY_CODEC_NONE;
1764 #endif
1765 sccp_autoanswer_t autoanswer_type = SCCP_AUTOANSWER_NONE;
1766 uint8_t autoanswer_cause = AST_CAUSE_NOTDEFINED;
1767 skinny_ringtype_t ringermode = GLOB(ringtype);
1768
1769 if (!(ast_format_cap_has_type(cap, AST_MEDIA_TYPE_AUDIO))) {
1770 ast_log(LOG_NOTICE, "Asked to get a channel with an unsupported format '%s'\n", ast_format_cap_get_names(cap, &codec_buf));
1771 }
1772
1773 *cause = AST_CAUSE_NOTDEFINED;
1774 if (!type) {
1775 pbx_log(LOG_NOTICE, "Attempt to call with unspecified type of channel\n");
1776 *cause = AST_CAUSE_REQUESTED_CHAN_UNAVAIL;
1777 return NULL;
1778 }
1779
1780 if (!dest) {
1781 pbx_log(LOG_NOTICE, "Attempt to call SCCP/ failed\n");
1782 *cause = AST_CAUSE_REQUESTED_CHAN_UNAVAIL;
1783 return NULL;
1784 }
1785 /* we leave the data unchanged */
1786 lineName = pbx_strdupa((const char *) dest);
1787
1788 /* parsing options string */
1789 char *options = NULL;
1790 if ((options = strchr(lineName, '/'))) {
1791 *options = '\0';
1792 options++;
1793 }
1794 // sccp_log(DEBUGCAT_CHANNEL) (VERBOSE_PREFIX_3 "SCCP: Asterisk asked us to create a channel with type=%s, format=" UI64FMT ", lineName=%s, options=%s\n", type, (uint64_t) ast_format_compatibility_codec2bitfield(cap), lineName, (options) ? options : "");
1795 sccp_log(DEBUGCAT_CHANNEL) (VERBOSE_PREFIX_3 "SCCP: Asterisk asked us to create a channel with type=%s, format=%s, lineName=%s, options=%s\n", type, ast_format_cap_get_names(cap, &codec_buf), lineName, (options) ? options : "");
1796 if (requestor) { /* get ringer mode from ALERT_INFO */
1797 sccp_parse_alertinfo((PBX_CHANNEL_TYPE *)requestor, &ringermode);
1798 }
1799 sccp_parse_dial_options(options, &autoanswer_type, &autoanswer_cause, &ringermode);
1800 if (autoanswer_cause) {
1801 *cause = autoanswer_cause;
1802 }
1803 sccp_parse_auto_answer((PBX_CHANNEL_TYPE *)requestor, &autoanswer_type);
1804
1805 /** getting remote capabilities */
1806 if (requestor) {
1807 struct ast_format_cap *acaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1808 if (acaps) {
1809 ast_format_cap_append_from_cap(acaps, cap, AST_MEDIA_TYPE_AUDIO); // Add Resquested
1810 if (ast_format_cap_count(acaps)) {
1811 audio_codec = sccp_astwrap_getSkinnyFormatSingle(acaps); // Is it compatible
1812 ast_format_cap_append_from_cap(acaps, ast_channel_nativeformats(requestor), AST_MEDIA_TYPE_AUDIO); // Add rest
1813 sccp_astwrap_getSkinnyFormatMultiple(acaps, audioCapabilities, ARRAY_LEN(audioCapabilities));
1814 if (audio_codec == SKINNY_CODEC_NONE && (audio_codec = audioCapabilities[0]) == SKINNY_CODEC_NONE) {
1815 pbx_log(LOG_NOTICE, "SCCP: remote native audio formats are not compatible with any skinny format. Transcoding required\n");
1816 audioCapabilities[0] = SKINNY_CODEC_WIDEBAND_256K;
1817 audio_codec =SKINNY_CODEC_WIDEBAND_256K;
1818 }
1819 }
1820 ao2_cleanup(acaps);
1821 }
1822 #if CS_SCCP_VIDEO
1823 struct ast_format_cap *vcaps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
1824 if (vcaps) {
1825 ast_format_cap_append_from_cap(vcaps, cap, AST_MEDIA_TYPE_VIDEO); // Add Resquested
1826 if (ast_format_cap_count(vcaps)) {
1827 video_codec = sccp_astwrap_getSkinnyFormatSingle(vcaps); // Is it compatible
1828 ast_format_cap_append_from_cap(vcaps, ast_channel_nativeformats(requestor), AST_MEDIA_TYPE_VIDEO); // Add rest
1829 sccp_astwrap_getSkinnyFormatMultiple(vcaps, videoCapabilities, ARRAY_LEN(videoCapabilities));
1830 if (video_codec == SKINNY_CODEC_NONE && (video_codec = videoCapabilities[0]) == SKINNY_CODEC_NONE) {
1831 pbx_log(LOG_NOTICE, "SCCP: remote native video formats are not compatible with any skinny format. Deny video\n");
1832 videoCapabilities[0] = SKINNY_CODEC_NONE;
1833 video_codec = SKINNY_CODEC_NONE;
1834 }
1835 }
1836 ao2_cleanup(vcaps);
1837 }
1838 #endif
1839 }
1840
1841 char cap_buf[512];
1842 sccp_codec_multiple2str(cap_buf, sizeof(cap_buf) - 1, audioCapabilities, ARRAY_LEN(audioCapabilities));
1843 sccp_log(DEBUGCAT_CODEC) (VERBOSE_PREFIX_4 "remote audio caps: %s\n", cap_buf);
1844
1845 sccp_codec_multiple2str(cap_buf, sizeof(cap_buf) - 1, videoCapabilities, ARRAY_LEN(videoCapabilities));
1846 sccp_log(DEBUGCAT_CODEC) (VERBOSE_PREFIX_4 "remote video caps: %s\n", cap_buf);
1847
1848 //sccp_log(DEBUGCAT_CODEC) (VERBOSE_PREFIX_4 "SCCP: requestedCodec in Skinny Format: audio:%s (video:%s)\n", codec2name(audio_codec), codec2name(video_codec));
1849 /** done getting remote capabilities */
1850
1851 callid = ast_read_threadstorage_callid();
1852
1853 AUTO_RELEASE(sccp_channel_t, channel , NULL);
1854 requestStatus = sccp_requestChannel(lineName, autoanswer_type, autoanswer_cause, ringermode, &channel);
1855 switch (requestStatus) {
1856 case SCCP_REQUEST_STATUS_SUCCESS: // everything is fine
1857 break;
1858 case SCCP_REQUEST_STATUS_LINEUNKNOWN:
1859 sccp_log(DEBUGCAT_CORE) (VERBOSE_PREFIX_4 "SCCP: sccp_requestChannel returned Line %s Unknown -> Not Successfull\n", lineName);
1860 *cause = AST_CAUSE_DESTINATION_OUT_OF_ORDER;
1861 goto EXITFUNC;
1862 case SCCP_REQUEST_STATUS_LINEUNAVAIL:
1863 sccp_log(DEBUGCAT_CORE) (VERBOSE_PREFIX_4 "SCCP: sccp_requestChannel returned Line %s not currently registered -> Try again later\n", lineName);
1864 *cause = AST_CAUSE_REQUESTED_CHAN_UNAVAIL;
1865 goto EXITFUNC;
1866 case SCCP_REQUEST_STATUS_ERROR:
1867 pbx_log(LOG_ERROR, "SCCP: sccp_requestChannel returned Status Error for lineName: %s\n", lineName);
1868 *cause = AST_CAUSE_UNALLOCATED;
1869 goto EXITFUNC;
1870 default:
1871 pbx_log(LOG_ERROR, "SCCP: sccp_requestChannel returned Status Error for lineName: %s\n", lineName);
1872 *cause = AST_CAUSE_UNALLOCATED;
1873 goto EXITFUNC;
1874 }
1875 #ifdef CS_SCCP_VIDEO
1876 memset(&channel->remoteCapabilities.video, 0, sizeof(channel->remoteCapabilities.video));
1877 if (videoCapabilities[0] != SKINNY_CODEC_NONE) {
1878 memcpy(channel->remoteCapabilities.video, videoCapabilities, ARRAY_LEN(videoCapabilities));
1879 } else if (video_codec) {
1880 channel->remoteCapabilities.video[0] = video_codec;
1881 }
1882 #endif
1883 if (callid) {
1884 channel->pbx_callid = (pbx_callid_t)callid;
1885 }
1886
1887 if (!sccp_pbx_channel_allocate(channel, assignedids, requestor)) {
1888 //! \todo handle error in more detail, cleanup sccp channel
1889 *cause = AST_CAUSE_REQUESTED_CHAN_UNAVAIL;
1890 goto EXITFUNC;
1891 }
1892
1893 /* set initial connected line information, to be exchange with remove party during first CONNECTED_LINE update */
1894 ast_set_callerid(channel->owner, channel->line->cid_num, channel->line->cid_name, channel->line->cid_num);
1895 struct ast_party_connected_line connected;
1896 ast_party_connected_line_set_init(&connected, ast_channel_connected(channel->owner));
1897 connected.id.number.valid = 1;
1898 connected.id.number.str = (char *)channel->line->cid_num;
1899 connected.id.number.presentation = AST_PRES_ALLOWED_NETWORK_NUMBER;
1900 connected.id.name.valid = 1;
1901 connected.id.name.str = (char *)channel->line->cid_name;
1902 connected.id.name.presentation = AST_PRES_ALLOWED_NETWORK_NUMBER;
1903 connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_UNKNOWN;
1904 ast_channel_set_connected_line(channel->owner, &connected, NULL);
1905 /* end */
1906
1907 if (requestor) {
1908 /* set calling party */
1909 sccp_callinfo_t *ci = sccp_channel_getCallInfo(channel);
1910 iCallInfo.Setter(ci,
1911 SCCP_CALLINFO_CALLINGPARTY_NAME, ast_channel_caller((PBX_CHANNEL_TYPE *) requestor)->id.name.str,
1912 SCCP_CALLINFO_CALLINGPARTY_NUMBER, ast_channel_caller((PBX_CHANNEL_TYPE *) requestor)->id.number.str,
1913 SCCP_CALLINFO_ORIG_CALLEDPARTY_NAME, ast_channel_redirecting((PBX_CHANNEL_TYPE *) requestor)->orig.name.str,
1914 SCCP_CALLINFO_ORIG_CALLEDPARTY_NUMBER, ast_channel_redirecting((PBX_CHANNEL_TYPE *) requestor)->orig.number.str,
1915 SCCP_CALLINFO_KEY_SENTINEL);
1916 }
1917
1918 /** workaround for asterisk console log flooded
1919 channel.c:5080 ast_write: Codec mismatch on channel SCCP/xxx-0000002d setting write format to g722 from unknown native formats (nothing)
1920 */
1921 if (!channel->capabilities.audio[0]) {
1922 skinny_codec_t codecs[SKINNY_MAX_CAPABILITIES] = { SKINNY_CODEC_WIDEBAND_256K, SKINNY_CODEC_NONE};
1923 sccp_astwrap_setNativeAudioFormats(channel, codecs);
1924 sccp_astwrap_setReadFormat(channel, SKINNY_CODEC_WIDEBAND_256K);
1925 sccp_astwrap_setWriteFormat(channel, SKINNY_CODEC_WIDEBAND_256K);
1926 }
1927
1928 /* get remote codecs from channel driver */
1929 //ast_rtp_instance_get_codecs(c->rtp.adio.rtp);
1930 //ast_rtp_instance_get_codecs(c->rtp.video.instance);
1931 /** done */
1932
1933 EXITFUNC:
1934 if (channel && channel->owner) {
1935 result_ast_channel = channel->owner;
1936 if (callid) {
1937 ast_channel_callid_set(result_ast_channel, callid);
1938 }
1939 } else if (callid) {
1940 #if !defined(CS_AST_CHANNEL_CALLID_TYPEDEF)
1941 ast_callid_unref(callid);
1942 #endif
1943 }
1944 return result_ast_channel;
1945 }
1946
sccp_astwrap_call(PBX_CHANNEL_TYPE * ast,const char * dest,int timeout)1947 static int sccp_astwrap_call(PBX_CHANNEL_TYPE * ast, const char *dest, int timeout)
1948 {
1949 int res = 0;
1950
1951 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: Asterisk request to call %s (dest:%s, timeout: %d)\n", pbx_channel_name(ast), dest, timeout);
1952
1953 if (!sccp_strlen_zero(pbx_channel_call_forward(ast))) {
1954 iPbx.queue_control(ast, (enum ast_control_frame_type) -1); /* Prod Channel if in the middle of a call_forward instead of proceed */
1955 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: Forwarding Call to '%s'\n", pbx_channel_call_forward(ast));
1956 return 0;
1957 }
1958
1959 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast));
1960 if (!c) {
1961 pbx_log(LOG_WARNING, "SCCP: Asterisk request to call %s on channel: %s, but we don't have this channel!\n", dest, pbx_channel_name(ast));
1962 return -1;
1963 }
1964
1965 /* Check whether there is MaxCallBR variables */
1966 const char *MaxCallBRStr = pbx_builtin_getvar_helper(ast, "MaxCallBR");
1967 if (MaxCallBRStr && !sccp_strlen_zero(MaxCallBRStr)) {
1968 sccp_astgenwrap_channel_write(ast, "CHANNEL", "MaxCallBR", MaxCallBRStr);
1969 }
1970
1971 res = sccp_pbx_call(c, (char *) dest, timeout);
1972 return res;
1973 }
1974
1975 /*
1976 * Remote side has answered the call switch to up state
1977 * Is being called with pbxchan locked by app_dial
1978 */
sccp_astwrap_answer(PBX_CHANNEL_TYPE * pbxchan)1979 static int sccp_astwrap_answer(PBX_CHANNEL_TYPE * pbxchan)
1980 {
1981 int res = -1;
1982 int timedout = 0;
1983 if(pbx_channel_state(pbxchan) == AST_STATE_UP) {
1984 pbx_log(LOG_NOTICE, "%s: Channel has already been answered remotely, skipping\n", pbx_channel_name(pbxchan));
1985 return 0;
1986 }
1987
1988 pbx_channel_ref(pbxchan);
1989 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(pbxchan));
1990 if(c && c->state < SCCP_GROUPED_CHANNELSTATE_CONNECTION) {
1991 sccp_log(DEBUGCAT_CORE)(VERBOSE_PREFIX_3 "%s: Remote has answered the call.\n", c->designator);
1992
1993 AUTO_RELEASE(sccp_device_t, d, sccp_channel_getDevice(c));
1994 if(d && d->session) {
1995 sccp_log(DEBUGCAT_PBX)(VERBOSE_PREFIX_3 "%s: Waiting for pendingRequests\n", c->designator);
1996 // this needs to be done with the pbx_channel unlocked to prevent lock investion
1997 // note we still have a pbx_channel_ref, so the channel cannot be removed under our feet
1998 pbx_channel_unlock(pbxchan);
1999 timedout = sccp_session_waitForPendingRequests(d->session);
2000 pbx_channel_lock(pbxchan);
2001 }
2002 if (!timedout) {
2003 // pbx_indicate(pbxchan, AST_CONTROL_PROGRESS);
2004 res = sccp_pbx_remote_answer(c);
2005 }
2006 }
2007 pbx_channel_unref(pbxchan);
2008 return res;
2009 }
2010
2011 /**
2012 *
2013 * \todo update remote capabilities after fixup
2014 */
sccp_astwrap_fixup(PBX_CHANNEL_TYPE * oldchan,PBX_CHANNEL_TYPE * newchan)2015 static int sccp_astwrap_fixup(PBX_CHANNEL_TYPE * oldchan, PBX_CHANNEL_TYPE * newchan)
2016 {
2017 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: we got a fixup request for %s and %s\n", pbx_channel_name(oldchan), pbx_channel_name(newchan));
2018 int res = 0;
2019
2020 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(newchan));
2021 if (c) {
2022 if (c->owner != oldchan) {
2023 ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, c->owner);
2024 res = -1;
