Asterisk - The Open Source Telephony Project  GIT-master-c559667
Data Structures | Macros | Functions | Variables
chan_pjsip.c File Reference

PSJIP SIP Channel Driver. More...

#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjlib.h>
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/causes.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/dsp.h"
#include "asterisk/stasis_endpoints.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/indications.h"
#include "asterisk/format_cache.h"
#include "asterisk/translate.h"
#include "asterisk/threadstorage.h"
#include "asterisk/features_config.h"
#include "asterisk/pickup.h"
#include "asterisk/test.h"
#include "asterisk/message.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/stream.h"
#include "pjsip/include/chan_pjsip.h"
#include "pjsip/include/dialplan_functions.h"
#include "pjsip/include/cli_functions.h"

Go to the source code of this file.

Data Structures

struct  answer_data
 
struct  hangup_data
 
struct  indicate_data
 
struct  info_dtmf_data
 
struct  request_data
 
struct  rtp_direct_media_data
 
struct  sendtext_data
 
struct  topology_change_refresh_data
 
struct  transfer_data
 

Macros

#define UNIQUEID_BUFSIZE   256
 

Functions

static void __init_uniqueid_threadbuf (void)
 
static void __reg_module (void)
 
static void __unreg_module (void)
 
static int answer (void *data)
 
struct ast_moduleAST_MODULE_SELF_SYM (void)
 
static int call (void *data)
 
static int call_pickup_incoming_request (struct ast_sip_session *session, pjsip_rx_data *rdata)
 
static int chan_pjsip_add_hold (const char *chan_uid)
 Add a channel ID to the list of PJSIP channels on hold. More...
 
static int chan_pjsip_answer (struct ast_channel *ast)
 Function called by core when we should answer a PJSIP session. More...
 
static int chan_pjsip_call (struct ast_channel *ast, const char *dest, int timeout)
 Function called by core to actually start calling a remote party. More...
 
static struct ast_framechan_pjsip_cng_tone_detected (struct ast_channel *ast, struct ast_sip_session *session, struct ast_frame *f)
 Internal helper function called when CNG tone is detected. More...
 
static int chan_pjsip_devicestate (const char *data)
 Function called to get the device state of an endpoint. More...
 
static int chan_pjsip_digit_begin (struct ast_channel *chan, char digit)
 Function called by core to start a DTMF digit. More...
 
static int chan_pjsip_digit_end (struct ast_channel *ast, char digit, unsigned int duration)
 Function called by core to stop a DTMF digit. More...
 
static int chan_pjsip_fixup (struct ast_channel *oldchan, struct ast_channel *newchan)
 Function called by core to change the underlying owner channel. More...
 
static void chan_pjsip_get_codec (struct ast_channel *chan, struct ast_format_cap *result)
 Function called by RTP engine to get peer capabilities. More...
 
static int chan_pjsip_get_hold (const char *chan_uid)
 Determine whether a channel ID is in the list of PJSIP channels on hold. More...
 
static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
 Function called by RTP engine to get local audio RTP peer. More...
 
static const char * chan_pjsip_get_uniqueid (struct ast_channel *ast)
 
static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
 Function called by RTP engine to get local video RTP peer. More...
 
static int chan_pjsip_hangup (struct ast_channel *ast)
 Function called by core to hang up a PJSIP session. More...
 
static int chan_pjsip_incoming_ack (struct ast_sip_session *session, struct pjsip_rx_data *rdata)
 
static int chan_pjsip_incoming_request (struct ast_sip_session *session, struct pjsip_rx_data *rdata)
 Function called when a request is received on the session. More...
 
static void chan_pjsip_incoming_response (struct ast_sip_session *session, struct pjsip_rx_data *rdata)
 Function called when a response is received on the session. More...
 
static void chan_pjsip_incoming_response_update_cause (struct ast_sip_session *session, struct pjsip_rx_data *rdata)
 Function called when a response is received on the session. More...
 
static int chan_pjsip_indicate (struct ast_channel *ast, int condition, const void *data, size_t datalen)
 Function called by core to ask the channel to indicate some sort of condition. More...
 
static struct ast_channelchan_pjsip_new (struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name)
 Function called to create a new PJSIP Asterisk channel. More...
 
static void chan_pjsip_pvt_dtor (void *obj)
 
static int chan_pjsip_queryoption (struct ast_channel *ast, int option, void *data, int *datalen)
 Function called to query options on a channel. More...
 
static struct ast_framechan_pjsip_read_stream (struct ast_channel *ast)
 Function called by core to read any waiting frames. More...
 
static void chan_pjsip_remove_hold (const char *chan_uid)
 Remove a channel ID from the list of PJSIP channels on hold. More...
 
static struct ast_channelchan_pjsip_request (const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
 
static struct ast_channelchan_pjsip_request_with_stream_topology (const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
 Function called by core to create a new outgoing PJSIP session. More...
 
static int chan_pjsip_sendtext (struct ast_channel *ast, const char *text)
 
static int chan_pjsip_sendtext_data (struct ast_channel *ast, struct ast_msg_data *msg)
 Function called by core to send text on PJSIP session. More...
 
static void chan_pjsip_session_begin (struct ast_sip_session *session)
 SIP session interaction functions. More...
 
static void chan_pjsip_session_end (struct ast_sip_session *session)
 Function called when the session ends. More...
 
static int chan_pjsip_set_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, const struct ast_format_cap *cap, int nat_active)
 Function called by RTP engine to change where the remote party should send media. More...
 
static int chan_pjsip_transfer (struct ast_channel *chan, const char *target)
 Function called by core for Asterisk initiated transfer. More...
 
static int chan_pjsip_write (struct ast_channel *ast, struct ast_frame *f)
 
static int chan_pjsip_write_stream (struct ast_channel *ast, int stream_num, struct ast_frame *f)
 
static int check_for_rtp_changes (struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_sip_session_media *media, struct ast_sip_session *session)
 
static void clear_session_and_channel (struct ast_sip_session *session, struct ast_channel *ast)
 Clear a channel from a session along with its PVT. More...
 
static int compatible_formats_exist (struct ast_stream_topology *top, struct ast_format_cap *cap)
 Determine if a topology is compatible with format capabilities. More...
 
static int direct_media_mitigate_glare (struct ast_sip_session *session)
 
static int handle_topology_request_change (struct ast_sip_session *session, const struct ast_stream_topology *proposed)
 
static int hangup (void *data)
 
static int hangup_cause2sip (int cause)
 Internal function which translates from Asterisk cause codes to SIP response codes. More...
 
static struct hangup_datahangup_data_alloc (int cause, struct ast_channel *chan)
 
static void hangup_data_destroy (void *obj)
 
static int hangup_sip2cause (int cause)
 Convert SIP hangup causes to Asterisk hangup causes. More...
 
static int indicate (void *data)
 
static struct indicate_dataindicate_data_alloc (struct ast_sip_session *session, int condition, int response_code, const void *frame_data, size_t datalen)
 
static void indicate_data_destroy (void *obj)
 
static struct info_dtmf_datainfo_dtmf_data_alloc (struct ast_sip_session *session, char digit, unsigned int duration)
 
static void info_dtmf_data_destroy (void *obj)
 
static int is_colp_update_allowed (struct ast_sip_session *session)
 
static int is_compatible_format (struct ast_sip_session *session, struct ast_frame *f)
 Determine if the given frame is in a format we've negotiated. More...
 
static int load_module (void)
 Load the module. More...
 
static void local_hold_set_state (struct ast_sip_session_media *session_media, unsigned int held)
 Callback which changes the value of locally held on the media stream. More...
 
static int on_topology_change_response (struct ast_sip_session *session, pjsip_rx_data *rdata)
 
static int pbx_start_incoming_request (struct ast_sip_session *session, pjsip_rx_data *rdata)
 
static int remote_send_hold (void *data)
 Update local hold state to be held. More...
 
static int remote_send_hold_refresh (struct ast_sip_session *session, unsigned int held)
 Update local hold state and send a re-INVITE with the new SDP. More...
 
static int remote_send_unhold (void *data)
 Update local hold state to be unheld. More...
 
static int request (void *obj)
 
static struct rtp_direct_media_datartp_direct_media_data_create (struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, const struct ast_format_cap *cap, struct ast_sip_session *session)
 
static void rtp_direct_media_data_destroy (void *data)
 
static int rtp_find_rtcp_fd_position (struct ast_sip_session *session, struct ast_rtp_instance *rtp)
 Helper function to find the position for RTCP. More...
 
static int send_direct_media_request (void *data)
 
static int send_topology_change_refresh (void *data)
 
static int sendtext (void *obj)
 
static struct sendtext_datasendtext_data_create (struct ast_channel *chan, struct ast_msg_data *msg)
 
static void sendtext_data_destroy (void *obj)
 
static void set_channel_on_rtp_instance (const struct ast_sip_session *session, const char *channel_id)
 
static void set_sipdomain_variable (struct ast_sip_session *session)
 
static struct topology_change_refresh_datatopology_change_refresh_data_alloc (struct ast_sip_session *session, const struct ast_stream_topology *topology)
 
static void topology_change_refresh_data_free (struct topology_change_refresh_data *refresh_data)
 
static int transfer (void *data)
 
static struct transfer_datatransfer_data_alloc (struct ast_sip_session *session, const char *target)
 
static void transfer_data_destroy (void *obj)
 
static void transfer_redirect (struct ast_sip_session *session, const char *target)
 
static void transfer_refer (struct ast_sip_session *session, const char *target)
 
static int transmit_info_dtmf (void *data)
 
static int transmit_info_with_vidupdate (void *data)
 Send SIP INFO with video update request. More...
 
static void transport_info_destroy (void *obj)
 Destructor function for transport_info_data. More...
 
static int uid_hold_hash_fn (const void *obj, const int flags)
 
static int uid_hold_sort_fn (const void *obj_left, const void *obj_right, const int flags)
 
static int unload_module (void)
 Unload the PJSIP channel from Asterisk. More...
 
static int update_connected_line_information (void *data)
 Update connected line information. More...
 
static int update_devstate (void *obj, void *arg, int flags)
 
static void update_initial_connected_line (struct ast_sip_session *session)
 
static void xfer_client_on_evsub_state (pjsip_evsub *sub, pjsip_event *event)
 Callback function to report status of implicit REFER-NOTIFY subscription. More...
 

Variables

static struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "PJSIP Channel Driver" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_pjsip,res_pjsip_session,res_pjsip_pubsub", }
 
static const struct ast_module_infoast_module_info = &__mod_info
 
static struct ast_sip_session_supplement call_pickup_supplement
 
static unsigned int chan_idx
 
static struct ast_sip_session_supplement chan_pjsip_ack_supplement
 
static struct ast_custom_function chan_pjsip_dial_contacts_function
 
static struct ast_custom_function chan_pjsip_parse_uri_function
 
static struct ast_rtp_glue chan_pjsip_rtp_glue
 Local glue for interacting with the RTP engine core. More...
 
static struct ast_sip_session_supplement chan_pjsip_supplement
 SIP session supplement structure. More...
 
static struct ast_sip_session_supplement chan_pjsip_supplement_response
 SIP session supplement structure just for responses. More...
 
struct ast_channel_tech chan_pjsip_tech
 PBX interface structure for channel registration. More...
 
static const char channel_type [] = "PJSIP"
 
static struct ast_datastore_info direct_media_mitigation_info = { }
 
static struct ast_custom_function dtmf_mode_function
 
static struct ast_custom_function media_offer_function
 
static struct ast_custom_function moh_passthrough_function
 
static struct ast_sip_session_supplement pbx_start_supplement
 
static struct ao2_containerpjsip_uids_onhold
 
static pjsip_module refer_callback_module
 REFER Callback module, used to attach session data structure to subscription. More...
 
static struct ast_custom_function session_refresh_function
 
static struct ast_datastore_info transport_info
 Datastore used to store local/remote addresses for the INVITE request that created the PJSIP channel. More...
 
static struct ast_threadstorage uniqueid_threadbuf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_uniqueid_threadbuf , .custom_init = NULL , }
 

Detailed Description

PSJIP SIP Channel Driver.

Author
Joshua Colp jcolp.nosp@m.@dig.nosp@m.ium.c.nosp@m.om

Definition in file chan_pjsip.c.

Macro Definition Documentation

◆ UNIQUEID_BUFSIZE

#define UNIQUEID_BUFSIZE   256

Definition at line 76 of file chan_pjsip.c.

Referenced by chan_pjsip_get_uniqueid().

Function Documentation

◆ __init_uniqueid_threadbuf()

static void __init_uniqueid_threadbuf ( void  )
static

Definition at line 75 of file chan_pjsip.c.

83 {

◆ __reg_module()

static void __reg_module ( void  )
static

Definition at line 3418 of file chan_pjsip.c.

◆ __unreg_module()

static void __unreg_module ( void  )
static

Definition at line 3418 of file chan_pjsip.c.

◆ answer()

static int answer ( void *  data)
static

Definition at line 682 of file chan_pjsip.c.

References ast_channel_name(), ast_log, ast_sip_session_get_name(), ast_sip_session_send_response(), ast_sip_session::channel, answer_data::indent, ast_sip_session::inv_session, LOG_ERROR, LOG_WARNING, NULL, SCOPE_ENTER_TASK, SCOPE_EXIT_RTN_VALUE, answer_data::session, and status.

Referenced by ast_raw_answer_with_stream_topology(), chan_pjsip_answer(), dns_parse_answer(), dns_parse_answer_ex(), dump_answer(), find_and_retrans(), pbx_builtin_incomplete(), session_inv_on_rx_offer(), stun_monitor_request(), tds_log(), verify_mock_cdr_record(), and zapateller_exec().

683 {
684  struct answer_data *ans_data = data;
685  pj_status_t status = PJ_SUCCESS;
686  pjsip_tx_data *packet = NULL;
687  struct ast_sip_session *session = ans_data->session;
688  SCOPE_ENTER_TASK(1, ans_data->indent, "%s\n", ast_sip_session_get_name(session));
689 
690  if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
691  ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
692  session->inv_session->cause,
693  pjsip_get_status_text(session->inv_session->cause)->ptr);
694  SCOPE_EXIT_RTN_VALUE(0, "Disconnected\n");
695  }
696 
697  pjsip_dlg_inc_lock(session->inv_session->dlg);
698  if (session->inv_session->invite_tsx) {
699  status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
700  } else {
701  ast_log(LOG_ERROR,"Cannot answer '%s' because there is no associated SIP transaction\n",
702  ast_channel_name(session->channel));
703  }
704  pjsip_dlg_dec_lock(session->inv_session->dlg);
705 
706  if (status == PJ_SUCCESS && packet) {
707  ast_sip_session_send_response(session, packet);
708  }
709 
710  if (status != PJ_SUCCESS) {
711  char err[PJ_ERR_MSG_SIZE];
712 
713  pj_strerror(status, err, sizeof(err));
714  ast_log(LOG_WARNING,"Cannot answer '%s': %s\n",
715  ast_channel_name(session->channel), err);
716  /*
717  * Return this value so we can distinguish between this
718  * failure and the threadpool synchronous push failing.
719  */
720  SCOPE_EXIT_RTN_VALUE(-2, "pjproject failure\n");
721  }
723 }
#define SCOPE_ENTER_TASK(level, indent,...)
Definition: logger.h:889
#define LOG_WARNING
Definition: logger.h:274
#define NULL
Definition: resample.c:96
struct pjsip_inv_session * inv_session
A structure describing a SIP session.
#define ast_log
Definition: astobj2.c:42
unsigned long indent
Definition: chan_pjsip.c:679
static struct ast_mansession session
struct ast_channel * channel
void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
Send a SIP response.
#define LOG_ERROR
Definition: logger.h:285
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
const char * ast_channel_name(const struct ast_channel *chan)
struct ast_sip_session * session
Definition: chan_pjsip.c:678
jack_status_t status
Definition: app_jack.c:146

◆ AST_MODULE_SELF_SYM()

struct ast_module* AST_MODULE_SELF_SYM ( void  )

Definition at line 3418 of file chan_pjsip.c.

◆ call()

static int call ( void *  data)
static

Definition at line 2347 of file chan_pjsip.c.

References ao2_ref, ast_channel_name(), ast_channel_uniqueid(), ast_queue_hangup(), ast_set_hangupsource(), ast_sip_session_create_invite(), ast_sip_session_get_name(), ast_sip_session_send_request(), ast_str_tmp, ast_stream_topology_to_str(), ast_sip_session::channel, ast_sip_session::pending_media_state, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, ast_sip_channel_pvt::session, set_channel_on_rtp_instance(), ast_sip_session_media_state::topology, and update_initial_connected_line().

Referenced by ast_call(), chan_pjsip_call(), and native_start().

2348 {
2349  struct ast_sip_channel_pvt *channel = data;
2350  struct ast_sip_session *session = channel->session;
2351  pjsip_tx_data *tdata;
2352  int res = 0;
2353  SCOPE_ENTER(1, "%s Topology: %s\n",
2354  ast_sip_session_get_name(session),
2356  );
2357 
2358 
2359  res = ast_sip_session_create_invite(session, &tdata);
2360 
2361  if (res) {
2362  ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
2363  ast_queue_hangup(session->channel);
2364  } else {
2367  ast_sip_session_send_request(session, tdata);
2368  }
2369  ao2_ref(channel, -1);
2370  SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
2371 }
int ast_queue_hangup(struct ast_channel *chan)
Queue a hangup frame.
Definition: channel.c:1146
void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
Send a SIP request.
struct ast_sip_session_media_state * pending_media_state
int ast_sip_session_create_invite(struct ast_sip_session *session, pjsip_tx_data **tdata)
Creates an INVITE request.
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
A structure describing a SIP session.
#define ast_str_tmp(init_len, __expr)
Definition: strings.h:1136
static void set_channel_on_rtp_instance(const struct ast_sip_session *session, const char *channel_id)
Definition: chan_pjsip.c:494
const char * ast_stream_topology_to_str(const struct ast_stream_topology *topology, struct ast_str **buf)
Get a string representing the topology for debugging/display purposes.
Definition: stream.c:936
void ast_set_hangupsource(struct ast_channel *chan, const char *source, int force)
Set the source of the hangup in this channel and it&#39;s bridge.
Definition: channel.c:2490
static struct ast_mansession session
#define ao2_ref(o, delta)
Definition: astobj2.h:464
struct ast_channel * channel
const char * ast_channel_uniqueid(const struct ast_channel *chan)
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
struct ast_stream_topology * topology
The media stream topology.
const char * ast_channel_name(const struct ast_channel *chan)
static void update_initial_connected_line(struct ast_sip_session *session)
Definition: chan_pjsip.c:2322

◆ call_pickup_incoming_request()

static int call_pickup_incoming_request ( struct ast_sip_session session,
pjsip_rx_data *  rdata 
)
static

Definition at line 3054 of file chan_pjsip.c.

References ao2_ref, AST_CAUSE_CALL_REJECTED, AST_CAUSE_NORMAL_CLEARING, ast_channel_hangupcause_set(), ast_channel_ref, ast_channel_unref, ast_get_chan_features_pickup_config(), ast_hangup(), ast_log, ast_pickup_call(), rtp_direct_media_data::chan, ast_sip_session::channel, ast_sip_session::exten, LOG_ERROR, and ast_features_pickup_config::pickupexten.