2025 } else {
2026 /* during a masquerade, fixup gets called twice */
2027 if(ast_channel_masqr(newchan)) { /* this is the channel that is masquaraded out */
2028 // set to simple channel requestHangup, we are out of pbx_run_pbx, upon returning from masquerade */
2029 // sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "%s: set c->hangupRequest = requestHangup\n", c->designator);
2030 c->hangupRequest = sccp_astgenwrap_requestHangup;
2031
2032 if (pbx_channel_hangupcause(newchan) == AST_CAUSE_ANSWERED_ELSEWHERE) {
2033 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "%s: Fixup Adding Redirecting Party from:%s\n", c->designator, pbx_channel_name(oldchan));
2034 iCallInfo.Setter(sccp_channel_getCallInfo(c),
2035 SCCP_CALLINFO_HUNT_PILOT_NAME, ast_channel_caller(oldchan)->id.name.str,
2036 SCCP_CALLINFO_HUNT_PILOT_NUMBER, ast_channel_caller(oldchan)->id.number.str,
2037 SCCP_CALLINFO_LAST_REDIRECTINGPARTY_NAME, ast_channel_caller(oldchan)->id.name.str,
2038 SCCP_CALLINFO_LAST_REDIRECTINGPARTY_NUMBER, ast_channel_caller(oldchan)->id.number.str,
2039 SCCP_CALLINFO_LAST_REDIRECT_REASON, 5,
2040 SCCP_CALLINFO_KEY_SENTINEL);
2041 }
2042 } else {
2043 // set channel requestHangup to use ast_hangup (as it will not be part of __ast_pbx_run, upon returning from masquerade) */
2044 // sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "%s: set c->hangupRequest = requestQueueHangup\n", c->designator);
2045 c->hangupRequest = sccp_astgenwrap_requestQueueHangup;
2046
2047 if (!sccp_strlen_zero(c->line->language)) {
2048 ast_channel_language_set(newchan, c->line->language);
2049 }
2050
2051 //! \todo update remote capabilities after fixup
2052 pbx_retrieve_remote_capabilities(c);
2053
2054 /* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native
2055 RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be
2056 able to do this if the masquerade happens before the bridge breaks (e.g., AMI
2057 redirect of both channels). Note that a channel can not be masqueraded *into*
2058 a native bridge. So there is no danger that this breaks a native bridge that
2059 should stay up. */
2060 // sccp_astwrap_update_rtp_peer(newchan, NULL, NULL, 0, 0, 0);
2061 }
2062 sccp_astwrap_setOwner(c, newchan);
2063 }
2064 } else {
2065 pbx_log(LOG_WARNING, "sccp_pbx_fixup(old: %s(%p), new: %s(%p)). no SCCP channel to fix\n", pbx_channel_name(oldchan), (void *) oldchan, pbx_channel_name(newchan), (void *) newchan);
2066 res = -1;
2067 }
2068 return res;
2069 }
2070
2071 #if 0
2072 #ifdef CS_AST_RTP_INSTANCE_BRIDGE
2073 static enum ast_bridge_result sccp_astwrap_rtpBridge(PBX_CHANNEL_TYPE * c0, PBX_CHANNEL_TYPE * c1, int flags, PBX_FRAME_TYPE ** fo, PBX_CHANNEL_TYPE ** rc, int timeoutms)
2074 {
2075 enum ast_bridge_result res;
2076 int new_flags = flags;
2077
2078 /* \note temporarily marked out until we figure out how to get directrtp back on track - DdG */
2079 AUTO_RELEASE(sccp_channel_t, sc0 , get_sccp_channel_from_pbx_channel(c0));
2080 AUTO_RELEASE(sccp_channel_t, sc1 , get_sccp_channel_from_pbx_channel(c1));
2081
2082 if ((sc0 && sc1)) {
2083 // Switch off DTMF between SCCP phones
2084 new_flags &= !AST_BRIDGE_DTMF_CHANNEL_0;
2085 new_flags &= !AST_BRIDGE_DTMF_CHANNEL_1;
2086 if (GLOB(directrtp)) {
2087 ast_channel_defer_dtmf(c0);
2088 ast_channel_defer_dtmf(c1);
2089 } else {
2090 AUTO_RELEASE(sccp_device_t, d0 , sccp_channel_getDevice(sc0));
2091 AUTO_RELEASE(sccp_device_t, d1 , sccp_channel_getDevice(sc1));
2092 if ((d0 && d1) && ((d0->directrtp && d1->directrtp)) {
2093 ast_channel_defer_dtmf(c0);
2094 ast_channel_defer_dtmf(c1);
2095 }
2096 }
2097 sc0->peerIsSCCP = TRUE;
2098 sc1->peerIsSCCP = TRUE;
2099 // SCCP Key handle direction to asterisk is still to be implemented here
2100 // sccp_pbx_senddigit
2101 } else {
2102 // Switch on DTMF between differing channels
2103 ast_channel_undefer_dtmf(c0);
2104 ast_channel_undefer_dtmf(c1);
2105 }
2106 //res = ast_rtp_bridge(c0, c1, new_flags, fo, rc, timeoutms);
2107 res = ast_rtp_instance_bridge(c0, c1, new_flags, fo, rc, timeoutms);
2108 switch (res) {
2109 case AST_BRIDGE_COMPLETE:
2110 sccp_log((DEBUGCAT_PBX)) (VERBOSE_PREFIX_1 "SCCP: Bridge chan %s and chan %s: Complete\n", ast_channel_name(c0), ast_channel_name(c1));
2111 break;
2112 case AST_BRIDGE_FAILED:
2113 sccp_log((DEBUGCAT_PBX)) (VERBOSE_PREFIX_1 "SCCP: Bridge chan %s and chan %s: Failed\n", ast_channel_name(c0), ast_channel_name(c1));
2114 break;
2115 case AST_BRIDGE_FAILED_NOWARN:
2116 sccp_log((DEBUGCAT_PBX)) (VERBOSE_PREFIX_1 "SCCP: Bridge chan %s and chan %s: Failed NoWarn\n", ast_channel_name(c0), ast_channel_name(c1));
2117 break;
2118 case AST_BRIDGE_RETRY:
2119 sccp_log((DEBUGCAT_PBX)) (VERBOSE_PREFIX_1 "SCCP: Bridge chan %s and chan %s: Failed Retry\n", ast_channel_name(c0), ast_channel_name(c1));
2120 break;
2121 }
2122 /*! \todo Implement callback function queue upon completion */
2123 return res;
2124 }
2125 #endif
2126 #endif
2127
sccp_astwrap_get_rtp_info(PBX_CHANNEL_TYPE * ast,PBX_RTP_TYPE ** rtp)2128 static enum ast_rtp_glue_result sccp_astwrap_get_rtp_info(PBX_CHANNEL_TYPE * ast, PBX_RTP_TYPE ** rtp)
2129 {
2130 //AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast)); // not following the refcount rules... channel is already retained
2131 sccp_channel_t *c = NULL;
2132 sccp_rtp_info_t rtpInfo = SCCP_RTP_INFO_NORTP;
2133 struct sccp_rtp *audioRTP = NULL;
2134 enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_REMOTE;
2135
2136 if (!(c = CS_AST_CHANNEL_PVT(ast))) {
2137 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "SCCP: (get_rtp_info) NO PVT\n");
2138 return AST_RTP_GLUE_RESULT_FORBID;
2139 }
2140
2141 if (pbx_channel_state(ast) != AST_STATE_UP) {
2142 sccp_log((DEBUGCAT_CHANNEL | DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (get_rtp_info) Asterisk requested EarlyRTP peer for channel %s\n", c->currentDeviceId, pbx_channel_name(ast));
2143 return AST_RTP_GLUE_RESULT_LOCAL;
2144 } else {
2145 sccp_log((DEBUGCAT_CHANNEL | DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (get_rtp_info) Asterisk requested RTP peer for channel %s\n", c->currentDeviceId, pbx_channel_name(ast));
2146 }
2147
2148 rtpInfo = sccp_rtp_getAudioPeerInfo(c, &audioRTP);
2149 if (rtpInfo == SCCP_RTP_INFO_NORTP) {
2150 sccp_log(DEBUGCAT_RTP)(VERBOSE_PREFIX_2 "%s: (get_rtp_info) got SCCP_RTP_INFO_NORTP => returning AST_RTP_GLUE_RESULT_FORBID\n", c->designator);
2151 return AST_RTP_GLUE_RESULT_FORBID;
2152 }
2153
2154 *rtp = audioRTP->instance;
2155 if (!*rtp) {
2156 sccp_log(DEBUGCAT_RTP)(VERBOSE_PREFIX_2 "%s: (get_rtp_info) no rtp instance yet => returning AST_RTP_GLUE_RESULT_FORBID\n", c->designator);
2157 return AST_RTP_GLUE_RESULT_FORBID;
2158 }
2159 #ifdef HAVE_PBX_RTP_ENGINE_H
2160 ao2_ref(*rtp, +1);
2161 #endif
2162 // struct ast_sockaddr ast_sockaddr_tmp;
2163 // ast_rtp_instance_get_remote_address(*rtp, &ast_sockaddr_tmp);
2164 // sccp_log((DEBUGCAT_RTP | DEBUGCAT_HIGH)) (VERBOSE_PREFIX_3 "%s: (get_rtp_info) remote address:'%s:%d'\n", c->currentDeviceId, ast_sockaddr_stringify_host(&ast_sockaddr_tmp), ast_sockaddr_port(&ast_sockaddr_tmp));
2165
2166 if (ast_test_flag(GLOB(global_jbconf), AST_JB_FORCED)) {
2167 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (get_rtp_info) JitterBuffer is Forced. AST_RTP_GET_FAILED\n", c->currentDeviceId);
2168 return AST_RTP_GLUE_RESULT_LOCAL;
2169 }
2170
2171 if (!(rtpInfo & SCCP_RTP_INFO_ALLOW_DIRECTRTP)) {
2172 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (get_rtp_info) Direct RTP disabled -> Using AST_RTP_TRY_PARTIAL for channel %s\n", c->currentDeviceId, pbx_channel_name(ast));
2173 return AST_RTP_GLUE_RESULT_LOCAL;
2174 }
2175
2176 sccp_log((DEBUGCAT_RTP | DEBUGCAT_HIGH)) (VERBOSE_PREFIX_1 "%s: (get_rtp_info) Channel %s Returning res: %s\n", c->currentDeviceId, pbx_channel_name(ast), ((res == 2) ? "indirect-rtp" : ((res == 1) ? "direct-rtp" : "forbid")));
2177 return res;
2178 }
2179
sccp_astwrap_get_vrtp_info(PBX_CHANNEL_TYPE * ast,PBX_RTP_TYPE ** rtp)2180 static enum ast_rtp_glue_result sccp_astwrap_get_vrtp_info(PBX_CHANNEL_TYPE * ast, PBX_RTP_TYPE ** rtp)
2181 {
2182 //AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast)); // not following the refcount rules... channel is already retained
2183 sccp_channel_t *c = NULL;
2184 sccp_rtp_info_t rtpInfo = SCCP_RTP_INFO_NORTP;
2185 struct sccp_rtp *videoRTP = NULL;
2186 enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_REMOTE;
2187
2188 if (!(c = CS_AST_CHANNEL_PVT(ast))) {
2189 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "SCCP: (get_vrtp_info) NO PVT\n");
2190 return AST_RTP_GLUE_RESULT_FORBID;
2191 }
2192
2193 if (pbx_channel_state(ast) != AST_STATE_UP) {
2194 sccp_log((DEBUGCAT_CHANNEL | DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (get_vrtp_info) Asterisk requested EarlyRTP peer for channel %s\n", c->currentDeviceId, pbx_channel_name(ast));
2195 } else {
2196 sccp_log((DEBUGCAT_CHANNEL | DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (get_vrtp_info) Asterisk requested RTP peer for channel %s\n", c->currentDeviceId, pbx_channel_name(ast));
2197 }
2198
2199 #ifdef CS_SCCP_VIDEO
2200 rtpInfo = sccp_rtp_getVideoPeerInfo(c, &videoRTP);
2201 #endif
2202 if (rtpInfo == SCCP_RTP_INFO_NORTP) {
2203 return AST_RTP_GLUE_RESULT_FORBID;
2204 }
2205
2206 *rtp = videoRTP->instance;
2207 if (!*rtp) {
2208 return AST_RTP_GLUE_RESULT_FORBID;
2209 }
2210 #ifdef HAVE_PBX_RTP_ENGINE_H
2211 ao2_ref(*rtp, +1);
2212 #endif
2213 // struct ast_sockaddr ast_sockaddr_tmp;
2214 // ast_rtp_instance_get_remote_address(*rtp, &ast_sockaddr_tmp);
2215 // sccp_log((DEBUGCAT_RTP | DEBUGCAT_HIGH)) (VERBOSE_PREFIX_3 "%s: (get_vrtp_info) remote address:'%s:%d'\n", c->currentDeviceId, ast_sockaddr_stringify_host(&ast_sockaddr_tmp), ast_sockaddr_port(&ast_sockaddr_tmp));
2216 if (ast_test_flag(GLOB(global_jbconf), AST_JB_FORCED)) {
2217 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (get_vrtp_info) JitterBuffer is Forced. AST_RTP_GET_FAILED\n", c->currentDeviceId);
2218 return AST_RTP_GLUE_RESULT_FORBID;
2219 }
2220
2221 if (!(rtpInfo & SCCP_RTP_INFO_ALLOW_DIRECTRTP)) {
2222 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (get_vrtp_info) Direct RTP disabled -> Using AST_RTP_TRY_PARTIAL for channel %s\n", c->currentDeviceId, pbx_channel_name(ast));
2223 return AST_RTP_GLUE_RESULT_LOCAL;
2224 }
2225
2226 sccp_log((DEBUGCAT_RTP | DEBUGCAT_HIGH)) (VERBOSE_PREFIX_1 "%s: (get_vrtp_info) Channel %s Returning res: %s\n", c->currentDeviceId, pbx_channel_name(ast), ((res == 2) ? "indirect-rtp" : ((res == 1) ? "direct-rtp" : "forbid")));
2227 return res;
2228 }
2229
sccp_astwrap_update_rtp_peer(PBX_CHANNEL_TYPE * ast,PBX_RTP_TYPE * rtp,PBX_RTP_TYPE * vrtp,PBX_RTP_TYPE * trtp,const struct ast_format_cap * codecs,int nat_active)2230 static int sccp_astwrap_update_rtp_peer(PBX_CHANNEL_TYPE * ast, PBX_RTP_TYPE * rtp, PBX_RTP_TYPE * vrtp, PBX_RTP_TYPE * trtp, const struct ast_format_cap *codecs, int nat_active)
2231 {
2232 //AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast)); // not following the refcount rules... channel is already retained
2233 sccp_channel_t *c = NULL;
2234 int result = 0;
2235 do {
2236 pbx_str_t *codec_buf = pbx_str_alloca(64);
2237
2238 if (!(c = CS_AST_CHANNEL_PVT(ast))) {
2239 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "SCCP: (update_rtp_peer) NO PVT\n");
2240 result = -1;
2241 break;
2242 }
2243
2244 if (!codecs || ast_format_cap_count(codecs) <= 0) {
2245 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (update_rtp_peer) NO Codecs\n", c->currentDeviceId);
2246 result = -1;
2247 break;
2248 }
2249 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_2 "%s: (update_rtp_peer) stage: %s, remote codecs capabilty: %s, nat_active: %d\n", c->currentDeviceId,
2250 S_COR(AST_STATE_UP == pbx_channel_state(ast), "RTP", "EarlyRTP"), ast_format_cap_get_names((struct ast_format_cap *) codecs, &codec_buf), nat_active);
2251
2252 if (!c->line) {
2253 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (update_rtp_peer) NO LINE\n", c->currentDeviceId);
2254 result = -1;
2255 break;
2256 }
2257 if (c->state == SCCP_CHANNELSTATE_HOLD) {
2258 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (update_rtp_peer) On Hold -> No Update\n", c->currentDeviceId);
2259 result = 0;
2260 break;
2261 }