3055 {
3056  struct ast_features_pickup_config *pickup_cfg;
3057  struct ast_channel *chan;
3058 
3059  /* Check for a to-tag to determine if this is a reinvite */
3060  if (rdata->msg_info.to->tag.slen) {
3061  /* We don't care about reinvites */
3062  return 0;
3063  }
3064 
3065  pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
3066  if (!pickup_cfg) {
3067  ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n");
3068  return 0;
3069  }
3070 
3071  if (strcmp(session->exten, pickup_cfg->pickupexten)) {
3072  ao2_ref(pickup_cfg, -1);
3073  return 0;
3074  }
3075  ao2_ref(pickup_cfg, -1);
3076 
3077  /* We can't directly use session->channel because the pickup operation will cause a masquerade to occur,
3078  * changing the channel pointer in session to a different channel. To ensure we work on the right channel
3079  * we store a pointer locally before we begin and keep a reference so it remains valid no matter what.
3080  */
3081  chan = ast_channel_ref(session->channel);
3082  if (ast_pickup_call(chan)) {
3084  } else {
3086  }
3087  /* A hangup always occurs because the pickup operation will have either failed resulting in the call
3088  * needing to be hung up OR the pickup operation was a success and the channel we now have is actually
3089  * the channel that was replaced, which should be hung up since it is literally in limbo not connected
3090  * to anything at all.
3091  */
3092  ast_hangup(chan);
3093  ast_channel_unref(chan);
3094 
3095  return 1;
3096 }
Main Channel structure associated with a channel.
struct ast_features_pickup_config * ast_get_chan_features_pickup_config(struct ast_channel *chan)
Get the pickup configuration options for a channel.
#define ast_channel_unref(c)
Decrease channel reference count.
Definition: channel.h:2926
void ast_channel_hangupcause_set(struct ast_channel *chan, int value)
char exten[AST_MAX_EXTENSION]
#define ast_log
Definition: astobj2.c:42
#define AST_CAUSE_NORMAL_CLEARING
Definition: causes.h:105
#define ao2_ref(o, delta)
Definition: astobj2.h:464
struct ast_channel * channel
#define LOG_ERROR
Definition: logger.h:285
int ast_pickup_call(struct ast_channel *chan)
Pickup a call.
Definition: pickup.c:200
void ast_hangup(struct ast_channel *chan)
Hang up a channel.
Definition: channel.c:2534
#define ast_channel_ref(c)
Increase channel reference count.
Definition: channel.h:2915
#define AST_CAUSE_CALL_REJECTED
Definition: causes.h:110
Configuration relating to call pickup.

◆ chan_pjsip_add_hold()

static int chan_pjsip_add_hold ( const char *  chan_uid)
static

Add a channel ID to the list of PJSIP channels on hold.

Parameters
chan_uid- Unique ID of the channel being put into the hold list
Return values
0Channel has been added to or was already in the hold list
-1Failed to add channel to the hold list

Definition at line 1117 of file chan_pjsip.c.

References AO2_ALLOC_OPT_LOCK_NOLOCK, ao2_alloc_options, ao2_cleanup, ao2_find, ao2_link, ast_copy_string(), NULL, OBJ_SEARCH_KEY, and RAII_VAR.

Referenced by chan_pjsip_indicate().

1118 {
1119  RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
1120 
1121  hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
1122  if (hold_uid) {
1123  /* Device is already on hold. Nothing to do. */
1124  return 0;
1125  }
1126 
1127  /* Device wasn't in hold list already. Create a new one. */
1128  hold_uid = ao2_alloc_options(strlen(chan_uid) + 1, NULL,
1130  if (!hold_uid) {
1131  return -1;
1132  }
1133 
1134  ast_copy_string(hold_uid, chan_uid, strlen(chan_uid) + 1);
1135 
1136  if (ao2_link(pjsip_uids_onhold, hold_uid) == 0) {
1137  return -1;
1138  }
1139 
1140  return 0;
1141 }
The arg parameter is a search key, but is not an object.
Definition: astobj2.h:1105
#define ao2_alloc_options(data_size, destructor_fn, options)
Definition: astobj2.h:406
#define NULL
Definition: resample.c:96
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition: utils.h:851
static struct ao2_container * pjsip_uids_onhold
Definition: chan_pjsip.c:1107
#define ao2_find(container, arg, flags)
Definition: astobj2.h:1756
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
void ast_copy_string(char *dst, const char *src, size_t size)
Size-limited null-terminating string copy.
Definition: strings.h:401
#define ao2_link(container, obj)
Definition: astobj2.h:1549

◆ chan_pjsip_answer()

static int chan_pjsip_answer ( struct ast_channel ast)
static

Function called by core when we should answer a PJSIP session.

Definition at line 726 of file chan_pjsip.c.

References answer(), ao2_bump, ao2_ref, ast_channel_lock, ast_channel_name(), ast_channel_tech_pvt(), ast_channel_unlock, ast_log, ast_setstate(), ast_sip_push_task_wait_serializer(), AST_STATE_UP, ast_trace_get_indent, ast_sip_session::channel, answer_data::indent, LOG_ERROR, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, ast_sip_session::serializer, rtp_direct_media_data::session, ast_sip_channel_pvt::session, and answer_data::session.

Referenced by chan_pjsip_pvt_dtor().

727 {
729  struct ast_sip_session *session;
730  struct answer_data ans_data = { 0, };
731  int res;
732  SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
733 
734  if (ast_channel_state(ast) == AST_STATE_UP) {
735  SCOPE_EXIT_RTN_VALUE(0, "Already up\n");
736  return 0;
737  }
738 
740  session = ao2_bump(channel->session);
741 
742  /* the answer task needs to be pushed synchronously otherwise a race condition
743  can occur between this thread and bridging (specifically when native bridging
744  attempts to do direct media) */
745  ast_channel_unlock(ast);
746  ans_data.session = session;
747  ans_data.indent = ast_trace_get_indent();
748  res = ast_sip_push_task_wait_serializer(session->serializer, answer, &ans_data);
749  if (res) {
750  if (res == -1) {
751  ast_log(LOG_ERROR,"Cannot answer '%s': Unable to push answer task to the threadpool.\n",
752  ast_channel_name(session->channel));
753  }
754  ao2_ref(session, -1);
755  ast_channel_lock(ast);
756  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
757  }
758  ao2_ref(session, -1);
759  ast_channel_lock(ast);
760 
762 }
#define ast_channel_lock(chan)
Definition: channel.h:2890
void * ast_channel_tech_pvt(const struct ast_channel *chan)
ast_channel_state
ast_channel states
Definition: channelstate.h:35
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
#define ast_trace_get_indent()
Definition: logger.h:879
A structure describing a SIP session.
#define ao2_bump(obj)
Definition: astobj2.h:491
#define ast_log
Definition: astobj2.c:42
unsigned long indent
Definition: chan_pjsip.c:679
static struct ast_mansession session
int ast_sip_push_task_wait_serializer(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Push a task to the serializer and wait for it to complete.
Definition: res_pjsip.c:5103
#define ao2_ref(o, delta)
Definition: astobj2.h:464
struct ast_channel * channel
static int answer(void *data)
Definition: chan_pjsip.c:682
struct ast_taskprocessor * serializer
#define LOG_ERROR
Definition: logger.h:285
#define ast_channel_unlock(chan)
Definition: channel.h:2891
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
const char * ast_channel_name(const struct ast_channel *chan)
int ast_setstate(struct ast_channel *chan, enum ast_channel_state)
Change the state of a channel.
Definition: channel.c:7349
struct ast_sip_session * session
Definition: chan_pjsip.c:678

◆ chan_pjsip_call()

static int chan_pjsip_call ( struct ast_channel ast,
const char *  dest,
int  timeout 
)
static

Function called by core to actually start calling a remote party.

Definition at line 2374 of file chan_pjsip.c.

References ao2_cleanup, ao2_ref, ast_channel_tech_pvt(), ast_log, ast_sip_push_task(), ast_sip_session_get_name(), ast_str_tmp, ast_stream_topology_to_str(), call(), LOG_WARNING, ast_sip_session::pending_media_state, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, ast_sip_session::serializer, ast_sip_channel_pvt::session, and ast_sip_session_media_state::topology.

Referenced by chan_pjsip_pvt_dtor().

2375 {
2377  SCOPE_ENTER(1, "%s Topology: %s\n", ast_sip_session_get_name(channel->session),
2379 
2380  ao2_ref(channel, +1);
2381  if (ast_sip_push_task(channel->session->serializer, call, channel)) {
2382  ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest);
2383  ao2_cleanup(channel);
2384  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
2385  }
2386 
2387  SCOPE_EXIT_RTN_VALUE(0, "'call' task pushed\n");
2388 }
struct ast_sip_session_media_state * pending_media_state
void * ast_channel_tech_pvt(const struct ast_channel *chan)
#define LOG_WARNING
Definition: logger.h:274
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
static int call(void *data)
Definition: chan_pjsip.c:2347
#define ast_str_tmp(init_len, __expr)
Definition: strings.h:1136
#define ast_log
Definition: astobj2.c:42
const char * ast_stream_topology_to_str(const struct ast_stream_topology *topology, struct ast_str **buf)
Get a string representing the topology for debugging/display purposes.
Definition: stream.c:936
#define ao2_ref(o, delta)
Definition: astobj2.h:464
struct ast_taskprocessor * serializer
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
int ast_sip_push_task(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Pushes a task to SIP servants.
Definition: res_pjsip.c:5023
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
struct ast_stream_topology * topology
The media stream topology.

◆ chan_pjsip_cng_tone_detected()

static struct ast_frame* chan_pjsip_cng_tone_detected ( struct ast_channel ast,
struct ast_sip_session session,
struct ast_frame f 
)
static

Internal helper function called when CNG tone is detected.

Definition at line 765 of file chan_pjsip.c.

References ast_async_goto(), ast_channel_caller(), ast_channel_context(), ast_channel_exten(), ast_channel_lock, ast_channel_macrocontext(), ast_channel_name(), ast_channel_unlock, ast_dsp_free(), ast_dsp_get_features(), ast_dsp_set_features(), ast_exists_extension(), ast_frfree, ast_log, ast_null_frame, ast_verb, ast_sip_session::dsp, DSP_FEATURE_FAX_DETECT, exists(), LOG_ERROR, LOG_NOTICE, NULL, pbx_builtin_setvar_helper(), S_COR, and S_OR.

Referenced by chan_pjsip_read_stream().

767 {
768  const char *target_context;
769  int exists;
770  int dsp_features;
771 
772  dsp_features = ast_dsp_get_features(session->dsp);
773  dsp_features &= ~DSP_FEATURE_FAX_DETECT;
774  if (dsp_features) {
775  ast_dsp_set_features(session->dsp, dsp_features);
776  } else {
777  ast_dsp_free(session->dsp);
778  session->dsp = NULL;
779  }
780 
781  /* If already executing in the fax extension don't do anything */
782  if (!strcmp(ast_channel_exten(ast), "fax")) {
783  return f;
784  }
785 
786  target_context = S_OR(ast_channel_macrocontext(ast), ast_channel_context(ast));
787 
788  /*
789  * We need to unlock the channel here because ast_exists_extension has the
790  * potential to start and stop an autoservice on the channel. Such action
791  * is prone to deadlock if the channel is locked.
792  *
793  * ast_async_goto() has its own restriction on not holding the channel lock.
794  */
795  ast_channel_unlock(ast);
796  ast_frfree(f);
797  f = &ast_null_frame;
798  exists = ast_exists_extension(ast, target_context, "fax", 1,
799  S_COR(ast_channel_caller(ast)->id.number.valid,
800  ast_channel_caller(ast)->id.number.str, NULL));
801  if (exists) {
802  ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
803  ast_channel_name(ast));
804  pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
805  if (ast_async_goto(ast, target_context, "fax", 1)) {
806  ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
807  ast_channel_name(ast), target_context);
808  }
809  } else {
810  ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
811  ast_channel_name(ast), target_context);
812  }
813 
814  /* It's possible for a masquerade to have occurred when doing the ast_async_goto resulting in
815  * the channel on the session having changed. Since we need to return with the original channel
816  * locked we lock the channel that was passed in and not session->channel.
817  */
818  ast_channel_lock(ast);
819 
820  return f;
821 }
struct ast_party_caller * ast_channel_caller(struct ast_channel *chan)
#define ast_channel_lock(chan)
Definition: channel.h:2890
static int exists(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
Definition: func_logic.c:124
void ast_dsp_free(struct ast_dsp *dsp)
Definition: dsp.c:1744
#define NULL
Definition: resample.c:96
#define ast_verb(level,...)
Definition: logger.h:455
Number structure.
Definition: app_followme.c:154
struct ast_dsp * dsp
#define ast_log
Definition: astobj2.c:42
int ast_dsp_get_features(struct ast_dsp *dsp)
Get features.
Definition: dsp.c:1738
#define S_COR(a, b, c)
returns the equivalent of logic or for strings, with an additional boolean check: second one if not e...
Definition: strings.h:85
#define DSP_FEATURE_FAX_DETECT
Definition: dsp.h:29
int ast_exists_extension(struct ast_channel *c, const char *context, const char *exten, int priority, const char *callerid)
Determine whether an extension exists.
Definition: pbx.c:4179
const char * ast_channel_exten(const struct ast_channel *chan)
#define LOG_ERROR
Definition: logger.h:285
#define LOG_NOTICE
Definition: logger.h:263
#define ast_channel_unlock(chan)
Definition: channel.h:2891
void ast_dsp_set_features(struct ast_dsp *dsp, int features)
Select feature set.
Definition: dsp.c:1729
int pbx_builtin_setvar_helper(struct ast_channel *chan, const char *name, const char *value)
Add a variable to the channel variable stack, removing the most recently set value for the same name...
struct ast_frame ast_null_frame
Definition: main/frame.c:79
#define S_OR(a, b)
returns the equivalent of logic or for strings: first one if not empty, otherwise second one...
Definition: strings.h:79
const char * ast_channel_name(const struct ast_channel *chan)
int ast_async_goto(struct ast_channel *chan, const char *context, const char *exten, int priority)
Set the channel to next execute the specified dialplan location.
Definition: pbx.c:7011
#define ast_frfree(fr)
const char * ast_channel_context(const struct ast_channel *chan)
const char * ast_channel_macrocontext(const struct ast_channel *chan)

◆ chan_pjsip_devicestate()

static int chan_pjsip_devicestate ( const char *  data)
static

Function called to get the device state of an endpoint.

Definition at line 1174 of file chan_pjsip.c.

References ao2_cleanup, ao2_ref, ast_channel_snapshot_get_latest(), AST_DEVICE_BUSY, AST_DEVICE_INVALID, AST_DEVICE_NOT_INUSE, AST_DEVICE_ONHOLD, AST_DEVICE_UNAVAILABLE, AST_DEVICE_UNKNOWN, ast_devstate_aggregate_add(), ast_devstate_aggregate_init(), ast_devstate_aggregate_result(), ast_endpoint_get_resource(), ast_endpoint_get_tech(), ast_endpoint_latest_snapshot(), AST_ENDPOINT_OFFLINE, AST_ENDPOINT_ONLINE, ast_sip_get_sorcery(), ast_sorcery_retrieve_by_id(), AST_STATE_BUSY, ast_state_chan2dev(), AST_STATE_RING, AST_STATE_UP, ast_channel_snapshot::base, chan_pjsip_get_hold(), ast_devstate_aggregate::inuse, NULL, RAII_VAR, ast_channel_snapshot::state, state, and ast_channel_snapshot_base::uniqueid.

Referenced by chan_pjsip_pvt_dtor().

1175 {
1176  RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
1178  RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
1179  struct ast_devstate_aggregate aggregate;
1180  int num, inuse = 0;
1181 
1182  if (!endpoint) {
1183  return AST_DEVICE_INVALID;
1184  }
1185 
1186  endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
1187  ast_endpoint_get_resource(endpoint->persistent));
1188 
1189  if (!endpoint_snapshot) {
1190  return AST_DEVICE_INVALID;
1191  }
1192 
1193  if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
1194  state = AST_DEVICE_UNAVAILABLE;
1195  } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
1196  state = AST_DEVICE_NOT_INUSE;
1197  }
1198 
1199  if (!endpoint_snapshot->num_channels) {
1200  return state;
1201  }
1202 
1203  ast_devstate_aggregate_init(&aggregate);
1204 
1205  for (num = 0; num < endpoint_snapshot->num_channels; num++) {
1206  struct ast_channel_snapshot *snapshot;
1207 
1208  snapshot = ast_channel_snapshot_get_latest(endpoint_snapshot->channel_ids[num]);
1209  if (!snapshot) {
1210  continue;
1211  }
1212 
1213  if (chan_pjsip_get_hold(snapshot->base->uniqueid)) {
1215  } else {
1216  ast_devstate_aggregate_add(&aggregate, ast_state_chan2dev(snapshot->state));
1217  }
1218 
1219  if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
1220  (snapshot->state == AST_STATE_BUSY)) {
1221  inuse++;
1222  }
1223 
1224  ao2_ref(snapshot, -1);
1225  }
1226 
1227  if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
1228  state = AST_DEVICE_BUSY;
1229  } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
1230  state = ast_devstate_aggregate_result(&aggregate);
1231  }
1232 
1233  return state;
1234 }
enum sip_cc_notify_state state
Definition: chan_sip.c:957
ast_device_state
Device States.
Definition: devicestate.h:52
struct ast_channel_snapshot_base * base
const char * ast_endpoint_get_tech(const struct ast_endpoint *endpoint)
Gets the technology of the given endpoint.
struct ast_endpoint_snapshot * ast_endpoint_latest_snapshot(const char *tech, const char *resource)
Retrieve the most recent snapshot for the endpoint with the given name.
Structure representing a snapshot of channel state.
const ast_string_field uniqueid
#define NULL
Definition: resample.c:96
const char * ast_endpoint_get_resource(const struct ast_endpoint *endpoint)
Gets the resource name of the given endpoint.
enum ast_device_state ast_devstate_aggregate_result(struct ast_devstate_aggregate *agg)
Get the aggregate device state result.
Definition: devicestate.c:663
void * ast_sorcery_retrieve_by_id(const struct ast_sorcery *sorcery, const char *type, const char *id)
Retrieve an object using its unique identifier.
Definition: sorcery.c:1850
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition: utils.h:851
#define ao2_ref(o, delta)
Definition: astobj2.h:464
enum ast_device_state ast_state_chan2dev(enum ast_channel_state chanstate)
Convert channel state to devicestate.
Definition: devicestate.c:242
An entity with which Asterisk communicates.
Definition: res_pjsip.h:812
A snapshot of an endpoint&#39;s state.
void ast_devstate_aggregate_add(struct ast_devstate_aggregate *agg, enum ast_device_state state)
Add a device state to the aggregate device state.
Definition: devicestate.c:636
static int chan_pjsip_get_hold(const char *chan_uid)
Determine whether a channel ID is in the list of PJSIP channels on hold.
Definition: chan_pjsip.c:1161
void ast_devstate_aggregate_init(struct ast_devstate_aggregate *agg)
Initialize aggregate device state.
Definition: devicestate.c:630
enum ast_channel_state state
struct ast_channel_snapshot * ast_channel_snapshot_get_latest(const char *uniqueid)
Obtain the latest ast_channel_snapshot from the Stasis Message Bus API cache. This is an ao2 object...
struct ast_sorcery * ast_sip_get_sorcery(void)
Get a pointer to the SIP sorcery structure.
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
You shouldn&#39;t care about the contents of this struct.
Definition: devicestate.h:230

◆ chan_pjsip_digit_begin()

static int chan_pjsip_digit_begin ( struct ast_channel ast,
char  digit 
)
static

Function called by core to start a DTMF digit.