2262 AUTO_RELEASE(sccp_device_t, d , sccp_channel_getDevice(c));
2263 if (!d) {
2264 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (update_rtp_peer) NO DEVICE\n", c->currentDeviceId);
2265 result = -1;
2266 break;
2267 }
2268
2269 if (!rtp && !vrtp && !trtp) {
2270 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (update_rtp_peer) RTP not ready\n", c->currentDeviceId);
2271 result = 0;
2272 break;
2273 }
2274
2275 PBX_RTP_TYPE *instance = { 0, };
2276 struct sockaddr_storage sas = { 0, };
2277 //struct sockaddr_in sin = { 0, };
2278 struct ast_sockaddr sin_tmp;
2279 boolean_t directmedia = FALSE;
2280
2281 if (rtp) { // generalize input
2282 instance = rtp;
2283 } else if (vrtp) {
2284 instance = vrtp;
2285 #ifdef CS_SCCP_VIDEO
2286 /* video requested by remote side, let's see if we support video */
2287 /* should be moved to sccp_rtp.c */
2288 /*
2289 if (ast_format_cap_has_type(codecs, AST_MEDIA_TYPE_VIDEO) && sccp_device_isVideoSupported(d) && sccp_channel_getVideoMode(c) == SCCP_VIDEO_MODE_AUTO) {
2290 if (!c->rtp.video.instance && !sccp_rtp_createVideoServer(c)) {
2291 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_3 "%s: can not start vrtp\n", DEV_ID_LOG(d));
2292 } else {
2293 if (!c->rtp.video.transmission.state) {
2294 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_3 "%s: video rtp started\n", DEV_ID_LOG(d));
2295 sccp_channel_startMultiMediaTransmission(c);
2296 }
2297 }
2298 }
2299 */
2300 #endif
2301 } else {
2302 instance = trtp;
2303 }
2304
2305 if (d->directrtp && d->nat < SCCP_NAT_ON && !nat_active && !c->conference) { // assume directrtp
2306 ast_rtp_instance_get_remote_address(instance, &sin_tmp);
2307 memcpy(&sas, &sin_tmp, sizeof(struct sockaddr_storage));
2308 //ast_sockaddr_to_sin(&sin_tmp, &sin);
2309 if (d->nat == SCCP_NAT_OFF) { // forced nat off to circumvent autodetection + direcrtp, requires checking both phone_ip and external session ip address against devices permit/deny
2310 struct ast_sockaddr sin_local;
2311 struct sockaddr_storage localsas = { 0, };
2312 ast_rtp_instance_get_local_address(instance, &sin_local);
2313 memcpy(&localsas, &sin_local, sizeof(struct sockaddr_storage));
2314 if (sccp_apply_ha(d->ha, &sas) == AST_SENSE_ALLOW && sccp_apply_ha(d->ha, &localsas) == AST_SENSE_ALLOW) {
2315 directmedia = TRUE;
2316 }
2317 } else if (sccp_apply_ha(d->ha, &sas) == AST_SENSE_ALLOW) { // check remote sin against local device acl (to match netmask)
2318 directmedia = TRUE;
2319 // ast_channel_defer_dtmf(ast);
2320 }
2321 }
2322 if (!directmedia) { // fallback to indirectrtp
2323 ast_rtp_instance_get_local_address(instance, &sin_tmp);
2324 memcpy(&sas, &sin_tmp, sizeof(struct sockaddr_storage));
2325 sccp_session_getOurIP(d->session, &sas, sccp_netsock_is_IPv4(&sas) ? AF_INET : AF_INET6);
2326 }
2327
2328 sccp_log((DEBUGCAT_RTP)) (VERBOSE_PREFIX_1 "%s: (update_rtp_peer) new remote rtp ip = '%s'\n (d->directrtp: %s && !d->nat: %s && !remote->nat_active: %s && d->acl_allow: %s) => directmedia=%s\n", c->currentDeviceId, sccp_netsock_stringify(&sas), S_COR(d->directrtp, "yes", "no"),
2329 sccp_nat2str(d->nat),
2330 S_COR(!nat_active, "yes", "no"), S_COR(directmedia, "yes", "no"), S_COR(directmedia, "yes", "no")
2331 );
2332
2333 if (rtp) { // send peer info to phone
2334 sccp_rtp_set_peer(c, &c->rtp.audio, &sas);
2335 c->rtp.audio.directMedia = directmedia;
2336 } else if (vrtp) {
2337 sccp_rtp_set_peer(c, &c->rtp.video, &sas);
2338 c->rtp.video.directMedia = directmedia;
2339 } else {
2340 //sccp_rtp_set_peer(c, &c->rtp.text, &sas);
2341 //c->rtp.text.directMedia = directmedia;
2342 }
2343 } while (0);
2344
2345 /* Need a return here to break the bridge */
2346 sccp_log(DEBUGCAT_RTP) (VERBOSE_PREFIX_3 "%s: (update_rtp_peer) Result: %d\n", c->currentDeviceId, result);
2347 return result;
2348 }
2349
sccp_astwrap_getCodec(PBX_CHANNEL_TYPE * ast,struct ast_format_cap * result)2350 static void sccp_astwrap_getCodec(PBX_CHANNEL_TYPE * ast, struct ast_format_cap *result)
2351 {
2352 uint8_t i;
2353 AUTO_RELEASE(sccp_channel_t, channel , get_sccp_channel_from_pbx_channel(ast));
2354
2355 if (!channel) {
2356 sccp_log((DEBUGCAT_RTP | DEBUGCAT_CODEC)) (VERBOSE_PREFIX_1 "SCCP: (getCodec) NO PVT\n");
2357 return;
2358 }
2359
2360 ast_debug(10, "asterisk requests format for channel %s, readFormat: %s(%d)\n", pbx_channel_name(ast), codec2str(channel->rtp.audio.transmission.format), channel->rtp.audio.transmission.format);
2361 for (i = 0; i < ARRAY_LEN(channel->preferences.audio); i++) {
2362 struct ast_format *ast_format = sccp_astwrap_skinny2ast_format(channel->preferences.audio[i]);
2363 if (ast_format != ast_format_none) {
2364 unsigned int framing = ast_format_get_default_ms(ast_format);
2365 ast_format_cap_append(result, ast_format, framing);
2366 }
2367 }
2368 for (i = 0; i < ARRAY_LEN(channel->preferences.video); i++) {
2369 struct ast_format *ast_format = sccp_astwrap_skinny2ast_format(channel->preferences.video[i]);
2370 if (ast_format != ast_format_none) {
2371 unsigned int framing = ast_format_get_default_ms(ast_format);
2372 ast_format_cap_append(result, ast_format, framing);
2373 }
2374 }
2375
2376 return;
2377 }
2378
2379 /*
2380 * \brief get callerid_name from pbx
2381 * \param sccp_channle Asterisk Channel
2382 * \param cid name result
2383 * \return parse result
2384 */
sccp_astwrap_callerid_name(PBX_CHANNEL_TYPE * pbx_chan,char ** cid_name)2385 static int sccp_astwrap_callerid_name(PBX_CHANNEL_TYPE *pbx_chan, char **cid_name)
2386 {
2387 if (pbx_chan && ast_channel_caller(pbx_chan)->id.name.str && strlen(ast_channel_caller(pbx_chan)->id.name.str) > 0) {
2388 *cid_name = pbx_strdup(ast_channel_caller(pbx_chan)->id.name.str);
2389 return 1;
2390 }
2391
2392 return 0;
2393 }
2394
2395 /*
2396 * \brief get callerid_name from pbx
2397 * \param ast_chan Asterisk Channel
2398 * \return char * with the caller number
2399 */
sccp_astwrap_callerid_number(PBX_CHANNEL_TYPE * pbx_chan,char ** cid_number)2400 static int sccp_astwrap_callerid_number(PBX_CHANNEL_TYPE *pbx_chan, char **cid_number)
2401 {
2402 if (pbx_chan && ast_channel_caller(pbx_chan)->id.number.str && strlen(ast_channel_caller(pbx_chan)->id.number.str) > 0) {
2403 *cid_number = pbx_strdup(ast_channel_caller(pbx_chan)->id.number.str);
2404 return 1;
2405 }
2406
2407 return 0;
2408 }
2409
2410 /*
2411 * \brief get callerid_ton from pbx
2412 * \param ast_chan Asterisk Channel
2413 * \return char * with the caller number
2414 */
sccp_astwrap_callerid_ton(PBX_CHANNEL_TYPE * pbx_chan,int * cid_ton)2415 static int sccp_astwrap_callerid_ton(PBX_CHANNEL_TYPE *pbx_chan, int *cid_ton)
2416 {
2417 if (pbx_chan && ast_channel_caller(pbx_chan)->id.number.valid) {
2418 *cid_ton = ast_channel_caller(pbx_chan)->ani.number.plan;
2419 return *cid_ton;
2420 }
2421 return 0;
2422 }
2423
2424 /*
2425 * \brief get callerid_ani from pbx
2426 * \param ast_chan Asterisk Channel
2427 * \return char * with the caller number
2428 */
sccp_astwrap_callerid_ani(PBX_CHANNEL_TYPE * pbx_chan,char ** cid_ani)2429 static int sccp_astwrap_callerid_ani(PBX_CHANNEL_TYPE *pbx_chan, char **cid_ani)
2430 {
2431 if (pbx_chan && ast_channel_caller(pbx_chan)->ani.number.valid && ast_channel_caller(pbx_chan)->ani.number.str && strlen(ast_channel_caller(pbx_chan)->ani.number.str) > 0) {
2432 *cid_ani = pbx_strdup(ast_channel_caller(pbx_chan)->ani.number.str);
2433 return 1;
2434 }
2435
2436 return 0;
2437 }
2438
2439 /*
2440 * \brief get callerid_dnid from pbx
2441 * \param ast_chan Asterisk Channel
2442 * \return char * with the caller number
2443 */
sccp_astwrap_callerid_subaddr(PBX_CHANNEL_TYPE * pbx_chan,char ** cid_subaddr)2444 static int sccp_astwrap_callerid_subaddr(PBX_CHANNEL_TYPE *pbx_chan, char **cid_subaddr)
2445 {
2446 if (pbx_chan && ast_channel_caller(pbx_chan)->id.subaddress.valid && ast_channel_caller(pbx_chan)->id.subaddress.str && strlen(ast_channel_caller(pbx_chan)->id.subaddress.str) > 0) {
2447 *cid_subaddr = pbx_strdup(ast_channel_caller(pbx_chan)->id.subaddress.str);
2448 return 1;
2449 }
2450
2451 return 0;
2452 }
2453
2454 /*
2455 * \brief get callerid_dnid from pbx (Destination ID)
2456 * \param ast_chan Asterisk Channel
2457 * \return char * with the caller number
2458 */
sccp_astwrap_callerid_dnid(PBX_CHANNEL_TYPE * pbx_chan,char ** cid_dnid)2459 static int sccp_astwrap_callerid_dnid(PBX_CHANNEL_TYPE *pbx_chan, char **cid_dnid)
2460 {
2461 if (pbx_chan && ast_channel_dialed(pbx_chan)->number.str && strlen(ast_channel_dialed(pbx_chan)->number.str) > 0) {
2462 *cid_dnid = pbx_strdup(ast_channel_dialed(pbx_chan)->number.str);
2463 return 1;
2464 }
2465
2466 return 0;
2467 }
2468
2469 /*
2470 * \brief get callerid_rdnis from pbx
2471 * \param ast_chan Asterisk Channel
2472 * \return char * with the caller number
2473 */
sccp_astwrap_callerid_rdnis(PBX_CHANNEL_TYPE * pbx_chan,char ** cid_rdnis)2474 static int sccp_astwrap_callerid_rdnis(PBX_CHANNEL_TYPE *pbx_chan, char **cid_rdnis)
2475 {
2476 if (pbx_chan && ast_channel_redirecting(pbx_chan)->from.number.valid && ast_channel_redirecting(pbx_chan)->from.number.str && strlen(ast_channel_redirecting(pbx_chan)->from.number.str) > 0) {
2477 *cid_rdnis = pbx_strdup(ast_channel_redirecting(pbx_chan)->from.number.str);
2478 return 1;
2479 }
2480
2481 return 0;
2482 }
2483
2484 /*
2485 * \brief get callerid_presence from pbx
2486 * \param ast_chan Asterisk Channel
2487 * \return char * with the caller number
2488 */
sccp_astwrap_callerid_presentation(PBX_CHANNEL_TYPE * pbx_chan)2489 static sccp_callerid_presentation_t sccp_astwrap_callerid_presentation(PBX_CHANNEL_TYPE *pbx_chan)
2490 {
2491 if (pbx_chan && (ast_party_id_presentation(&ast_channel_caller(pbx_chan)->id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
2492 return CALLERID_PRESENTATION_ALLOWED;
2493 }
2494 return CALLERID_PRESENTATION_FORBIDDEN;
2495 }
2496
sccp_astwrap_createRtpInstance(constDevicePtr d,constChannelPtr c,sccp_rtp_t * rtp)2497 static boolean_t sccp_astwrap_createRtpInstance(constDevicePtr d, constChannelPtr c, sccp_rtp_t *rtp)
2498 {
2499 uint32_t tos = 0, cos = 0;
2500
2501 if (!c || !d) {
2502 return FALSE;
2503 }
2504 struct sockaddr_storage ourip = { 0 };
2505 sccp_session_getOurIP(d->session, &ourip, 0);
2506 struct ast_sockaddr sock = { {
2507 0,
2508 } };
2509 storage2ast_sockaddr(&ourip, &sock);
2510
2511 if ((rtp->instance = ast_rtp_instance_new("asterisk", sched, &sock, NULL))) {
2512 struct ast_sockaddr instance_addr = { {0,} };
2513 ast_rtp_instance_get_local_address(rtp->instance, &instance_addr);
2514 sccp_log(DEBUGCAT_RTP) (VERBOSE_PREFIX_3 "%s: rtp server instance created at %s\n", c->designator, ast_sockaddr_stringify(&instance_addr));
2515 } else {
2516 return FALSE;
2517 }
2518
2519 /* rest below should be moved out of here (refactoring required) */
2520 PBX_RTP_TYPE *instance = rtp->instance;
2521 char *rtp_map_filter = NULL;
2522 //enum ast_media_type format_type = AST_MEDIA_TYPE_AUDIO;
2523 int fd_offset = 0;
2524 switch(rtp->type) {
2525 case SCCP_RTP_AUDIO:
2526 tos = d->audio_tos;
2527 cos = d->audio_cos;
2528 rtp_map_filter = "audio";
2529 //format_type = AST_MEDIA_TYPE_AUDIO;
2530 break;
2531
2532 #if CS_SCCP_VIDEO
2533 case SCCP_RTP_VIDEO:
2534 tos = d->video_tos;
2535 cos = d->video_cos;
2536 rtp_map_filter = "video";
2537 //format_type = AST_MEDIA_TYPE_VIDEO;
2538 fd_offset = 2;
2539 break;
2540 #endif
2541 default:
2542 pbx_log(LOG_ERROR, "%s: (wrapper_create_rtp) unknown/unhandled rtp type, returning instance for now\n", c->designator);
2543 return TRUE;
2544 }
2545
2546 if (c->owner) {
2547 ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(c->owner));
2548 ast_channel_set_fd(c->owner, fd_offset, ast_rtp_instance_fd(instance, 0)); // RTP
2549 ast_channel_set_fd(c->owner, fd_offset + 1, ast_rtp_instance_fd(instance, 1)); // RTCP
2550 }
2551 ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, 1);
2552 ast_rtp_instance_set_qos(instance, tos, cos, "SCCP RTP");
2553
2554 if (rtp->type == SCCP_RTP_AUDIO) {
2555 ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_DTMF, 1);
2556 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_2 "%s: update rtpmap: format:%s, payload:%d, mime:%s, rate:%d\n",
2557 c->designator, "telephone-event", c->dtmfmode == SCCP_DTMFMODE_SKINNY ? 121 : 101, "audio", 0);