Definition at line 2139 of file chan_pjsip.c.

References ast_sip_session::active_media_state, ast_channel_tech_pvt(), AST_MEDIA_TYPE_AUDIO, AST_RTP_DTMF_MODE_INBAND, AST_RTP_DTMF_MODE_NONE, ast_rtp_instance_dtmf_begin(), ast_rtp_instance_dtmf_mode_get(), AST_SIP_DTMF_AUTO, AST_SIP_DTMF_AUTO_INFO, AST_SIP_DTMF_INBAND, AST_SIP_DTMF_NONE, AST_SIP_DTMF_RFC_4733, ast_sip_session_media_state::default_session, ast_sip_session::dtmf, ast_sip_session_media::rtp, and ast_sip_channel_pvt::session.

Referenced by chan_pjsip_pvt_dtor().

2140 {
2142  struct ast_sip_session_media *media;
2143 
2145 
2146  switch (channel->session->dtmf) {
2147  case AST_SIP_DTMF_RFC_4733:
2148  if (!media || !media->rtp) {
2149  return 0;
2150  }
2151 
2153  break;
2154  case AST_SIP_DTMF_AUTO:
2155  if (!media || !media->rtp) {
2156  return 0;
2157  }
2158 
2160  return -1;
2161  }
2162 
2164  break;
2166  if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) {
2167  return 0;
2168  }
2170  break;
2171  case AST_SIP_DTMF_NONE:
2172  break;
2173  case AST_SIP_DTMF_INBAND:
2174  return -1;
2175  default:
2176  break;
2177  }
2178 
2179  return 0;
2180 }
char digit
void * ast_channel_tech_pvt(const struct ast_channel *chan)
struct ast_sip_session_media * default_session[AST_MEDIA_TYPE_END]
Default media sessions for each type.
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
struct ast_sip_session_media_state * active_media_state
enum ast_sip_dtmf_mode dtmf
int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
Begin sending a DTMF digit.
Definition: rtp_engine.c:2081
A structure containing SIP session media information.
struct ast_rtp_instance * rtp
RTP instance itself.
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
Get the DTMF mode of an RTP instance.
Definition: rtp_engine.c:2137

◆ chan_pjsip_digit_end()

static int chan_pjsip_digit_end ( struct ast_channel ast,
char  digit,
unsigned int  duration 
)
static

Function called by core to stop a DTMF digit.

Definition at line 2251 of file chan_pjsip.c.

References ast_sip_session::active_media_state, ao2_cleanup, ast_channel_name(), ast_channel_tech_pvt(), ast_debug, ast_log, AST_MEDIA_TYPE_AUDIO, AST_RTP_DTMF_MODE_INBAND, AST_RTP_DTMF_MODE_NONE, ast_rtp_instance_dtmf_end_with_duration(), ast_rtp_instance_dtmf_mode_get(), AST_SIP_DTMF_AUTO, AST_SIP_DTMF_AUTO_INFO, AST_SIP_DTMF_INBAND, AST_SIP_DTMF_INFO, AST_SIP_DTMF_NONE, AST_SIP_DTMF_RFC_4733, ast_sip_push_task(), ast_sip_session_media_state::default_session, ast_sip_session::dtmf, info_dtmf_data_alloc(), LOG_WARNING, ast_sip_session_media::rtp, ast_sip_session::serializer, ast_sip_channel_pvt::session, and transmit_info_dtmf().

Referenced by chan_pjsip_pvt_dtor().

2252 {
2254  struct ast_sip_session_media *media;
2255 
2256  if (!channel || !channel->session) {
2257  /* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */
2258  ast_debug(3, "Channel %s disappeared while calling digit_end\n", ast_channel_name(ast));
2259  return -1;
2260  }
2261 
2263 
2264  switch (channel->session->dtmf) {
2266  {
2267  if (!media || !media->rtp) {
2268  return 0;
2269  }
2270 
2272  ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
2274  break;
2275  }
2276  /* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */
2277  ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 NOT negotiated using INFO instead.\n", ast_channel_name(ast));
2278  }
2279 
2280  case AST_SIP_DTMF_INFO:
2281  {
2282  struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
2283 
2284  if (!dtmf_data) {
2285  return -1;
2286  }
2287 
2288  if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
2289  ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
2290  ao2_cleanup(dtmf_data);
2291  return -1;
2292  }
2293  break;
2294  }
2295  case AST_SIP_DTMF_RFC_4733:
2296  if (!media || !media->rtp) {
2297  return 0;
2298  }
2299 
2301  break;
2302  case AST_SIP_DTMF_AUTO:
2303  if (!media || !media->rtp) {
2304  return 0;
2305  }
2306 
2308  return -1;
2309  }
2310 
2312  break;
2313  case AST_SIP_DTMF_NONE:
2314  break;
2315  case AST_SIP_DTMF_INBAND:
2316  return -1;
2317  }
2318 
2319  return 0;
2320 }
char digit
void * ast_channel_tech_pvt(const struct ast_channel *chan)
#define LOG_WARNING
Definition: logger.h:274
struct ast_sip_session_media * default_session[AST_MEDIA_TYPE_END]
Default media sessions for each type.
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
unsigned int duration
Definition: chan_pjsip.c:2185
Definition: muted.c:95
struct ast_sip_session_media_state * active_media_state
#define ast_debug(level,...)
Log a DEBUG message.
Definition: logger.h:444
#define ast_log
Definition: astobj2.c:42
static int transmit_info_dtmf(void *data)
Definition: chan_pjsip.c:2207
struct ast_taskprocessor * serializer
int ast_sip_push_task(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Pushes a task to SIP servants.
Definition: res_pjsip.c:5023
enum ast_sip_dtmf_mode dtmf
static struct info_dtmf_data * info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
Definition: chan_pjsip.c:2194
A structure containing SIP session media information.
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
struct ast_rtp_instance * rtp
RTP instance itself.
const char * ast_channel_name(const struct ast_channel *chan)
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
Get the DTMF mode of an RTP instance.
Definition: rtp_engine.c:2137
int ast_rtp_instance_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
Definition: rtp_engine.c:2109

◆ chan_pjsip_fixup()

static int chan_pjsip_fixup ( struct ast_channel oldchan,
struct ast_channel newchan 
)
static

Function called by core to change the underlying owner channel.

Definition at line 1045 of file chan_pjsip.c.

References ast_channel_tech_pvt(), ast_channel_uniqueid(), ast_sip_session::channel, ast_sip_channel_pvt::session, and set_channel_on_rtp_instance().

Referenced by chan_pjsip_pvt_dtor().

1046 {
1048 
1049  if (channel->session->channel != oldchan) {
1050  return -1;
1051  }
1052 
1053  /*
1054  * The masquerade has suspended the channel's session
1055  * serializer so we can safely change it outside of
1056  * the serializer thread.
1057  */
1058  channel->session->channel = newchan;
1059 
1061 
1062  return 0;
1063 }
void * ast_channel_tech_pvt(const struct ast_channel *chan)
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
static void set_channel_on_rtp_instance(const struct ast_sip_session *session, const char *channel_id)
Definition: chan_pjsip.c:494
struct ast_channel * channel
const char * ast_channel_uniqueid(const struct ast_channel *chan)

◆ chan_pjsip_get_codec()

static void chan_pjsip_get_codec ( struct ast_channel chan,
struct ast_format_cap result 
)
static

Function called by RTP engine to get peer capabilities.

Definition at line 244 of file chan_pjsip.c.

References ast_channel_name(), ast_channel_nativeformats(), ast_format_cap_append_from_cap(), ast_format_cap_get_names(), AST_FORMAT_CAP_NAMES_LEN, AST_MEDIA_TYPE_UNKNOWN, ast_str_tmp, SCOPE_ENTER, and SCOPE_EXIT_RTN.

245 {
246  SCOPE_ENTER(1, "%s Native formats %s\n", ast_channel_name(chan),
249  SCOPE_EXIT_RTN();
250 }
#define AST_FORMAT_CAP_NAMES_LEN
Definition: format_cap.h:326
#define ast_str_tmp(init_len, __expr)
Definition: strings.h:1136
const char * ast_format_cap_get_names(const struct ast_format_cap *cap, struct ast_str **buf)
Get the names of codecs of a set of formats.
Definition: format_cap.c:736
#define SCOPE_EXIT_RTN(...)
Definition: logger.h:900
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
struct ast_format_cap * ast_channel_nativeformats(const struct ast_channel *chan)
const char * ast_channel_name(const struct ast_channel *chan)
int ast_format_cap_append_from_cap(struct ast_format_cap *dst, const struct ast_format_cap *src, enum ast_media_type type)
Append the formats of provided type in src to dst.
Definition: format_cap.c:269

◆ chan_pjsip_get_hold()

static int chan_pjsip_get_hold ( const char *  chan_uid)
static

Determine whether a channel ID is in the list of PJSIP channels on hold.

Parameters
chan_uid- Channel being checked
Return values
0The channel is not in the hold list
1The channel is in the hold list

Definition at line 1161 of file chan_pjsip.c.

References ao2_cleanup, ao2_find, NULL, OBJ_SEARCH_KEY, and RAII_VAR.

Referenced by chan_pjsip_devicestate().

1162 {
1163  RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
1164 
1165  hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
1166  if (!hold_uid) {
1167  return 0;
1168  }
1169 
1170  return 1;
1171 }
The arg parameter is a search key, but is not an object.
Definition: astobj2.h:1105
#define NULL
Definition: resample.c:96
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition: utils.h:851
static struct ao2_container * pjsip_uids_onhold
Definition: chan_pjsip.c:1107
#define ao2_find(container, arg, flags)
Definition: astobj2.h:1756
#define ao2_cleanup(obj)
Definition: astobj2.h:1958

◆ chan_pjsip_get_rtp_peer()

static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer ( struct ast_channel chan,
struct ast_rtp_instance **  instance 
)
static

Function called by RTP engine to get local audio RTP peer.

Definition at line 171 of file chan_pjsip.c.

References ast_sip_session::active_media_state, ao2_ref, ast_assert, ast_channel_tech_pvt(), AST_MEDIA_TYPE_AUDIO, AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_LOCAL, AST_RTP_GLUE_RESULT_REMOTE, AST_SIP_MEDIA_ENCRYPT_NONE, ast_sip_session_get_datastore(), ast_sip_session_media_state::default_session, ast_sip_endpoint_media_configuration::direct_media, ast_sip_direct_media_configuration::enabled, ast_sip_media_rtp_configuration::encryption, ast_sip_session::endpoint, ast_sip_endpoint::media, NULL, ast_sip_session_media::rtp, ast_sip_endpoint_media_configuration::rtp, and ast_sip_channel_pvt::session.

172 {
174  struct ast_sip_endpoint *endpoint;
175  struct ast_datastore *datastore;
176  struct ast_sip_session_media *media;
177 
178  if (!channel || !channel->session) {
180  }
181 
182  /* XXX Getting the first RTP instance for direct media related stuff seems just
183  * absolutely wrong. But the native RTP bridge knows no other method than single-stream
184  * for direct media. So this is the best we can do.
185  */
187  if (!media || !media->rtp) {
189  }
190 
191  datastore = ast_sip_session_get_datastore(channel->session, "t38");
192  if (datastore) {
193  ao2_ref(datastore, -1);
195  }
196 
197  endpoint = channel->session->endpoint;
198 
199  *instance = media->rtp;
200  ao2_ref(*instance, +1);
201 
202  ast_assert(endpoint != NULL);
203  if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
205  }
206 
207  if (endpoint->media.direct_media.enabled) {
209  }
210 
212 }
struct ast_sip_endpoint * endpoint
void * ast_channel_tech_pvt(const struct ast_channel *chan)
struct ast_sip_session_media * default_session[AST_MEDIA_TYPE_END]
Default media sessions for each type.
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
#define ast_assert(a)
Definition: utils.h:650
Definition: muted.c:95
Structure for a data store object.
Definition: datastore.h:68
#define NULL
Definition: resample.c:96
struct ast_sip_direct_media_configuration direct_media
Definition: res_pjsip.h:766
struct ast_sip_session_media_state * active_media_state
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
struct ast_datastore * ast_sip_session_get_datastore(struct ast_sip_session *session, const char *name)
Retrieve a session datastore.
struct ast_sip_media_rtp_configuration rtp
Definition: res_pjsip.h:764
#define ao2_ref(o, delta)
Definition: astobj2.h:464
An entity with which Asterisk communicates.
Definition: res_pjsip.h:812
enum ast_sip_session_media_encryption encryption
Definition: res_pjsip.h:711
A structure containing SIP session media information.
struct ast_rtp_instance * rtp
RTP instance itself.

◆ chan_pjsip_get_uniqueid()

static const char * chan_pjsip_get_uniqueid ( struct ast_channel ast)
static

Definition at line 1278 of file chan_pjsip.c.

References ast_channel_tech_pvt(), ast_copy_pj_str(), ast_threadstorage_get(), ast_sip_session::inv_session, ast_sip_channel_pvt::session, UNIQUEID_BUFSIZE, and uniqueid_threadbuf.

Referenced by chan_pjsip_pvt_dtor().

1279 {
1282 
1283  if (!uniqueid) {
1284  return "";
1285  }
1286 
1287  ast_copy_pj_str(uniqueid, &channel->session->inv_session->dlg->call_id->id, UNIQUEID_BUFSIZE);
1288 
1289  return uniqueid;
1290 }
void * ast_threadstorage_get(struct ast_threadstorage *ts, size_t init_size)
Retrieve thread storage.
void * ast_channel_tech_pvt(const struct ast_channel *chan)
#define UNIQUEID_BUFSIZE
Definition: chan_pjsip.c:76
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size)
Copy a pj_str_t into a standard character buffer.
Definition: res_pjsip.c:5125
struct pjsip_inv_session * inv_session
static struct ast_threadstorage uniqueid_threadbuf
Definition: chan_pjsip.c:75

◆ chan_pjsip_get_vrtp_peer()

static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer ( struct ast_channel chan,
struct ast_rtp_instance **  instance 
)
static

Function called by RTP engine to get local video RTP peer.

Definition at line 215 of file chan_pjsip.c.

References ast_sip_session::active_media_state, ao2_ref, ast_assert, ast_channel_tech_pvt(), AST_MEDIA_TYPE_VIDEO, AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_LOCAL, AST_SIP_MEDIA_ENCRYPT_NONE, ast_sip_session_media_state::default_session, ast_sip_media_rtp_configuration::encryption, ast_sip_session::endpoint, ast_sip_endpoint::media, NULL, ast_sip_session_media::rtp, ast_sip_endpoint_media_configuration::rtp, and ast_sip_channel_pvt::session.

216 {
218  struct ast_sip_endpoint *endpoint;
219  struct ast_sip_session_media *media;
220 
221  if (!channel || !channel->session) {
223  }
224 
226  if (!media || !media->rtp) {
228  }
229 
230  endpoint = channel->session->endpoint;
231 
232  *instance = media->rtp;
233  ao2_ref(*instance, +1);
234 
235  ast_assert(endpoint != NULL);
236  if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
238  }
239 
241 }
struct ast_sip_endpoint * endpoint
void * ast_channel_tech_pvt(const struct ast_channel *chan)
struct ast_sip_session_media * default_session[AST_MEDIA_TYPE_END]
Default media sessions for each type.
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
#define ast_assert(a)
Definition: utils.h:650
Definition: muted.c:95
#define NULL
Definition: resample.c:96
struct ast_sip_session_media_state * active_media_state
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
struct ast_sip_media_rtp_configuration rtp
Definition: res_pjsip.h:764
#define ao2_ref(o, delta)
Definition: astobj2.h:464
An entity with which Asterisk communicates.
Definition: res_pjsip.h:812
enum ast_sip_session_media_encryption encryption
Definition: res_pjsip.h:711
A structure containing SIP session media information.
struct ast_rtp_instance * rtp
RTP instance itself.

◆ chan_pjsip_hangup()

static int chan_pjsip_hangup ( struct ast_channel ast)
static

Function called by core to hang up a PJSIP session.

Definition at line 2505 of file chan_pjsip.c.

References ao2_cleanup, ast_channel_hangupcause(), ast_channel_name(), ast_channel_tech_pvt(), ast_log, ast_sip_push_task(), ast_sip_session::channel, clear_session_and_channel(), hangup(), hangup_cause2sip(), hangup_data_alloc(), LOG_WARNING, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, ast_sip_session::serializer, and ast_sip_channel_pvt::session.

Referenced by chan_pjsip_pvt_dtor().

2506 {
2508  int cause;
2509  struct hangup_data *h_data;
2510  SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
2511 
2512  if (!channel || !channel->session) {
2513  SCOPE_EXIT_RTN_VALUE(-1, "No channel or session\n");
2514  }
2515 
2517  h_data = hangup_data_alloc(cause, ast);
2518 
2519  if (!h_data) {
2520  goto failure;
2521  }
2522 
2523  if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
2524  ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
2525  goto failure;
2526  }
2527 
2528  SCOPE_EXIT_RTN_VALUE(0, "Cause: %d\n", cause);
2529 
2530 failure:
2531  /* Go ahead and do our cleanup of the session and channel even if we're not going
2532  * to be able to send our SIP request/response
2533  */
2534  clear_session_and_channel(channel->session, ast);
2535  ao2_cleanup(channel);
2536  ao2_cleanup(h_data);
2537 
2538  SCOPE_EXIT_RTN_VALUE(-1, "Cause: %d\n", cause);
2539 }
void * ast_channel_tech_pvt(const struct ast_channel *chan)
#define LOG_WARNING
Definition: logger.h:274
static int hangup_cause2sip(int cause)
Internal function which translates from Asterisk cause codes to SIP response codes.
Definition: chan_pjsip.c:2391
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
static int hangup(void *data)
Definition: chan_pjsip.c:2472
#define ast_log
Definition: astobj2.c:42
struct ast_channel * channel
struct ast_taskprocessor * serializer
int ast_sip_push_task(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Pushes a task to SIP servants.
Definition: res_pjsip.c:5023
static struct hangup_data * hangup_data_alloc(int cause, struct ast_channel *chan)
Definition: chan_pjsip.c:2450
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
int ast_channel_hangupcause(const struct ast_channel *chan)
const char * ast_channel_name(const struct ast_channel *chan)
static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast)
Clear a channel from a session along with its PVT.
Definition: chan_pjsip.c:2465

◆ chan_pjsip_incoming_ack()

static int chan_pjsip_incoming_ack ( struct ast_sip_session session,
struct pjsip_rx_data *  rdata 
)
static

Definition at line 3221 of file chan_pjsip.c.

References AST_CONTROL_SRCCHANGE, ast_queue_control(), ast_sip_session_get_name(), ast_trace, ast_sip_session::channel, ast_sip_endpoint_media_configuration::direct_media, ast_sip_direct_media_configuration::enabled, ast_sip_session::endpoint, ast_sip_endpoint::media, SCOPE_ENTER, and SCOPE_EXIT_RTN_VALUE.