2558 if (c->dtmfmode == SCCP_DTMFMODE_SKINNY) {
2559 /* untested */
2560 //ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(instance), instance, 121);
2561 //if (ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(instance), instance, 121, rtp_map_filter, "cisco-telephone-event", 0)) {
2562 // ast_rtp_codecs_payloads_unset(ast_rtp_instance_get_codecs(instance), instance, 121);
2563 //}
2564 /* \untested */
2565 ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE, 1);
2566 ast_rtp_instance_dtmf_mode_set(instance, AST_RTP_DTMF_MODE_INBAND);
2567 } else {
2568 ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(instance), instance, 101);
2569 if (ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(instance), instance, 101, rtp_map_filter, "telephone-event", (enum ast_rtp_options)0)) {
2570 ast_rtp_codecs_payloads_unset(ast_rtp_instance_get_codecs(instance), instance, 101);
2571 }
2572 ast_rtp_instance_dtmf_mode_set(instance, AST_RTP_DTMF_MODE_RFC2833);
2573 }
2574 sccp_log(DEBUGCAT_CODEC)(VERBOSE_PREFIX_2 "%s: update rtpmap: format:%s, payload:%d, mime:%s, rate:%d\n",
2575 c->designator, "slin16", 25, "audio", 16000);
2576 ast_rtp_codecs_payload_replace_format(ast_rtp_instance_get_codecs(instance), 25, ast_format_slin16); // replace slin16 RTPPayloadType=25 (wideband-256)
2577 }
2578 ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(instance), ast_format_cap_get_framing(ast_channel_nativeformats(c->owner)));
2579
2580 ast_rtp_instance_activate(instance);
2581
2582 if (c->owner) {
2583 ast_queue_frame(c->owner, &ast_null_frame);
2584 }
2585
2586 return TRUE;
2587 }
2588
sccp_wrapper_get_codec_framing(constChannelPtr c)2589 static uint sccp_wrapper_get_codec_framing(constChannelPtr c)
2590 {
2591 return ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(c->rtp.audio.instance));
2592 }
2593
sccp_wrapper_get_dtmf_payload_code(constChannelPtr c)2594 static uint sccp_wrapper_get_dtmf_payload_code(constChannelPtr c)
2595 {
2596 int rtp_code = 0;
2597 if (SCCP_DTMFMODE_SKINNY == c->dtmfmode) {
2598 /* untested*/
2599 //rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(c->rtp.audio.instance), 0, NULL, AST_RTP_CISCO_DTMF); // 121
2600 /* \untested */
2601 } else {
2602 rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(c->rtp.audio.instance), 0, NULL, AST_RTP_DTMF); // 101
2603 }
2604 sccp_log(DEBUGCAT_RTP)(VERBOSE_PREFIX_3 "%s: Using dtmf rtp_code : %d\n", c->designator, rtp_code);
2605 return rtp_code != -1 ? rtp_code : 0;
2606 }
2607
sccp_astwrap_destroyRTP(PBX_RTP_TYPE * rtp)2608 static boolean_t sccp_astwrap_destroyRTP(PBX_RTP_TYPE * rtp)
2609 {
2610 int res;
2611
2612 res = ast_rtp_instance_destroy(rtp);
2613 return (!res) ? TRUE : FALSE;
2614 }
2615
sccp_astwrap_checkHangup(constChannelPtr channel)2616 static boolean_t sccp_astwrap_checkHangup(constChannelPtr channel)
2617 {
2618 int res;
2619
2620 res = ast_check_hangup(channel->owner);
2621 return (!res) ? TRUE : FALSE;
2622 }
2623
sccp_astwrap_rtpGetPeer(PBX_RTP_TYPE * rtp,struct sockaddr_storage * address)2624 static boolean_t sccp_astwrap_rtpGetPeer(PBX_RTP_TYPE * rtp, struct sockaddr_storage *address)
2625 {
2626 struct ast_sockaddr addr;
2627
2628 ast_rtp_instance_get_remote_address(rtp, &addr);
2629 memcpy(address, &addr.ss, sizeof(struct sockaddr_storage));
2630 return TRUE;
2631 }
2632
sccp_astwrap_rtpGetUs(PBX_RTP_TYPE * rtp,struct sockaddr_storage * address)2633 static boolean_t sccp_astwrap_rtpGetUs(PBX_RTP_TYPE * rtp, struct sockaddr_storage *address)
2634 {
2635 struct ast_sockaddr addr;
2636
2637 ast_rtp_instance_get_local_address(rtp, &addr);
2638 memcpy(address, &addr.ss, sizeof(struct sockaddr_storage));
2639 return TRUE;
2640 }
2641
sccp_astwrap_getChannelByName(const char * name,PBX_CHANNEL_TYPE ** pbx_channel)2642 static boolean_t sccp_astwrap_getChannelByName(const char *name, PBX_CHANNEL_TYPE ** pbx_channel)
2643 {
2644 PBX_CHANNEL_TYPE *ast = ast_channel_get_by_name(name);
2645
2646 if (!ast) {
2647 return FALSE;
2648 }
2649 pbx_channel_lock(ast);
2650 *pbx_channel = pbx_channel_ref(ast);
2651 pbx_channel_unlock(ast);
2652 return TRUE;
2653 }
2654
2655 //static int sccp_astwrap_rtp_set_peer(const struct sccp_rtp *rtp, const struct sockaddr_storage *new_peer, int nat_active)
sccp_astwrap_setPhoneRTPAddress(const struct sccp_rtp * rtp,const struct sockaddr_storage * new_peer,int nat_active)2656 static int sccp_astwrap_setPhoneRTPAddress(const struct sccp_rtp *rtp, const struct sockaddr_storage *new_peer, int nat_active)
2657 {
2658 struct ast_sockaddr ast_sockaddr_dest;
2659 int res;
2660
2661 memcpy(&ast_sockaddr_dest.ss, new_peer, sizeof(struct sockaddr_storage));
2662 ast_sockaddr_dest.len = (ast_sockaddr_dest.ss.ss_family == AF_INET6) ? sizeof(struct sockaddr_in6) : sizeof(struct sockaddr_in);
2663 res = ast_rtp_instance_set_remote_address(rtp->instance, &ast_sockaddr_dest);
2664
2665 sccp_log((DEBUGCAT_RTP | DEBUGCAT_HIGH)) (VERBOSE_PREFIX_3 "SCCP: (setPhoneRTPAddress) Update PBX to send RTP/UDP media to '%s' (new remote) (NAT: %s)\n", ast_sockaddr_stringify(&ast_sockaddr_dest), S_COR(nat_active, "yes", "no"));
2666 if (nat_active) {
2667 ast_rtp_instance_set_prop(rtp->instance, AST_RTP_PROPERTY_NAT, 1);
2668 } else {
2669 ast_rtp_instance_set_prop(rtp->instance, AST_RTP_PROPERTY_NAT, 0);
2670 }
2671 return res;
2672 }
2673
2674
sccp_astwrap_setCalleridName(PBX_CHANNEL_TYPE * pbx_channel,const char * name)2675 static void sccp_astwrap_setCalleridName(PBX_CHANNEL_TYPE *pbx_channel, const char *name)
2676 {
2677 if (pbx_channel && name) {
2678 ast_party_name_free(&ast_channel_caller(pbx_channel)->id.name);
2679 ast_channel_caller(pbx_channel)->id.name.str = pbx_strdup(name);
2680 ast_channel_caller(pbx_channel)->id.name.valid = 1;
2681 }
2682 }
2683
sccp_astwrap_setCalleridNumber(PBX_CHANNEL_TYPE * pbx_channel,const char * number)2684 static void sccp_astwrap_setCalleridNumber(PBX_CHANNEL_TYPE *pbx_channel, const char *number)
2685 {
2686 if (pbx_channel && number) {
2687 ast_party_number_free(&ast_channel_caller(pbx_channel)->id.number);
2688 ast_channel_caller(pbx_channel)->id.number.str = pbx_strdup(number);
2689 ast_channel_caller(pbx_channel)->id.number.valid = 1;
2690 }
2691 }
2692
sccp_astwrap_setCalleridAni(PBX_CHANNEL_TYPE * pbx_channel,const char * number)2693 static void sccp_astwrap_setCalleridAni(PBX_CHANNEL_TYPE *pbx_channel, const char *number)
2694 {
2695 if (pbx_channel && number) {
2696 ast_party_number_free(&ast_channel_caller(pbx_channel)->ani.number);
2697 ast_channel_caller(pbx_channel)->ani.number.str = pbx_strdup(number);
2698 ast_channel_caller(pbx_channel)->ani.number.valid = 1;
2699 }
2700 }
2701
sccp_astwrap_setRedirectingParty(PBX_CHANNEL_TYPE * pbx_channel,const char * number,const char * name)2702 static void sccp_astwrap_setRedirectingParty(PBX_CHANNEL_TYPE *pbx_channel, const char *number, const char *name)
2703 {
2704 if (pbx_channel) {
2705 if (number) {
2706 ast_party_number_free(&ast_channel_redirecting(pbx_channel)->from.number);
2707 ast_channel_redirecting(pbx_channel)->from.number.str = pbx_strdup(number);
2708 ast_channel_redirecting(pbx_channel)->from.number.valid = 1;
2709 }
2710
2711 if (name) {
2712 ast_party_name_free(&ast_channel_redirecting(pbx_channel)->from.name);
2713 ast_channel_redirecting(pbx_channel)->from.name.str = pbx_strdup(name);
2714 ast_channel_redirecting(pbx_channel)->from.name.valid = 1;
2715 }
2716 }
2717 }
2718
sccp_astwrap_setRedirectedParty(PBX_CHANNEL_TYPE * pbx_channel,const char * number,const char * name)2719 static void sccp_astwrap_setRedirectedParty(PBX_CHANNEL_TYPE *pbx_channel, const char *number, const char *name)
2720 {
2721 if (pbx_channel) {
2722 if (number) {
2723 ast_party_number_free(&ast_channel_redirecting(pbx_channel)->to.number);
2724 ast_channel_redirecting(pbx_channel)->to.number.str = pbx_strdup(number);
2725 ast_channel_redirecting(pbx_channel)->to.number.valid = 1;
2726 }
2727
2728 if (name) {
2729 ast_party_name_free(&ast_channel_redirecting(pbx_channel)->to.name);
2730 ast_channel_redirecting(pbx_channel)->to.name.str = pbx_strdup(name);
2731 ast_channel_redirecting(pbx_channel)->to.name.valid = 1;
2732 }
2733 }
2734 }
2735
sccp_astwrap_updateConnectedLine(constChannelPtr channel,const char * number,const char * name,uint8_t reason)2736 static void sccp_astwrap_updateConnectedLine(constChannelPtr channel, const char *number, const char *name, uint8_t reason)
2737 {
2738 if (!channel || !channel->owner) {
2739 return;
2740 }
2741 __sccp_astwrap_updateConnectedLine(channel->owner, number,name, reason);
2742 }
2743
sccp_astwrap_sched_add(int when,sccp_sched_cb callback,const void * data)2744 static int sccp_astwrap_sched_add(int when, sccp_sched_cb callback, const void *data)
2745 {
2746 if (sched) {
2747 return ast_sched_add(sched, when, callback, data);
2748 }
2749 return -1;
2750 }
2751
sccp_astwrap_sched_del(int id)2752 static int sccp_astwrap_sched_del(int id)
2753 {
2754 if (sched) {
2755 return AST_SCHED_DEL(sched, id);
2756 }
2757 return -1;
2758 }
2759
sccp_astwrap_sched_add_ref(int * id,int when,sccp_sched_cb callback,sccp_channel_t * channel)2760 static int sccp_astwrap_sched_add_ref(int *id, int when, sccp_sched_cb callback, sccp_channel_t * channel)
2761 {
2762 if (sched && channel) {
2763 sccp_channel_t *c = sccp_channel_retain(channel);
2764
2765 if (c) {
2766 if ((*id = ast_sched_add(sched, when, callback, c)) < 0) {
2767 sccp_log(DEBUGCAT_CORE)(VERBOSE_PREFIX_3 "%s: sched add id:%d, when:%d, failed\n", c->designator, *id, when);
2768 sccp_channel_release(&channel); /* explicit release during failure */
2769 }
2770 return *id;
2771 }
2772 }
2773 return -2;
2774 }
2775
sccp_astwrap_sched_del_ref(int * id,sccp_channel_t * channel)2776 static int sccp_astwrap_sched_del_ref(int *id, sccp_channel_t *channel)
2777 {
2778 if (sched) {
2779 AST_SCHED_DEL_UNREF(sched, *id, sccp_channel_release(&channel));
2780 return *id;
2781 }
2782 return -2;
2783 }
2784
sccp_astwrap_sched_replace_ref(int * id,int when,ast_sched_cb callback,sccp_channel_t * channel)2785 static int sccp_astwrap_sched_replace_ref(int *id, int when, ast_sched_cb callback, sccp_channel_t *channel)
2786 {
2787 if (sched) {
2788 //sccp_log(DEBUGCAT_CORE)(VERBOSE_PREFIX_3 "%s: (sched_replace_ref) replacing id: %d\n", channel->designator, *id);
2789 AST_SCHED_REPLACE_UNREF(*id, sched, when, callback, channel, sccp_channel_release((sccp_channel_t **)&_data), sccp_channel_release(&channel), sccp_channel_retain(channel)); /* explicit retain/release */
2790 //sccp_log(DEBUGCAT_CORE)(VERBOSE_PREFIX_3 "%s: (sched_replace_ref) returning id: %d\n", channel->designator, *id);
2791 return *id;
2792 }
2793 return -2;
2794 }
2795
sccp_astwrap_sched_when(int id)2796 static long sccp_astwrap_sched_when(int id)
2797 {
2798 if (sched) {
2799 return ast_sched_when(sched, id);
2800 }
2801 return FALSE;
2802 }
2803
sccp_astwrap_sched_wait(int id)2804 static int sccp_astwrap_sched_wait(int id)
2805 {
2806 if (sched) {
2807 return ast_sched_wait(sched);
2808 }
2809 return FALSE;
2810 }
2811
sccp_astwrap_setCallState(constChannelPtr channel,enum ast_channel_state state)2812 static int sccp_astwrap_setCallState(constChannelPtr channel, enum ast_channel_state state)
2813 {
2814 if (channel && channel->owner) {
2815 pbx_setstate(channel->owner, state);
2816 }
2817 return 0;
2818 }
2819
sccp_astwrap_getRemoteChannel(constChannelPtr channel,PBX_CHANNEL_TYPE ** pbx_channel)2820 static boolean_t sccp_astwrap_getRemoteChannel(constChannelPtr channel, PBX_CHANNEL_TYPE ** pbx_channel)
2821 {
2822
2823 PBX_CHANNEL_TYPE *remotePeer = NULL;
2824
2825 /*
2826 struct ast_channel_iterator *iterator = ast_channel_iterator_all_new();
2827 ((struct ao2_iterator *)iterator)->flags |= AO2_ITERATOR_DONTLOCK;
2828 for (; (remotePeer = ast_channel_iterator_next(iterator)); pbx_channel_unref(remotePeer)) {
2829 if (pbx_find_channel_by_linkid(remotePeer, (void *)ast_channel_linkedid(channel->owner))) {
2830 break;
2831 }
2832 }
2833 while(!(remotePeer = ast_channel_iterator_next(iterator) ){
2834 pbx_channel_unref(remotePeer);
2835 }
2836 ast_channel_iterator_destroy(iterator);
2837 if (remotePeer) {
2838 *pbx_channel = remotePeer;
2839 remotePeer = pbx_channel_unref(remotePeer); // should we be releasing th referenec here, it has not been taken explicitly.
2840 return TRUE;
2841 }
2842 */
2843 *pbx_channel = remotePeer;
2844 return FALSE;
2845 }
2846
2847 /*!
2848 * \brief Send Text to Asterisk Channel
2849 * \param ast Asterisk Channel as ast_channel
2850 * \param text Text to be send as char
2851 * \return Succes as int
2852 *
2853 * \called_from_asterisk
2854 */
sccp_pbx_sendtext(PBX_CHANNEL_TYPE * ast,const char * text)2855 static int sccp_pbx_sendtext(PBX_CHANNEL_TYPE * ast, const char *text)
2856 {
2857 uint8_t instance;
2858
2859 if (!ast) {
2860 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: No PBX CHANNEL to send text to\n");
2861 return -1;
2862 }
2863
2864 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast));
2865 if (!c) {
2866 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: No SCCP CHANNEL to send text to (%s)\n", pbx_channel_name(ast));
2867 return -1;
2868 }
2869
2870 AUTO_RELEASE(sccp_device_t, d , sccp_channel_getDevice(c));
2871 if (!d) {
2872 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: No SCCP DEVICE to send text to (%s)\n", pbx_channel_name(ast));
2873 return -1;
2874 }
2875
2876 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "%s: Sending text %s on %s\n", d->id, text, pbx_channel_name(ast));
2877
2878 instance = sccp_device_find_index_for_line(d, c->line->name);
2879 sccp_dev_displayprompt(d, instance, c->callid, (char *) text, 10);
2880 return 0;
2881 }
2882
2883 /*!
2884 * \brief Receive First Digit from Asterisk Channel
2885 * \param ast Asterisk Channel as ast_channel
2886 * \param digit First Digit as char
2887 * \return Always Return -1 as int
2888 *
2889 * \called_from_asterisk
2890 */
sccp_wrapper_recvdigit_begin(PBX_CHANNEL_TYPE * ast,char digit)2891 static int sccp_wrapper_recvdigit_begin(PBX_CHANNEL_TYPE * ast, char digit)
2892 {
2893 return -1;
2894 }
2895
2896 /*!
2897 * \brief Receive Last Digit from Asterisk Channel
2898 * \param ast Asterisk Channel as ast_channel
2899 * \param digit Last Digit as char
2900 * \param duration Duration as int
2901 * \return boolean
2902 *
2903 * \called_from_asterisk
2904 */
sccp_wrapper_recvdigit_end(PBX_CHANNEL_TYPE * ast,char digit,unsigned int duration)2905 static int sccp_wrapper_recvdigit_end(PBX_CHANNEL_TYPE * ast, char digit, unsigned int duration)
2906 {
2907 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast));
2908 if (c) {
2909 if (c->dtmfmode == SCCP_DTMFMODE_RFC2833) {
2910 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "%s: Channel(%s) DTMF Mode is RFC2833. Skipping...\n", c->designator, pbx_channel_name(ast));
2911 return -1;
2912 }
2913
2914 AUTO_RELEASE(sccp_device_t, d , sccp_channel_getDevice(c));
2915 if (!d) {
2916 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: No SCCP DEVICE to send digit to (%s)\n", pbx_channel_name(ast));
2917 return -1;
2918 }
2919
2920 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "%s: Asterisk asked to send dtmf '%d' to channel %s. Trying to send it %s\n", d->id, digit, pbx_channel_name(ast), sccp_dtmfmode2str(d->dtmfmode));
2921 if (c->state != SCCP_CHANNELSTATE_CONNECTED) {
2922 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "%s: Can't send the dtmf '%d' %c to a not connected channel %s\n", d->id, digit, digit, pbx_channel_name(ast));
2923 return -1;
2924 }
2925 sccp_dev_keypadbutton(d, digit, sccp_device_find_index_for_line(d, c->line->name), c->callid);
2926 } else {
2927 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_3 "SCCP: No SCCP CHANNEL to send digit to (%s)\n", pbx_channel_name(ast));
2928 return -1;
2929 }
2930
2931 return -1;
2932 }
2933
sccp_astwrap_findChannelWithCallback(int (* const found_cb)(PBX_CHANNEL_TYPE * c,void * data),void * data,boolean_t lock)2934 static PBX_CHANNEL_TYPE *sccp_astwrap_findChannelWithCallback(int (*const found_cb) (PBX_CHANNEL_TYPE * c, void *data), void *data, boolean_t lock)
2935 {
2936 PBX_CHANNEL_TYPE *remotePeer = NULL;
2937
2938 /*
2939 struct ast_channel_iterator *iterator = ast_channel_iterator_all_new();
2940 if (!lock) {
2941 ((struct ao2_iterator *)iterator)->flags |= AO2_ITERATOR_DONTLOCK;
2942 }
2943 for (; (remotePeer = ast_channel_iterator_next(iterator)); remotePeer = pbx_channel_unref(remotePeer)) {
2944 if (found_cb(remotePeer, data)) {
2945 // ast_channel_lock(remotePeer);
2946 pbx_channel_unref(remotePeer);
2947 break;
2948 }
2949 }
2950 ast_channel_iterator_destroy(iterator);
2951 */
2952
2953 return remotePeer;
2954 }
2955
2956 /*! \brief Set an option on a asterisk channel */
2957 #if 0
2958 static int sccp_astwrap_setOption(PBX_CHANNEL_TYPE * ast, int option, void *data, int datalen)
2959 {
2960 int res = -1;
2961 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast));
2962
2963 if (c) {
2964 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_2 "%s: channel: %s(%s) setOption: %d\n", c->currentDeviceId, sccp_channel_toString(c), pbx_channel_name(ast), option);
2965 //! if AST_OPTION_FORMAT_READ / AST_OPTION_FORMAT_WRITE are available we might be indication that we can do transcoding (channel.c:set_format). Correct ? */
2966 switch (option) {
2967 case AST_OPTION_FORMAT_READ:
2968 if (c->rtp.audio.instance) {
2969 res = ast_rtp_instance_set_read_format(c->rtp.audio.instance, (struct ast_format *) data);
2970 }
2971 //sccp_astwrap_setReadFormat(c, (struct ast_format *) data);
2972 break;
2973 case AST_OPTION_FORMAT_WRITE:
2974 if (c->rtp.audio.instance) {
2975 res = ast_rtp_instance_set_write_format(c->rtp.audio.instance, (struct ast_format *) data);
2976 }
2977 //sccp_astwrap_setWriteFormat(c, (struct ast_format *) data);
2978 break;
2979
2980 case AST_OPTION_MAKE_COMPATIBLE:
2981 if (c->rtp.audio.instance) {
2982 res = ast_rtp_instance_make_compatible(ast, c->rtp.audio.instance, (PBX_CHANNEL_TYPE *) data);
2983 }
2984 break;
2985 case AST_OPTION_DIGIT_DETECT:
2986 case AST_OPTION_SECURE_SIGNALING:
2987 case AST_OPTION_SECURE_MEDIA:
2988 res = -1;
2989 break;
2990 default:
2991 break;
2992 }
2993 }
2994 return res;
2995 }
2996 #endif
2997
sccp_astwrap_set_pbxchannel_linkedid(PBX_CHANNEL_TYPE * pbx_channel,const char * new_linkedid)2998 static void sccp_astwrap_set_pbxchannel_linkedid(PBX_CHANNEL_TYPE * pbx_channel, const char *new_linkedid)
2999 {
3000 if (pbx_channel) {
3001 if (!strcmp(ast_channel_linkedid(pbx_channel), new_linkedid)) {
3002 return;
3003 }
3004 // ast_cel_check_retire_linkedid(pbx_channel);
3005 // ast_channel_linkedid_set(pbx_channel, new_linkedid);
3006 // ast_cel_linkedid_ref(new_linkedid);
3007 }
3008 };
3009
3010 #define DECLARE_PBX_CHANNEL_STRGET(_field) \
3011 static const char *sccp_astwrap_get_channel_##_field(constChannelPtr channel) \
3012 { \
3013 static const char *empty_channel_##_field = "--no-channel" #_field "--"; \
3014 if (channel && channel->owner) { \
3015 return ast_channel_##_field(channel->owner); \
3016 } \
3017 return empty_channel_##_field; \
3018 };
3019
3020 #define DECLARE_PBX_CHANNEL_STRSET(_field) \
3021 static void sccp_astwrap_set_channel_##_field(constChannelPtr channel, const char * (_field)) \
3022 { \
3023 if (channel && channel->owner) { \
3024 ast_channel_##_field##_set(channel->owner, _field); \
3025 } \
3026 };
3027
3028 DECLARE_PBX_CHANNEL_STRGET(name)
DECLARE_PBX_CHANNEL_STRSET(name)3029 DECLARE_PBX_CHANNEL_STRSET(name)
3030 DECLARE_PBX_CHANNEL_STRGET(uniqueid)
3031 DECLARE_PBX_CHANNEL_STRGET(appl)
3032 DECLARE_PBX_CHANNEL_STRGET(exten)
3033 DECLARE_PBX_CHANNEL_STRSET(exten)
3034 DECLARE_PBX_CHANNEL_STRGET(linkedid)
3035 DECLARE_PBX_CHANNEL_STRGET(context)
3036 DECLARE_PBX_CHANNEL_STRSET(context)
3037 DECLARE_PBX_CHANNEL_STRGET(macroexten)
3038 DECLARE_PBX_CHANNEL_STRSET(macroexten)
3039 DECLARE_PBX_CHANNEL_STRGET(macrocontext)
3040 DECLARE_PBX_CHANNEL_STRSET(macrocontext)
3041 DECLARE_PBX_CHANNEL_STRGET(call_forward)
3042 DECLARE_PBX_CHANNEL_STRSET(call_forward)
3043
3044 static void sccp_astwrap_set_channel_linkedid(constChannelPtr channel, const char *new_linkedid)
3045 {
3046 if (channel && channel->owner) {
3047 sccp_astwrap_set_pbxchannel_linkedid(channel->owner, new_linkedid);
3048 }
3049 };
3050
sccp_astwrap_get_channel_state(constChannelPtr channel)3051 static enum ast_channel_state sccp_astwrap_get_channel_state(constChannelPtr channel)
3052 {
3053 if (channel && channel->owner) {
3054 return ast_channel_state(channel->owner);
3055 }
3056 return (enum ast_channel_state) 0;
3057 }
3058
sccp_astwrap_get_channel_pbx(constChannelPtr channel)3059 static const struct ast_pbx *sccp_astwrap_get_channel_pbx(constChannelPtr channel)
3060 {
3061 if (channel && channel->owner) {
3062 return ast_channel_pbx(channel->owner);
3063 }
3064 return NULL;
3065 }
3066
sccp_pbx_sendHTML(PBX_CHANNEL_TYPE * ast,int subclass,const char * data,int datalen)3067 static int sccp_pbx_sendHTML(PBX_CHANNEL_TYPE * ast, int subclass, const char *data, int datalen)
3068 {
3069 int res = -1;
3070 if (!datalen || sccp_strlen_zero(data) || !(!strncmp(data, "http://", 7) || !strncmp(data, "file://", 7) || !strncmp(data, "ftp://", 6))) {
3071 pbx_log(LOG_NOTICE, "SCCP: Received a non valid URL\n");
3072 return res;
3073 }
3074 struct ast_frame fr;
3075
3076 AUTO_RELEASE(sccp_channel_t, c , get_sccp_channel_from_pbx_channel(ast));
3077 if (c) {
3078 AUTO_RELEASE(sccp_device_t, d , c->getDevice(c));
3079 if (d) {
3080 memset(&fr, 0, sizeof(fr));
3081 fr.frametype = AST_FRAME_HTML;
3082 fr.data.ptr = (char *) data;
3083 fr.src = "SCCP Send URL";
3084 fr.datalen = datalen;
3085
3086 sccp_push_result_t pushResult = d->pushURL(d, data, 1, SKINNY_TONE_ZIP);
3087
3088 if (SCCP_PUSH_RESULT_SUCCESS == pushResult) {
3089 fr.subclass.integer = AST_HTML_LDCOMPLETE;
3090 } else {
3091 fr.subclass.integer = AST_HTML_NOSUPPORT;
3092 }
3093 ast_queue_frame(ast, ast_frisolate(&fr));
3094 res = 0;
3095 }
3096 }
3097 return res;
3098 }
3099
3100 /*!
3101 * \brief Queue a control frame
3102 * \param pbx_channel PBX Channel
3103 * \param control as Asterisk Control Frame Type
3104 */
sccp_astwrap_queue_control(const PBX_CHANNEL_TYPE * pbx_channel,enum ast_control_frame_type control)3105 int sccp_astwrap_queue_control(const PBX_CHANNEL_TYPE * pbx_channel, enum ast_control_frame_type control)
3106 {
3107 struct ast_frame f = { AST_FRAME_CONTROL,.subclass.integer = control };
3108 return ast_queue_frame((PBX_CHANNEL_TYPE *) pbx_channel, &f);
3109 }
3110
3111 /*!
3112 * \brief Queue a control frame with payload
3113 * \param pbx_channel PBX Channel
3114 * \param control as Asterisk Control Frame Type
3115 * \param data Payload
3116 * \param datalen Payload Length
3117 */
sccp_astwrap_queue_control_data(const PBX_CHANNEL_TYPE * pbx_channel,enum ast_control_frame_type control,const void * data,size_t datalen)3118 int sccp_astwrap_queue_control_data(const PBX_CHANNEL_TYPE * pbx_channel, enum ast_control_frame_type control, const void *data, size_t datalen)
3119 {
3120 struct ast_frame f = { AST_FRAME_CONTROL,.subclass.integer = control,.data.ptr = (void *) data,.datalen = datalen };
3121 return ast_queue_frame((PBX_CHANNEL_TYPE *) pbx_channel, &f);
3122 }
3123
3124 /*!