3222 {
3223  SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
3224 
3225  if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
3226  if (session->endpoint->media.direct_media.enabled && session->channel) {
3227  ast_trace(-1, "%s: Queueing SRCCHANGE\n", ast_sip_session_get_name(session));
3229  }
3230  }
3231  SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
3232 }
struct ast_sip_endpoint * endpoint
int ast_queue_control(struct ast_channel *chan, enum ast_control_frame_type control)
Queue a control frame without payload.
Definition: channel.c:1227
#define ast_trace(level,...)
Definition: logger.h:876
struct ast_sip_direct_media_configuration direct_media
Definition: res_pjsip.h:766
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
struct ast_channel * channel
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904

◆ chan_pjsip_incoming_request()

static int chan_pjsip_incoming_request ( struct ast_sip_session session,
struct pjsip_rx_data *  rdata 
)
static

Function called when a request is received on the session.

Definition at line 2996 of file chan_pjsip.c.

References ao2_cleanup, ast_calloc, ast_sip_session_add_datastore(), ast_sip_session_alloc_datastore(), ast_sip_session_get_name(), ast_sip_session_send_response(), ast_sip_session_terminate(), AST_STATE_RING, chan_pjsip_new(), ast_sip_session::channel, ast_sip_session::defer_terminate, ast_sip_session::exten, ast_sip_session::inv_session, transport_info_data::local_addr, LOG_ERROR, NULL, RAII_VAR, transport_info_data::remote_addr, SCOPE_ENTER, SCOPE_EXIT_LOG_RTN_VALUE, SCOPE_EXIT_RTN_VALUE, and set_sipdomain_variable().

2997 {
2998  RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
2999  struct transport_info_data *transport_data;
3000  pjsip_tx_data *packet = NULL;
3001  SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
3002 
3003  if (session->channel) {
3004  SCOPE_EXIT_RTN_VALUE(0, "%s: No channel\n", ast_sip_session_get_name(session));
3005  }
3006 
3007  /* Check for a to-tag to determine if this is a reinvite */
3008  if (rdata->msg_info.to->tag.slen) {
3009  /* Weird case. We've received a reinvite but we don't have a channel. The most
3010  * typical case for this happening is that a blind transfer fails, and so the
3011  * transferer attempts to reinvite himself back into the call. We already got
3012  * rid of that channel, and the other side of the call is unrecoverable.
3013  *
3014  * We treat this as a failure, so our best bet is to just hang this call
3015  * up and not create a new channel. Clearing defer_terminate here ensures that
3016  * calling ast_sip_session_terminate() can result in a BYE being sent ASAP.
3017  */
3018  session->defer_terminate = 0;
3019  ast_sip_session_terminate(session, 400);
3020  SCOPE_EXIT_RTN_VALUE(-1, "%s: We have a To tag but no channel. Terminating session\n", ast_sip_session_get_name(session));
3021  }
3022 
3023  datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
3024  if (!datastore) {
3025  SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info datastore\n", ast_sip_session_get_name(session));
3026  }
3027 
3028  transport_data = ast_calloc(1, sizeof(*transport_data));
3029  if (!transport_data) {
3030  SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info\n", ast_sip_session_get_name(session));
3031  }
3032  pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
3033  pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
3034  datastore->data = transport_data;
3035  ast_sip_session_add_datastore(session, datastore);
3036 
3037  if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL, NULL))) {
3038  if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS
3039  && packet) {
3040  ast_sip_session_send_response(session, packet);
3041  }
3042 
3043  SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n",
3044  ast_sip_session_get_name(session));
3045  }
3046 
3047  set_sipdomain_variable(session);
3048 
3049  /* channel gets created on incoming request, but we wait to call start
3050  so other supplements have a chance to run */
3051  SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
3052 }
#define SCOPE_EXIT_LOG_RTN_VALUE(__value, __log_level,...)
Definition: logger.h:927
pj_sockaddr local_addr
Our address that received the request.
Definition: chan_pjsip.h:34
void ast_sip_session_terminate(struct ast_sip_session *session, int response)
Terminate a session and, if possible, send the provided response code.
unsigned int defer_terminate
static struct ast_channel * chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name)
Function called to create a new PJSIP Asterisk channel.
Definition: chan_pjsip.c:547
static struct ast_datastore_info transport_info
Datastore used to store local/remote addresses for the INVITE request that created the PJSIP channel...
Definition: chan_pjsip.c:261
Structure for a data store object.
Definition: datastore.h:68
char exten[AST_MAX_EXTENSION]
#define NULL
Definition: resample.c:96
struct pjsip_inv_session * inv_session
Transport information stored in transport_info datastore.
Definition: chan_pjsip.h:30
struct ast_datastore * ast_sip_session_alloc_datastore(const struct ast_datastore_info *info, const char *uid)
Alternative for ast_datastore_alloc()
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition: utils.h:851
struct ast_channel * channel
void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
Send a SIP response.
pj_sockaddr remote_addr
The address that sent the request.
Definition: chan_pjsip.h:32
#define LOG_ERROR
Definition: logger.h:285
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
#define ast_calloc(num, len)
A wrapper for calloc()
Definition: astmm.h:204
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
static void set_sipdomain_variable(struct ast_sip_session *session)
Definition: chan_pjsip.c:2983
int ast_sip_session_add_datastore(struct ast_sip_session *session, struct ast_datastore *datastore)
Add a datastore to a SIP session.

◆ chan_pjsip_incoming_response()

static void chan_pjsip_incoming_response ( struct ast_sip_session session,
struct pjsip_rx_data *  rdata 
)
static

Function called when a response is received on the session.

Definition at line 3175 of file chan_pjsip.c.

References ast_channel_lock, ast_channel_unlock, AST_CONTROL_ANSWER, AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, ast_queue_control(), ast_setstate(), ast_sip_session_get_name(), AST_STATE_RINGING, AST_STATE_UP, ast_trace, ast_sip_session::channel, ast_sip_session::endpoint, ast_sip_endpoint::ignore_183_without_sdp, SCOPE_ENTER, and SCOPE_EXIT_RTN.

3176 {
3177  struct pjsip_status_line status = rdata->msg_info.msg->line.status;
3178  SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
3179 
3180  if (!session->channel) {
3181  SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
3182  }
3183 
3184  switch (status.code) {
3185  case 180:
3186  ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
3188  ast_channel_lock(session->channel);
3189  if (ast_channel_state(session->channel) != AST_STATE_UP) {
3191  }
3192  ast_channel_unlock(session->channel);
3193  break;
3194  case 183:
3195  ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
3196  if (session->endpoint->ignore_183_without_sdp) {
3197  pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
3198  if (sdp && sdp->body.ptr) {
3199  ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS with SDP\n", ast_sip_session_get_name(session),
3200  (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
3202  }
3203  } else {
3204  ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS without SDP\n", ast_sip_session_get_name(session),
3205  (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
3207  }
3208  break;
3209  case 200:
3210  ast_trace(-1, "%s: Queueing ANSWER\n", ast_sip_session_get_name(session));
3212  break;
3213  default:
3214  ast_trace(-1, "%s: Not queueing anything\n", ast_sip_session_get_name(session));
3215  break;
3216  }
3217 
3218  SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
3219 }
#define ast_channel_lock(chan)
Definition: channel.h:2890
struct ast_sip_endpoint * endpoint
int ast_queue_control(struct ast_channel *chan, enum ast_control_frame_type control)
Queue a control frame without payload.
Definition: channel.c:1227
unsigned int ignore_183_without_sdp
Definition: res_pjsip.h:901
#define ast_trace(level,...)
Definition: logger.h:876
ast_channel_state
ast_channel states
Definition: channelstate.h:35
struct ast_channel * channel
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
#define SCOPE_EXIT_RTN(...)
Definition: logger.h:900
#define ast_channel_unlock(chan)
Definition: channel.h:2891
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
int ast_setstate(struct ast_channel *chan, enum ast_channel_state)
Change the state of a channel.
Definition: channel.c:7349
jack_status_t status
Definition: app_jack.c:146

◆ chan_pjsip_incoming_response_update_cause()

static void chan_pjsip_incoming_response_update_cause ( struct ast_sip_session session,
struct pjsip_rx_data *  rdata 
)
static

Function called when a response is received on the session.

Definition at line 3145 of file chan_pjsip.c.

References ast_alloca, ast_control_pvt_cause_code::ast_cause, ast_channel_hangupcause_hash_set(), AST_CHANNEL_NAME, ast_channel_name(), AST_CONTROL_PVT_CAUSE_CODE, ast_copy_string(), ast_queue_control_data(), ast_sip_session_get_name(), ast_control_pvt_cause_code::chan_name, ast_sip_session::channel, ast_control_pvt_cause_code::code, hangup_sip2cause(), SCOPE_ENTER, and SCOPE_EXIT_RTN.