3125 * \brief Get Hint Extension State and return the matching Busy Lamp Field State
3126 */
sccp_astwrap_getExtensionState(const char * extension,const char * context)3127 static skinny_busylampfield_state_t sccp_astwrap_getExtensionState(const char *extension, const char *context)
3128 {
3129 skinny_busylampfield_state_t result = SKINNY_BLF_STATUS_UNKNOWN;
3130
3131 if (sccp_strlen_zero(extension) || sccp_strlen_zero(context)) {
3132 pbx_log(LOG_ERROR, "SCCP: iPbx.getExtensionState: Either extension:'%s' or context:;%s' provided is empty\n", extension, context);
3133 return result;
3134 }
3135
3136 int state = ast_extension_state(NULL, context, extension);
3137
3138 switch (state) {
3139 case AST_EXTENSION_REMOVED:
3140 case AST_EXTENSION_DEACTIVATED:
3141 case AST_EXTENSION_UNAVAILABLE:
3142 result = SKINNY_BLF_STATUS_UNKNOWN;
3143 break;
3144 case AST_EXTENSION_NOT_INUSE:
3145 result = SKINNY_BLF_STATUS_IDLE;
3146 break;
3147 case AST_EXTENSION_INUSE:
3148 case AST_EXTENSION_ONHOLD:
3149 case AST_EXTENSION_ONHOLD + AST_EXTENSION_INUSE:
3150 result = SKINNY_BLF_STATUS_INUSE;
3151 break;
3152 case AST_EXTENSION_BUSY:
3153 result = SKINNY_BLF_STATUS_DND;
3154 break;
3155 case AST_EXTENSION_RINGING + AST_EXTENSION_INUSE:
3156 case AST_EXTENSION_RINGING:
3157 result = SKINNY_BLF_STATUS_ALERTING;
3158 break;
3159 }
3160 sccp_log(DEBUGCAT_HINT) (VERBOSE_PREFIX_4 "SCCP: (getExtensionState) extension: %s@%s, extension_state: '%s (%d)' -> blf state '%d'\n", extension, context, ast_extension_state2str(state), state, result);
3161 return result;
3162 }
3163
sccp_astwrap_endpoint_create(const char * tech,const char * resource)3164 static struct ast_endpoint *sccp_astwrap_endpoint_create(const char *tech, const char *resource)
3165 {
3166 return ast_endpoint_create(tech, resource);
3167 }
3168
sccp_astwrap_endpoint_online(struct ast_endpoint * endpoint,const char * address)3169 static void sccp_astwrap_endpoint_online(struct ast_endpoint *endpoint, const char *address)
3170 {
3171 RAII(struct ast_json *, blob, NULL, ast_json_unref);
3172 ast_endpoint_set_state(endpoint, AST_ENDPOINT_ONLINE);
3173 blob = ast_json_pack("{s: s, s: s}", "peer_status", "Registered", "address", address);
3174 ast_endpoint_blob_publish(endpoint, ast_endpoint_state_type(), blob);
3175 }
3176
sccp_astwrap_endpoint_offline(struct ast_endpoint * endpoint,const char * cause)3177 static void sccp_astwrap_endpoint_offline(struct ast_endpoint *endpoint, const char *cause)
3178 {
3179 RAII(struct ast_json *, blob, NULL, ast_json_unref);
3180 ast_endpoint_set_state(endpoint, AST_ENDPOINT_OFFLINE);
3181 blob = ast_json_pack("{s: s, s: s}", "peer_status", "Unregistered", "cause", cause);
3182 ast_endpoint_blob_publish(endpoint, ast_endpoint_state_type(), blob);
3183 }
3184
sccp_astwrap_endpoint_shutdown(struct ast_endpoint ** endpoint)3185 static void sccp_astwrap_endpoint_shutdown(struct ast_endpoint **endpoint)
3186 {
3187 ast_endpoint_shutdown(*endpoint);
3188 *endpoint = NULL;
3189 }
3190
sccp_astwrap_dumpchan(PBX_CHANNEL_TYPE * const c,char * const buf,size_t size)3191 static int sccp_astwrap_dumpchan(PBX_CHANNEL_TYPE * const c, char * const buf, size_t size)
3192 {
3193 long elapsed_seconds = 0;
3194 int hour = 0, min = 0, sec = 0;
3195 char cgrp[256];
3196 char pgrp[256];
3197 pbx_str_t *write_transpath = pbx_str_alloca(256);
3198 pbx_str_t *read_transpath = pbx_str_alloca(256);
3199 struct ast_bridge *bridge;
3200 struct ast_str *vars = ast_str_thread_get(&ast_str_thread_global_buf, 16);
3201 pbx_str_t *codec_buf = pbx_str_alloca(64);
3202
3203 memset(buf, 0, size);
3204 if (!c) {
3205 return -1;
3206 }
3207
3208 elapsed_seconds = ast_channel_get_duration(c);
3209 hour = elapsed_seconds / 3600;
3210 min = (elapsed_seconds % 3600) / 60;
3211 sec = elapsed_seconds % 60;
3212
3213 ast_channel_lock(c);
3214 bridge = ast_channel_get_bridge(c);
3215 ast_channel_unlock(c);
3216
3217 pbx_builtin_serialize_variables(c, &vars);
3218
3219 snprintf(buf, size, "Name= %s\n"
3220 "Type= %s\n"
3221 "UniqueID= %s\n"
3222 "LinkedID= %s\n"
3223 "CallerIDNum= %s\n"
3224 "CallerIDName= %s\n"
3225 "ConnectedLineIDNum= %s\n"
3226 "ConnectedLineIDName=%s\n"
3227 "DNIDDigits= %s\n"
3228 "RDNIS= %s\n"
3229 "Parkinglot= %s\n"
3230 "Language= %s\n"
3231 "State= %s (%d)\n"
3232 "Rings= %d\n"
3233 "NativeFormat= %s\n"
3234 "WriteFormat= %s\n"
3235 "ReadFormat= %s\n"
3236 "RawWriteFormat= %s\n"
3237 "RawReadFormat= %s\n"
3238 "WriteTranscode= %s %s\n"
3239 "ReadTranscode= %s %s\n"
3240 "1stFileDescriptor= %d\n"
3241 "Framesin= %d %s\n"
3242 "Framesout= %d %s\n"
3243 "TimetoHangup= %ld\n"
3244 "ElapsedTime= %dh%dm%ds\n"
3245 "BridgeID= %s\n"
3246 "Context= %s\n"
3247 "Extension= %s\n"
3248 "Priority= %d\n"
3249 "CallGroup= %s\n"
3250 "PickupGroup= %s\n"
3251 "Application= %s\n"
3252 "Data= %s\n"
3253 "Blocking_in= %s\n"
3254 "Variables= %s\n",
3255 ast_channel_name(c),
3256 ast_channel_tech(c)->type,
3257 ast_channel_uniqueid(c),
3258 ast_channel_linkedid(c),
3259 S_COR(ast_channel_caller(c)->id.number.valid, ast_channel_caller(c)->id.number.str, "(N/A)"),
3260 S_COR(ast_channel_caller(c)->id.name.valid, ast_channel_caller(c)->id.name.str, "(N/A)"),
3261 S_COR(ast_channel_connected(c)->id.number.valid, ast_channel_connected(c)->id.number.str, "(N/A)"),
3262 S_COR(ast_channel_connected(c)->id.name.valid, ast_channel_connected(c)->id.name.str, "(N/A)"),
3263 S_OR(ast_channel_dialed(c)->number.str, "(N/A)"),
3264 S_COR(ast_channel_redirecting(c)->from.number.valid, ast_channel_redirecting(c)->from.number.str, "(N/A)"),
3265 ast_channel_parkinglot(c),
3266 ast_channel_language(c),
3267 ast_state2str(ast_channel_state(c)),
3268 ast_channel_state(c),
3269 ast_channel_rings(c),
3270 ast_format_cap_get_names(ast_channel_nativeformats(c), &codec_buf), //ast_getformatname_multiple(nf, sizeof(nf), ast_channel_nativeformats(c)),
3271 ast_format_get_name(ast_channel_writeformat(c)),
3272 ast_format_get_name(ast_channel_readformat(c)),
3273 ast_format_get_name(ast_channel_rawwriteformat(c)),
3274 ast_format_get_name(ast_channel_rawreadformat(c)),
3275 ast_channel_writetrans(c) ? "Yes" : "No",
3276 ast_translate_path_to_str(ast_channel_writetrans(c), &write_transpath),
3277 ast_channel_readtrans(c) ? "Yes" : "No",
3278 ast_translate_path_to_str(ast_channel_readtrans(c), &read_transpath),
3279 ast_channel_fd(c, 0),
3280 ast_channel_fin(c) & ~DEBUGCHAN_FLAG, (ast_channel_fin(c) & DEBUGCHAN_FLAG) ? " (DEBUGGED)" : "",
3281 ast_channel_fout(c) & ~DEBUGCHAN_FLAG, (ast_channel_fout(c) & DEBUGCHAN_FLAG) ? " (DEBUGGED)" : "",
3282 (long) ast_channel_whentohangup(c)->tv_sec,
3283 hour,
3284 min,
3285 sec,
3286 bridge ? bridge->uniqueid : "(Not bridged)",
3287 ast_channel_context(c),
3288 ast_channel_exten(c),
3289 ast_channel_priority(c),
3290 ast_print_group(cgrp, sizeof(cgrp), ast_channel_callgroup(c)),
3291 ast_print_group(pgrp, sizeof(pgrp), ast_channel_pickupgroup(c)),
3292 ast_channel_appl(c) ? ast_channel_appl(c) : "(N/A)",
3293 ast_channel_data(c) ? S_OR(ast_channel_data(c), "(Empty)") : "(None)",
3294 (ast_test_flag(ast_channel_flags(c), AST_FLAG_BLOCKING) ? ast_channel_blockproc(c) : "(Not Blocking)"),
3295 ast_str_buffer(vars)
3296 );
3297
3298 ao2_cleanup(bridge);
3299 return 0;
3300 }
3301
sccp_astwrap_channelIsBridged(sccp_channel_t * channel)3302 static boolean_t sccp_astwrap_channelIsBridged(sccp_channel_t * channel)
3303 {
3304 boolean_t res = FALSE;
3305
3306 if (!channel || !channel->owner) {
3307 return res;
3308 }
3309 ast_channel_lock(channel->owner);
3310 res = ast_channel_is_bridged(channel->owner);
3311 ast_channel_unlock(channel->owner);
3312 return res;
3313 }
3314
sccp_astwrap_getBridgeChannel(PBX_CHANNEL_TYPE * pbx_channel)3315 static PBX_CHANNEL_TYPE *sccp_astwrap_getBridgeChannel(PBX_CHANNEL_TYPE * pbx_channel)
3316 {
3317 PBX_CHANNEL_TYPE *bridgePeer = NULL;
3318 if (pbx_channel) {
3319 bridgePeer = ast_channel_bridge_peer(pbx_channel); /* return pbx_channel_ref +1 */
3320 }
3321 return bridgePeer;
3322 }
3323
sccp_astwrap_attended_transfer(sccp_channel_t * destination_channel,sccp_channel_t * source_channel)3324 static boolean_t sccp_astwrap_attended_transfer(sccp_channel_t * destination_channel, sccp_channel_t * source_channel)
3325 {
3326 enum ast_transfer_result res;
3327 // possibly move transfer related callinfo updates here
3328 if (!destination_channel || !source_channel || !destination_channel->owner || !source_channel->owner) {
3329 return FALSE;
3330 }
3331 PBX_CHANNEL_TYPE *pbx_destination_local_channel = destination_channel->owner;
3332 PBX_CHANNEL_TYPE *pbx_source_local_channel = source_channel->owner;
3333
3334 res = ast_bridge_transfer_attended(pbx_destination_local_channel, pbx_source_local_channel);
3335 if (res != AST_BRIDGE_TRANSFER_SUCCESS) {
3336 pbx_log(LOG_ERROR, "%s: Failed to transfer %s to %s (%u)\n", source_channel->designator, source_channel->designator, destination_channel->designator, res);
3337 ast_queue_control(pbx_destination_local_channel, AST_CONTROL_HOLD);
3338 return FALSE;
3339 }
3340 return TRUE;
3341 }
3342
3343 /*!
3344 * \brief using RTP Glue Engine
3345 */
3346 #if defined(__cplusplus) || defined(c_plusplus)
3347 struct ast_rtp_glue sccp_rtp = {
3348 /* *INDENT-OFF* */
3349 type: SCCP_TECHTYPE_STR,
3350 mod: NULL,
3351 get_rtp_info:sccp_astwrap_get_rtp_info,
3352 //allow_rtp_remote:sccp_astwrap_allow_rtp_remote, /* check c->directmedia and return 1 if ok */
3353 get_vrtp_info:sccp_astwrap_get_vrtp_info,
3354 //allow_vrtp_remote:sccp_astwrap_allow_vrtp_remote, /* check c->directmedia and return 1 if ok */
3355 get_trtp_info:NULL,
3356 update_peer:sccp_astwrap_update_rtp_peer,
3357 get_codec:sccp_astwrap_getCodec,
3358 /* *INDENT-ON* */
3359 };
3360 #else
3361 struct ast_rtp_glue sccp_rtp = {
3362 .type = SCCP_TECHTYPE_STR,
3363 .get_rtp_info = sccp_astwrap_get_rtp_info,
3364 //.allow_rtp_remote = sccp_astwrap_allow_rtp_remote, /* check c->directmedia and return 1 if ok */
3365 .get_vrtp_info = sccp_astwrap_get_vrtp_info,
3366 //.allow_vrtp_remote = sccp_astwrap_allow_vrtp_remote, /* check c->directmedia and return 1 if ok */
3367 .update_peer = sccp_astwrap_update_rtp_peer,
3368 .get_codec = sccp_astwrap_getCodec,
3369 };
3370 #endif
3371
3372 #ifdef HAVE_PBX_MESSAGE_H
3373 #include <asterisk/message.h>
sccp_astwrap_message_send(const struct ast_msg * msg,const char * to,const char * from)3374 static int sccp_astwrap_message_send(const struct ast_msg *msg, const char *to, const char *from)
3375 {
3376 char *lineName;
3377 const char *messageText = ast_msg_get_body(msg);
3378 int res = -1;
3379
3380 lineName = pbx_strdupa(to);
3381 if (strchr(lineName, '@')) {
3382 strsep(&lineName, "@");
3383 } else {
3384 strsep(&lineName, ":");
3385 }
3386 if (sccp_strlen_zero(lineName)) {
3387 pbx_log(LOG_WARNING, "MESSAGE(to) is invalid for SCCP - '%s'\n", to);
3388 return -1;
3389 }
3390
3391 AUTO_RELEASE(sccp_line_t, line , sccp_line_find_byname(lineName, FALSE));
3392 if (!line) {
3393 pbx_log(LOG_WARNING, "line '%s' not found\n", lineName);
3394 return -1;
3395 }
3396
3397 /** \todo move this to line implementation */
3398 sccp_linedevice_t * ld = NULL;
3399 sccp_push_result_t pushResult;
3400
3401 SCCP_LIST_LOCK(&line->devices);
3402 SCCP_LIST_TRAVERSE(&line->devices, ld, list) {
3403 pushResult = ld->device->pushTextMessage(ld->device, messageText, from, 1, SKINNY_TONE_ZIP);
3404 if (SCCP_PUSH_RESULT_SUCCESS == pushResult) {
3405 res = 0;
3406 }
3407 }
3408 SCCP_LIST_UNLOCK(&line->devices);
3409
3410 return res;
3411 }
3412
3413 #if defined(__cplusplus) || defined(c_plusplus)
3414 static const struct ast_msg_tech sccp_msg_tech = {
3415 /* *INDENT-OFF* */
3416 name: "sccp",
3417 msg_send:sccp_astwrap_message_send,
3418 /* *INDENT-ON* */
3419 };
3420 #else
3421 static const struct ast_msg_tech sccp_msg_tech = {
3422 /* *INDENT-OFF* */
3423 .name = "sccp",
3424 .msg_send = sccp_astwrap_message_send,
3425 /* *INDENT-ON* */
3426 };
3427 #endif
3428
3429 #endif
3430
3431 /*!
3432 * \brief pbx_manager_register
3433 *
3434 * \note this functions needs to be defined here, because it depends on the static declaration of pbx_module_info->self
3435 */
pbx_manager_register(const char * action,int authority,int (* func)(struct mansession * s,const struct message * m),const char * synopsis,const char * description)3436 int pbx_manager_register(const char * action, int authority, int (*func)(struct mansession * s, const struct message * m), const char * synopsis, const char * description)
3437 {
3438 #if defined(__cplusplus) || defined(c_plusplus)
3439 return 0;
3440 #else
3441 return ast_manager_register2(action, authority, func, pbx_module_info->self, synopsis, description);
3442 #endif
3443 }
3444
sccp_wrapper_register_manager(const char * action,int authority,int (* func)(struct mansession * s,const struct message * m),const char * synopsis,const char * description)3445 static int sccp_wrapper_register_manager(const char * action, int authority, int (*func)(struct mansession * s, const struct message * m), const char * synopsis, const char * description)
3446 {
3447 #if defined(__cplusplus) || defined(c_plusplus)
3448 return 0;
3449 #else
3450 return ast_manager_register2(action, authority, func, ast_module_info->self, synopsis, description);
3451 #endif
3452 }
3453
sccp_wrapper_register_application(const char * app_name,int (* execute)(struct ast_channel *,const char *))3454 static int sccp_wrapper_register_application(const char *app_name, int (*execute)(struct ast_channel *, const char *))
3455 {
3456 #if defined(__cplusplus) || defined(c_plusplus)
3457 return 0;
3458 #else
3459 return ast_register_application2(app_name, execute, NULL, NULL, ast_module_info->self);
3460 #endif
3461 }
3462
sccp_wrapper_unregister_application(const char * app_name)3463 static int sccp_wrapper_unregister_application(const char *app_name)
3464 {
3465 return ast_unregister_application(app_name);
3466 }
3467
sccp_wrapper_register_function(struct pbx_custom_function * custom_function)3468 static int sccp_wrapper_register_function(struct pbx_custom_function *custom_function)
3469 {
3470 #if defined(__cplusplus) || defined(c_plusplus)
3471 return 0;
3472 #else
3473 return __ast_custom_function_register(custom_function, ast_module_info->self);
3474 #endif
3475 }
3476
sccp_wrapper_unregister_function(struct pbx_custom_function * custom_function)3477 static int sccp_wrapper_unregister_function(struct pbx_custom_function *custom_function)
3478 {
3479 return ast_custom_function_unregister(custom_function);
3480 }
3481
sccp_astwrap_setLanguage(PBX_CHANNEL_TYPE * pbxChannel,const char * language)3482 static boolean_t sccp_astwrap_setLanguage(PBX_CHANNEL_TYPE * pbxChannel, const char *language)
3483 {
3484 ast_channel_language_set(pbxChannel, language);
3485 return TRUE;
3486 }
3487
3488 #if defined(__cplusplus) || defined(c_plusplus)
3489 //sccp_pbx_cb sccp_pbx = {
3490 const PbxInterface iPbx = {
3491 /* *INDENT-OFF* */
3492
3493 /* channel */
3494 alloc_pbxChannel: sccp_astwrap_allocPBXChannel,
3495 extension_status: sccp_astwrap_extensionStatus,
3496 setPBXChannelLinkedId: sccp_astwrap_set_pbxchannel_linkedid,
3497
3498 getChannelByName: sccp_astwrap_getChannelByName,
3499 getChannelLinkedId: sccp_astwrap_get_channel_linkedid,
3500 setChannelLinkedId: sccp_astwrap_set_channel_linkedid,
3501 getChannelName: sccp_astwrap_get_channel_name,
3502 setChannelName: sccp_astwrap_set_channel_name,
3503 getChannelUniqueID: sccp_astwrap_get_channel_uniqueid,
3504 getChannelExten: sccp_astwrap_get_channel_exten,
3505 setChannelExten: sccp_astwrap_set_channel_exten,
3506 getChannelContext: sccp_astwrap_get_channel_context,
3507 setChannelContext: sccp_astwrap_set_channel_context,
3508 getChannelMacroExten: sccp_astwrap_get_channel_macroexten,
3509 setChannelMacroExten: sccp_astwrap_set_channel_macroexten,
3510 getChannelMacroContext: sccp_astwrap_get_channel_macrocontext,
3511 setChannelMacroContext: sccp_astwrap_set_channel_macrocontext,
3512 getChannelCallForward: sccp_astwrap_get_channel_call_forward,
3513 setChannelCallForward: sccp_astwrap_set_channel_call_forward,
3514
3515 getChannelAppl: sccp_astwrap_get_channel_appl,
3516 getChannelState: sccp_astwrap_get_channel_state,
3517 getChannelPbx: sccp_astwrap_get_channel_pbx,
3518
3519 getRemoteChannel: sccp_astwrap_getRemoteChannel,
3520 checkhangup: sccp_astwrap_checkHangup,
3521
3522 /* digits */
3523 send_digits: sccp_wrapper_sendDigits,
3524 send_digit: sccp_wrapper_sendDigit,
3525
3526 /* schedulers */
3527 sched_add: sccp_astwrap_sched_add,
3528 sched_del: sccp_astwrap_sched_del,
3529 sched_add_ref: sccp_astwrap_sched_add_ref,
3530 sched_del_ref: sccp_astwrap_sched_del_ref,
3531 sched_replace_ref: sccp_astwrap_sched_replace_ref,
3532 sched_when: sccp_astwrap_sched_when,
3533 sched_wait: sccp_astwrap_sched_wait,
3534
3535 /* callstate / indicate */
3536 set_callstate: sccp_astwrap_setCallState,
3537
3538 /* codecs */
3539 set_nativeAudioFormats: sccp_astwrap_setNativeAudioFormats,
3540 set_nativeVideoFormats: sccp_astwrap_setNativeVideoFormats,
3541
3542 /* rtp */
3543 rtp_getPeer: sccp_astwrap_rtpGetPeer,
3544 rtp_getUs: sccp_astwrap_rtpGetUs,
3545 rtp_stop: ast_rtp_instance_stop,
3546 rtp_create_instance: sccp_astwrap_createRtpInstance,
3547 rtp_get_payloadType: sccp_astwrap_get_payloadType,
3548 rtp_get_sampleRate: sccp_astwrap_get_sampleRate,
3549 rtp_destroy: sccp_astwrap_destroyRTP,
3550 rtp_setWriteFormat: sccp_astwrap_setWriteFormat,
3551 rtp_setReadFormat: sccp_astwrap_setReadFormat,
3552 rtp_setPhoneAddress: sccp_astwrap_setPhoneRTPAddress,
3553
3554 /* callerid */
3555 get_callerid_name: sccp_astwrap_callerid_name,
3556 get_callerid_number: sccp_astwrap_callerid_number,
3557 get_callerid_ton: sccp_astwrap_callerid_ton,
3558 get_callerid_ani: sccp_astwrap_callerid_ani,
3559 get_callerid_subaddr: sccp_astwrap_callerid_subaddr,
3560 get_callerid_dnid: sccp_astwrap_callerid_dnid,
3561 get_callerid_rdnis: sccp_astwrap_callerid_rdnis,
3562 get_callerid_presentation: sccp_astwrap_callerid_presentation,
3563
3564 set_callerid_name: sccp_astwrap_setCalleridName,
3565 set_callerid_number: sccp_astwrap_setCalleridNumber,
3566 set_callerid_ani: sccp_astwrap_setCalleridAni,
3567 set_callerid_dnid: NULL, //! \todo implement callback
3568 set_callerid_redirectingParty: sccp_astwrap_setRedirectingParty,
3569 set_callerid_redirectedParty: sccp_astwrap_setRedirectedParty,
3570 set_callerid_presentation: sccp_astwrap_setCalleridPresentation,
3571 set_dialed_number: sccp_astwrap_setDialedNumber,
3572 set_connected_line: sccp_astwrap_updateConnectedLine,
3573 sendRedirectedUpdate: sccp_astwrap_sendRedirectedUpdate,
3574
3575 /* database */
3576 feature_addToDatabase: sccp_astwrap_addToDatabase,
3577 feature_getFromDatabase: sccp_astwrap_getFromDatabase,
3578 feature_removeFromDatabase: sccp_astwrap_removeFromDatabase,
3579 feature_removeTreeFromDatabase: sccp_astwrap_removeTreeFromDatabase,
3580 feature_monitor: sccp_astgenwrap_featureMonitor,
3581
3582 feature_park: sccp_astwrap_park,
3583 getFeatureExtension: sccp_astwrap_getFeatureExtension,
3584 getPickupExtension: sccp_astwrap_getPickupExtension,
3585
3586 findChannelByCallback: sccp_astwrap_findChannelWithCallback,
3587
3588 moh_start: sccp_astwrap_moh_start,
3589 moh_stop: sccp_astwrap_moh_stop,
3590 queue_control: sccp_astwrap_queue_control,
3591 queue_control_data: sccp_astwrap_queue_control_data,
3592
3593 allocTempPBXChannel: sccp_astwrap_allocTempPBXChannel,
3594 masqueradeHelper: sccp_astwrap_masqueradeHelper,
3595 requestAnnouncementChannel: sccp_astwrap_requestAnnouncementChannel,
3596
3597 set_language: sccp_astwrap_setLanguage,
3598
3599 getExtensionState: sccp_astwrap_getExtensionState,
3600 findPickupChannelByExtenLocked: sccp_astwrap_findPickupChannelByExtenLocked,
3601 findPickupChannelByGroupLocked: sccp_astwrap_findPickupChannelByGroupLocked,
3602
3603 endpoint_create: sccp_astwrap_endpoint_create,
3604 endpoint_online: sccp_astwrap_endpoint_online,
3605 endpoint_offline: sccp_astwrap_endpoint_offline,
3606 endpoint_shutdown: sccp_astwrap_endpoint_shutdown,
3607
3608 set_owner: sccp_astwrap_setOwner,
3609 removeTimingFD: sccp_astwrap_removeTimingFD,
3610 dumpchan: sccp_astwrap_dumpchan,
3611 channel_is_bridged: sccp_astwrap_channelIsBridged,
3612 get_bridged_channel: sccp_astwrap_getBridgeChannel,
3613 get_underlying_channel: sccp_astwrap_getBridgeChannel,
3614 attended_transfer: sccp_astwrap_attended_transfer,
3615
3616 set_callgroup: sccp_astgenwrap_set_callgroup,
3617 set_pickupgroup: sccp_astgenwrap_set_pickupgroup,
3618 set_named_callgroups: sccp_astgenwrap_set_named_callgroups,
3619 set_named_pickupgroups: sccp_astgenwrap_set_named_pickupgroups,
3620
3621 register_manager: sccp_wrapper_register_manager,
3622 register_application: sccp_wrapper_register_application,
3623 unregister_application: sccp_wrapper_unregister_application,
3624 register_function: sccp_wrapper_register_function,
3625 unregister_function: sccp_wrapper_unregister_function,
3626
3627 get_codec_framing: sccp_wrapper_get_codec_framing,
3628 get_dtmf_payload_code: sccp_wrapper_get_dtmf_payload_code,
3629
3630 retrieve_remote_capabilities: pbx_retrieve_remote_capabilities,
3631 /* *INDENT-ON* */
3632 };
3633
3634 #else
3635
3636 /*!
3637 * \brief SCCP - PBX Callback Functions
3638 * (Decoupling Tight Dependencies on Asterisk Functions)
3639 */
3640 //struct sccp_pbx_cb sccp_pbx = {
3641 const PbxInterface iPbx = {
3642 /* *INDENT-OFF* */
3643
3644 /* channel */
3645 .alloc_pbxChannel = sccp_astwrap_allocPBXChannel,
3646 .extension_status = sccp_astwrap_extensionStatus,
3647 .setPBXChannelLinkedId = sccp_astwrap_set_pbxchannel_linkedid,
3648
3649 .getChannelByName = sccp_astwrap_getChannelByName,
3650 .getChannelLinkedId = sccp_astwrap_get_channel_linkedid,
3651 .setChannelLinkedId = sccp_astwrap_set_channel_linkedid,
3652 .getChannelName = sccp_astwrap_get_channel_name,
3653 .setChannelName = sccp_astwrap_set_channel_name,
3654 .getChannelUniqueID = sccp_astwrap_get_channel_uniqueid,
3655 .getChannelExten = sccp_astwrap_get_channel_exten,
3656 .setChannelExten = sccp_astwrap_set_channel_exten,
3657 .getChannelContext = sccp_astwrap_get_channel_context,
3658 .setChannelContext = sccp_astwrap_set_channel_context,
3659 .getChannelMacroExten = sccp_astwrap_get_channel_macroexten,
3660 .setChannelMacroExten = sccp_astwrap_set_channel_macroexten,
3661 .getChannelMacroContext = sccp_astwrap_get_channel_macrocontext,
3662 .setChannelMacroContext = sccp_astwrap_set_channel_macrocontext,
3663 .getChannelCallForward = sccp_astwrap_get_channel_call_forward,
3664 .setChannelCallForward = sccp_astwrap_set_channel_call_forward,
3665
3666 .getChannelAppl = sccp_astwrap_get_channel_appl,
3667 .getChannelState = sccp_astwrap_get_channel_state,
3668 .getChannelPbx = sccp_astwrap_get_channel_pbx,
3669
3670 .getRemoteChannel = sccp_astwrap_getRemoteChannel,
3671 .checkhangup = sccp_astwrap_checkHangup,
3672
3673 /* digits */
3674 .send_digits = sccp_wrapper_sendDigits,
3675 .send_digit = sccp_wrapper_sendDigit,
3676
3677 /* schedulers */
3678 .sched_add = sccp_astwrap_sched_add,
3679 .sched_del = sccp_astwrap_sched_del,
3680 .sched_add_ref = sccp_astwrap_sched_add_ref,
3681 .sched_del_ref = sccp_astwrap_sched_del_ref,
3682 .sched_replace_ref = sccp_astwrap_sched_replace_ref,
3683 .sched_when = sccp_astwrap_sched_when,
3684 .sched_wait = sccp_astwrap_sched_wait,
3685
3686 /* callstate / indicate */
3687 .set_callstate = sccp_astwrap_setCallState,
3688
3689 /* codecs */
3690 .set_nativeAudioFormats = sccp_astwrap_setNativeAudioFormats,
3691 .set_nativeVideoFormats = sccp_astwrap_setNativeVideoFormats,
3692
3693 /* rtp */
3694 .rtp_getPeer = sccp_astwrap_rtpGetPeer,
3695 .rtp_getUs = sccp_astwrap_rtpGetUs,
3696 .rtp_stop = ast_rtp_instance_stop,
3697 .rtp_create_instance = sccp_astwrap_createRtpInstance,
3698 .rtp_get_payloadType = sccp_astwrap_get_payloadType,
3699 .rtp_get_sampleRate = sccp_astwrap_get_sampleRate,
3700 .rtp_destroy = sccp_astwrap_destroyRTP,
3701 .rtp_setWriteFormat = sccp_astwrap_setWriteFormat,
3702 .rtp_setReadFormat = sccp_astwrap_setReadFormat,
3703 .rtp_setPhoneAddress = sccp_astwrap_setPhoneRTPAddress,
3704
3705 /* callerid */
3706 .get_callerid_name = sccp_astwrap_callerid_name,
3707 .get_callerid_number = sccp_astwrap_callerid_number,
3708 .get_callerid_ton = sccp_astwrap_callerid_ton,
3709 .get_callerid_ani = sccp_astwrap_callerid_ani,
3710 .get_callerid_subaddr = sccp_astwrap_callerid_subaddr,
3711 .get_callerid_dnid = sccp_astwrap_callerid_dnid,
3712 .get_callerid_rdnis = sccp_astwrap_callerid_rdnis,
3713 .get_callerid_presentation = sccp_astwrap_callerid_presentation,
3714
3715 .set_callerid_name = sccp_astwrap_setCalleridName,
3716 .set_callerid_number = sccp_astwrap_setCalleridNumber,
3717 .set_callerid_ani = sccp_astwrap_setCalleridAni,
3718 .set_callerid_dnid = NULL, //! \todo implement callback
3719 .set_callerid_redirectingParty = sccp_astwrap_setRedirectingParty,
3720 .set_callerid_redirectedParty = sccp_astwrap_setRedirectedParty,
3721 .set_callerid_presentation = sccp_astwrap_setCalleridPresentation,
3722 .set_dialed_number = sccp_astwrap_setDialedNumber,
3723 .set_connected_line = sccp_astwrap_updateConnectedLine,
3724 .sendRedirectedUpdate = sccp_astwrap_sendRedirectedUpdate,
3725
3726 /* database */
3727 .feature_addToDatabase = sccp_astwrap_addToDatabase,
3728 .feature_getFromDatabase = sccp_astwrap_getFromDatabase,
3729 .feature_removeFromDatabase = sccp_astwrap_removeFromDatabase,
3730 .feature_removeTreeFromDatabase = sccp_astwrap_removeTreeFromDatabase,
3731 .feature_monitor = sccp_astgenwrap_featureMonitor,
3732
3733 .feature_park = sccp_astwrap_park,
3734 .getFeatureExtension = sccp_astwrap_getFeatureExtension,
3735 .getPickupExtension = sccp_astwrap_getPickupExtension,
3736
3737 .findChannelByCallback = sccp_astwrap_findChannelWithCallback,
3738
3739 .moh_start = sccp_astwrap_moh_start,
3740 .moh_stop = sccp_astwrap_moh_stop,
3741 .queue_control = sccp_astwrap_queue_control,
3742 .queue_control_data = sccp_astwrap_queue_control_data,
3743
3744 .allocTempPBXChannel = sccp_astwrap_allocTempPBXChannel,
3745 .masqueradeHelper = sccp_astwrap_masqueradeHelper,
3746 .requestAnnouncementChannel = sccp_astwrap_requestAnnouncementChannel,
3747
3748 .set_language = sccp_astwrap_setLanguage,
3749
3750 .getExtensionState = sccp_astwrap_getExtensionState,
3751 .findPickupChannelByExtenLocked = sccp_astwrap_findPickupChannelByExtenLocked,
3752 .findPickupChannelByGroupLocked = sccp_astwrap_findPickupChannelByGroupLocked,
3753
3754 .endpoint_create = sccp_astwrap_endpoint_create,
3755 .endpoint_online = sccp_astwrap_endpoint_online,
3756 .endpoint_offline = sccp_astwrap_endpoint_offline,
3757 .endpoint_shutdown = sccp_astwrap_endpoint_shutdown,
3758
3759 .set_owner = sccp_astwrap_setOwner,
3760 .removeTimingFD = sccp_astwrap_removeTimingFD,
3761 .dumpchan = sccp_astwrap_dumpchan,
3762 .channel_is_bridged = sccp_astwrap_channelIsBridged,
3763 .get_bridged_channel = sccp_astwrap_getBridgeChannel,
3764 .get_underlying_channel = sccp_astwrap_getBridgeChannel,
3765 .attended_transfer = sccp_astwrap_attended_transfer,
3766
3767 .set_callgroup = sccp_astgenwrap_set_callgroup,
3768 .set_pickupgroup = sccp_astgenwrap_set_pickupgroup,
3769 .set_named_callgroups = sccp_astgenwrap_set_named_callgroups,
3770 .set_named_pickupgroups = sccp_astgenwrap_set_named_pickupgroups,
3771
3772 .register_manager = sccp_wrapper_register_manager,
3773 .register_application = sccp_wrapper_register_application,
3774 .unregister_application = sccp_wrapper_unregister_application,
3775 .register_function = sccp_wrapper_register_function,
3776 .unregister_function = sccp_wrapper_unregister_function,
3777
3778 .get_codec_framing = sccp_wrapper_get_codec_framing,
3779 .get_dtmf_payload_code = sccp_wrapper_get_dtmf_payload_code,
3780
3781 .retrieve_remote_capabilities = pbx_retrieve_remote_capabilities,
3782 /* *INDENT-ON* */
3783 };
3784 #endif
3785
register_channel_tech(struct ast_channel_tech * tech)3786 static int register_channel_tech(struct ast_channel_tech *tech)
3787 {
3788 tech->capabilities = ast_format_cap_alloc((enum ast_format_cap_flags)0);
3789 if (!tech->capabilities) {
3790 ao2_cleanup(tech->capabilities);
3791 return -1;
3792 }
3793 ast_format_cap_append_by_type(tech->capabilities, AST_MEDIA_TYPE_AUDIO);
3794 ast_format_cap_append_by_type(tech->capabilities, AST_MEDIA_TYPE_VIDEO);
3795 //ast_format_cap_append_by_type(tech->capabilities, AST_MEDIA_TYPE_TEXT);
3796 //pbx_format_cap_append_skinny(tech->capabilities, GLOB(global_preferences));
3797
3798 if (ast_channel_register(tech)) {
3799 ast_log(LOG_ERROR, "Unable to register channel technology %s(%s).\n", tech->type, tech->description);
3800 return -1;
3801 }
3802 return 0;
3803 }
3804
unregister_channel_tech(struct ast_channel_tech * tech)3805 static void unregister_channel_tech(struct ast_channel_tech *tech)
3806 {
3807 ast_channel_unregister(tech);
3808 if (tech->capabilities) {
3809 ao2_ref(tech->capabilities, -1);
3810 }
3811 tech->capabilities = NULL;
3812 tech = NULL;
3813 }
3814
unload_module(void)3815 static int unload_module(void)
3816 {
3817 pbx_log(LOG_NOTICE, "SCCP: Module Unload\n");
3818 sccp_preUnload();
3819 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_2 "SCCP: Unregister SCCP RTP protocol\n");
3820 ast_rtp_glue_unregister(&sccp_rtp);
3821 sccp_log((DEBUGCAT_CORE)) (VERBOSE_PREFIX_2 "SCCP: Unregister SCCP Channel Tech\n");
3822
3823 unregister_channel_tech(&sccp_tech);
3824 sccp_unregister_dialplan_functions();
3825 sccp_unregister_cli();
3826 #ifdef CS_SCCP_MANAGER
3827 sccp_unregister_management();
3828 #endif
3829 #ifdef HAVE_PBX_MESSAGE_H
3830 ast_msg_tech_unregister(&sccp_msg_tech);
3831 #endif
3832 #ifdef CS_SCCP_CONFERENCE
3833 unregister_channel_tech(sccpconf_announce_get_tech());
3834 #endif
3835 if (io) {
3836 io_context_destroy(io);
3837 io = NULL;
3838 }
3839
3840 while (SCCP_REF_DESTROYED != sccp_refcount_isRunning()) {
3841 usleep(SCCP_TIME_TO_KEEP_REFCOUNTEDOBJECT); // give enough time for all schedules to end and refcounted object to be cleanup completely
3842 }
3843
3844 if (sched) {
3845 pbx_log(LOG_NOTICE, "Cleaning up scheduled items:\n");
3846 int scheduled_items = 0;
3847
3848 ast_sched_dump(sched);
3849 while ((scheduled_items = ast_sched_runq(sched))) {
3850 pbx_log(LOG_NOTICE, "Cleaning up %d scheduled items... please wait\n", scheduled_items);
3851 usleep(ast_sched_wait(sched));
3852 }
3853 ast_sched_context_destroy(sched);
3854 sched = NULL;
3855 }
3856
3857 pbx_log(LOG_NOTICE, "Running Cleanup\n");
3858 sccp_free(sccp_globals);
3859 pbx_log(LOG_NOTICE, "Module chan_sccp unloaded\n");
3860 return 0;
3861 }
3862
load_module(void)3863 static enum ast_module_load_result load_module(void)
3864 {
3865 if (ast_module_check("chan_skinny.so")) {
3866 pbx_log(LOG_ERROR, "Chan_skinny is loaded. Please check modules.conf and remove chan_skinny before loading chan_sccp.\n");
3867 return AST_MODULE_LOAD_SUCCESS;
3868 }
3869
3870 enum ast_module_load_result res = AST_MODULE_LOAD_DECLINE;
3871 do {
3872 if (!(sched = ast_sched_context_create())) {
3873 pbx_log(LOG_WARNING, "Unable to create schedule context. SCCP channel type disabled\n");
3874 break;
3875 }
3876 if (ast_sched_start_thread(sched)) {
3877 pbx_log(LOG_ERROR, "Unable to start scheduler\n");
3878 ast_sched_context_destroy(sched);
3879 sched = NULL;
3880 break;
3881 }
3882 #if defined(CS_DEVSTATE_FEATURE) || defined(CS_USE_ASTERISK_DISTRIBUTED_DEVSTATE)
3883 // ast_enable_distributed_devstate();
3884 #endif
3885 if (!sccp_prePBXLoad()) {
3886 pbx_log(LOG_ERROR, "SCCP: prePBXLoad Failed\n");
3887 break;
3888 }
3889 if (!(io = io_context_create())) {
3890 pbx_log(LOG_ERROR, "Unable to create I/O context. SCCP channel type disabled\n");
3891 break;
3892 }
3893 if (!load_config()) {
3894 pbx_log(LOG_ERROR, "SCCP: config file could not be parsed\n");
3895 res = AST_MODULE_LOAD_DECLINE;
3896 break;
3897 }
3898 if (register_channel_tech(&sccp_tech)) {
3899 pbx_log(LOG_ERROR, "Unable to register channel class SCCP\n");
3900 break;
3901 }
3902 #ifdef HAVE_PBX_MESSAGE_H
3903 if (ast_msg_tech_register(&sccp_msg_tech)) {
3904 pbx_log(LOG_ERROR, "Unable to register message interface\n");
3905 break;
3906 }
3907 #endif
3908
3909 #ifdef CS_SCCP_CONFERENCE
3910 if (register_channel_tech(sccpconf_announce_get_tech())) {
3911 pbx_log(LOG_ERROR, "Unable to register channel class ANNOUNCE (conference)\n");
3912 break;
3913 }
3914 #endif
3915 if (ast_rtp_glue_register(&sccp_rtp)) {
3916 pbx_log(LOG_ERROR, "Unable to register RTP Glue\n");
3917 break;
3918 }
3919 if (sccp_register_management()) {
3920 pbx_log(LOG_ERROR, "Unable to register management functions");
3921 break;
3922 }
3923 if (sccp_register_cli()) {
3924 pbx_log(LOG_ERROR, "Unable to register CLI functions");
3925 break;
3926 }
3927 if (sccp_register_dialplan_functions()) {
3928 pbx_log(LOG_ERROR, "Unable to register dialplan functions");
3929 break;
3930 }
3931 if (!sccp_postPBX_load()) {
3932 pbx_log(LOG_ERROR, "SCCP: postPBXLoad Failed\n");
3933 break;
3934 }
3935 res = AST_MODULE_LOAD_SUCCESS;
3936 } while (0);
3937
3938 if (res != AST_MODULE_LOAD_SUCCESS) {
3939 pbx_log(LOG_ERROR, "SCCP: Module Load Failed, unloading...\n");
3940 unload_module();
3941 return res;
3942 }
3943 return res;
3944 }
3945
module_reload(void)3946 static int module_reload(void)
3947 {
3948 return sccp_reload();
3949 }
3950
3951 #if defined(__cplusplus) || defined(c_plusplus)
3952 static struct ast_module_info __mod_info = {
3953 NULL,
3954 load_module,
3955 module_reload,
3956 unload_module,
3957 NULL,
3958 NULL,
3959 AST_MODULE,
3960 SCCP_VERSIONSTR,
3961 ASTERISK_GPL_KEY,
3962 AST_MODFLAG_LOAD_ORDER,
3963 AST_BUILDOPT_SUM,
3964 AST_MODPRI_CHANNEL_DRIVER,
3965 NULL,
3966 };
3967
__reg_module(void)3968 static void __attribute__ ((constructor)) __reg_module(void)
3969 {
3970 ast_module_register(&__mod_info);
3971 }
3972
__unreg_module(void)3973 static void __attribute__ ((destructor)) __unreg_module(void)
3974 {
3975 ast_module_unregister(&__mod_info);
3976 }
3977
3978 static const __attribute__ ((unused))
3979 struct ast_module_info *ast_module_info = &__mod_info;
3980 #else
3981 AST_MODULE_INFO(
3982 ASTERISK_GPL_KEY,
3983 AST_MODFLAG_LOAD_ORDER,
3984 SCCP_VERSIONSTR,
3985 .load = load_module,
3986 .unload = unload_module,
3987 .reload = module_reload,
3988 .load_pri = AST_MODPRI_DEFAULT,
3989 .nonoptreq = "chan_local"
3990 );
3991 #endif
3992
sccp_astwrap_findPickupChannelByExtenLocked(PBX_CHANNEL_TYPE * chan,const char * exten,const char * context)3993 PBX_CHANNEL_TYPE *sccp_astwrap_findPickupChannelByExtenLocked(PBX_CHANNEL_TYPE * chan, const char *exten, const char *context)
3994 {
3995 struct ast_channel *target = NULL; /*!< Potential pickup target */
3996 struct ast_channel_iterator *iter;
3997
3998 if (!(iter = ast_channel_iterator_by_exten_new(exten, context))) {
3999 return NULL;
4000 }
4001
4002 //ast_log(LOG_NOTICE, "(findPickupChannelByExtenLocked) checking pickup of channel: %s, context: %s, exten: %s, iter: %p\n", pbx_channel_name(chan), exten, context, iter);
4003 while ((target = ast_channel_iterator_next(iter))) {
4004 ast_channel_lock(target);
4005 //ast_log(LOG_NOTICE, "(findPickupChannelByExtenLocked) checking channel: %s, target:%s, can_pickup: %d\n", pbx_channel_name(chan), pbx_channel_name(target), ast_can_pickup(target));
4006 if ((chan != target) && ast_can_pickup(target)) {
4007 ast_log(LOG_NOTICE, "%s pickup by %s\n", ast_channel_name(target), ast_channel_name(chan));
4008 break;
4009 }
4010 ast_channel_unlock(target);
4011 target = pbx_channel_unref(target);
4012 }
4013
4014 ast_channel_iterator_destroy(iter);
4015 return target;
4016 }
4017
sccp_astwrap_findPickupChannelByGroupLocked(PBX_CHANNEL_TYPE * chan)4018 PBX_CHANNEL_TYPE *sccp_astwrap_findPickupChannelByGroupLocked(PBX_CHANNEL_TYPE * chan)
4019 {
4020 struct ast_channel *target = NULL; /*!< Potential pickup target */
4021
4022 target = ast_pickup_find_by_group(chan);
4023 return target;
4024 }
4025
4026 // kate: indent-width 8; replace-tabs off; indent-mode cstyle; auto-insert-doxygen on; line-numbers on; tab-indents on; keep-extra-spaces off; auto-brackets off;
4027