3146 {
3147  struct pjsip_status_line status = rdata->msg_info.msg->line.status;
3148  struct ast_control_pvt_cause_code *cause_code;
3149  int data_size = sizeof(*cause_code);
3150  SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
3151 
3152  if (!session->channel) {
3153  SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
3154  }
3155 
3156  /* Build and send the tech-specific cause information */
3157  /* size of the string making up the cause code is "SIP " number + " " + reason length */
3158  data_size += 4 + 4 + pj_strlen(&status.reason);
3159  cause_code = ast_alloca(data_size);
3160  memset(cause_code, 0, data_size);
3161 
3163 
3164  snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %d %.*s", status.code,
3165  (int) pj_strlen(&status.reason), pj_strbuf(&status.reason));
3166 
3167  cause_code->ast_cause = hangup_sip2cause(status.code);
3168  ast_queue_control_data(session->channel, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
3169  ast_channel_hangupcause_hash_set(session->channel, cause_code, data_size);
3170 
3171  SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
3172 }
char chan_name[AST_CHANNEL_NAME]
static int hangup_sip2cause(int cause)
Convert SIP hangup causes to Asterisk hangup causes.
Definition: chan_pjsip.c:2849
struct ast_channel * channel
#define ast_alloca(size)
call __builtin_alloca to ensure we get gcc builtin semantics
Definition: astmm.h:290
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
#define SCOPE_EXIT_RTN(...)
Definition: logger.h:900
#define AST_CHANNEL_NAME
Definition: channel.h:172
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
void ast_copy_string(char *dst, const char *src, size_t size)
Size-limited null-terminating string copy.
Definition: strings.h:401
const char * ast_channel_name(const struct ast_channel *chan)
int ast_queue_control_data(struct ast_channel *chan, enum ast_control_frame_type control, const void *data, size_t datalen)
Queue a control frame with payload.
Definition: channel.c:1234
void ast_channel_hangupcause_hash_set(struct ast_channel *chan, const struct ast_control_pvt_cause_code *cause_code, int datalen)
Sets the HANGUPCAUSE hash and optionally the SIP_CAUSE hash on the given channel. ...
Definition: channel.c:4360
jack_status_t status
Definition: app_jack.c:146

◆ chan_pjsip_indicate()

static int chan_pjsip_indicate ( struct ast_channel ast,
int  condition,
const void *  data,
size_t  datalen 
)
static

Function called by core to ask the channel to indicate some sort of condition.

Definition at line 1612 of file chan_pjsip.c.

References ast_sip_session::active_media_state, ao2_bump, ao2_cleanup, ao2_ref, ast_assert, ast_channel_get_device_name(), ast_channel_lock, ast_channel_name(), ast_channel_nativeformats(), ast_channel_tech_pvt(), ast_channel_uniqueid(), ast_channel_unlock, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_CONNECTED_LINE, AST_CONTROL_HOLD, AST_CONTROL_INCOMPLETE, AST_CONTROL_MASQUERADE_NOTIFY, AST_CONTROL_PROCEEDING, AST_CONTROL_PROGRESS, AST_CONTROL_PVT_CAUSE_CODE, AST_CONTROL_REDIRECTING, AST_CONTROL_RINGING, AST_CONTROL_SRCCHANGE, AST_CONTROL_SRCUPDATE, AST_CONTROL_STREAM_TOPOLOGY_CHANGED, AST_CONTROL_STREAM_TOPOLOGY_REQUEST_CHANGE, AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED, AST_CONTROL_T38_PARAMETERS, AST_CONTROL_UNHOLD, AST_CONTROL_UPDATE_RTP_PEER, AST_CONTROL_VIDUPDATE, AST_DEVICE_ONHOLD, AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, ast_devstate_changed(), ast_devstate_changed_literal(), ast_format_cap_iscompatible_format(), AST_FORMAT_CMP_NOT_EQUAL, ast_format_h264, ast_format_h265, ast_format_vp8, ast_format_vp9, AST_FRAME_CONTROL, ast_frame_subclass2str(), ast_log, AST_MEDIA_TYPE_VIDEO, ast_moh_start(), ast_moh_stop(), ast_rtp_instance_write(), ast_sip_push_task(), ast_sip_session_suspend(), ast_sip_session_unsuspend(), ast_sorcery_object_get_id(), AST_STATE_RING, AST_STATE_UP, ast_str_tmp, ast_stream_topology_to_str(), AST_T38_REQUEST_PARMS, ast_test_suite_event_notify, ast_trace, AST_VECTOR_GET, AST_VECTOR_SIZE, chan_pjsip_add_hold(), chan_pjsip_remove_hold(), ast_sip_session::endpoint, ast_frame::frametype, handle_topology_request_change(), ast_sip_endpoint::inband_progress, indicate(), indicate_data_alloc(), ast_frame_subclass::integer, ast_sip_session::inv_session, LOG_ERROR, LOG_WARNING, ast_sip_endpoint::media, ast_sip_session::moh_passthrough, NULL, remote_send_hold(), remote_send_unhold(), ast_control_t38_parameters::request_response, ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_LOG_RTN_VALUE, SCOPE_EXIT_RTN_VALUE, ast_sip_session::serializer, ast_sip_channel_pvt::session, ast_frame::subclass, T38_PEER_REINVITE, ast_sip_session::t38state, transmit_info_with_vidupdate(), ast_sip_session_media::type, update_connected_line_information(), and ast_sip_endpoint_media_configuration::webrtc.

Referenced by chan_pjsip_pvt_dtor().

1613 {
1615  struct ast_sip_session_media *media;
1616  int response_code = 0;
1617  int res = 0;
1618  char *device_buf;
1619  size_t device_buf_size;
1620  int i;
1621  const struct ast_stream_topology *topology;
1622  struct ast_frame f = {
1624  .subclass = {
1625  .integer = condition
1626  }
1627  };
1628  char condition_name[256];
1629  SCOPE_ENTER(3, "%s: Indicated %s\n", ast_channel_name(ast),
1630  ast_frame_subclass2str(&f, condition_name, sizeof(condition_name), NULL, 0));
1631 
1632  switch (condition) {
1633  case AST_CONTROL_RINGING:
1634  if (ast_channel_state(ast) == AST_STATE_RING) {
1635  if (channel->session->endpoint->inband_progress ||
1636  (channel->session->inv_session && channel->session->inv_session->neg &&
1637  pjmedia_sdp_neg_get_state(channel->session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE)) {
1638  response_code = 183;
1639  res = -1;
1640  } else {
1641  response_code = 180;
1642  }
1643  } else {
1644  res = -1;
1645  }
1647  break;
1648  case AST_CONTROL_BUSY:
1649  if (ast_channel_state(ast) != AST_STATE_UP) {
1650  response_code = 486;
1651  } else {
1652  res = -1;
1653  }
1654  break;
1656  if (ast_channel_state(ast) != AST_STATE_UP) {
1657  response_code = 503;
1658  } else {
1659  res = -1;
1660  }
1661  break;
1663  if (ast_channel_state(ast) != AST_STATE_UP) {
1664  response_code = 484;
1665  } else {
1666  res = -1;
1667  }
1668  break;
1670  if (ast_channel_state(ast) != AST_STATE_UP) {
1671  response_code = 100;
1672  } else {
1673  res = -1;
1674  }
1675  break;
1676  case AST_CONTROL_PROGRESS:
1677  if (ast_channel_state(ast) != AST_STATE_UP) {
1678  response_code = 183;
1679  } else {
1680  res = -1;
1681  }
1683  break;
1684  case AST_CONTROL_VIDUPDATE:
1685  for (i = 0; i < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions); ++i) {
1686  media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, i);
1687  if (!media || media->type != AST_MEDIA_TYPE_VIDEO) {
1688  continue;
1689  }
1690  if (media->rtp) {
1691  /* FIXME: Only use this for VP8. Additional work would have to be done to
1692  * fully support other video codecs */
1693 
1697  (channel->session->endpoint->media.webrtc &&
1699  /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
1700  * RTP engine would provide a way to externally write/schedule RTCP
1701  * packets */
1702  struct ast_frame fr;
1704  fr.subclass.integer = AST_CONTROL_VIDUPDATE;
1705  res = ast_rtp_instance_write(media->rtp, &fr);
1706  } else {
1707  ao2_ref(channel->session, +1);
1709  ao2_cleanup(channel->session);
1710  }
1711  }
1712  ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Success");
1713  } else {
1714  ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Failure");
1715  res = -1;
1716  }
1717  }
1718  /* XXX If there were no video streams, then this should set
1719  * res to -1
1720  */
1721  break;
1723  ao2_ref(channel->session, +1);
1725  ao2_cleanup(channel->session);
1726  }
1727  break;
1729  break;
1731  res = -1;
1732  break;
1734  ast_assert(datalen == sizeof(int));
1735  if (*(int *) data) {
1736  /*
1737  * Masquerade is beginning:
1738  * Wait for session serializer to get suspended.
1739  */
1740  ast_channel_unlock(ast);
1741  ast_sip_session_suspend(channel->session);
1742  ast_channel_lock(ast);
1743  } else {
1744  /*
1745  * Masquerade is complete:
1746  * Unsuspend the session serializer.
1747  */
1749  }
1750  break;
1751  case AST_CONTROL_HOLD:
1753  device_buf_size = strlen(ast_channel_name(ast)) + 1;
1754  device_buf = alloca(device_buf_size);
1755  ast_channel_get_device_name(ast, device_buf, device_buf_size);
1757  if (!channel->session->moh_passthrough) {
1758  ast_moh_start(ast, data, NULL);
1759  } else {
1761  ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n",
1763  ao2_ref(channel->session, -1);
1764  }
1765  }
1766  break;
1767  case AST_CONTROL_UNHOLD:
1769  device_buf_size = strlen(ast_channel_name(ast)) + 1;
1770  device_buf = alloca(device_buf_size);
1771  ast_channel_get_device_name(ast, device_buf, device_buf_size);
1773  if (!channel->session->moh_passthrough) {
1774  ast_moh_stop(ast);
1775  } else {
1777  ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n",
1779  ao2_ref(channel->session, -1);
1780  }
1781  }
1782  break;
1783  case AST_CONTROL_SRCUPDATE:
1784  break;
1785  case AST_CONTROL_SRCCHANGE:
1786  break;
1788  if (ast_channel_state(ast) != AST_STATE_UP) {
1789  response_code = 181;
1790  } else {
1791  res = -1;
1792  }
1793  break;
1795  res = 0;
1796 
1797  if (channel->session->t38state == T38_PEER_REINVITE) {
1798  const struct ast_control_t38_parameters *parameters = data;
1799 
1800  if (parameters->request_response == AST_T38_REQUEST_PARMS) {
1801  res = AST_T38_REQUEST_PARMS;
1802  }
1803  }
1804 
1805  break;
1807  topology = data;
1808  ast_trace(-1, "%s: New topology: %s\n", ast_channel_name(ast),
1809  ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
1810  res = handle_topology_request_change(channel->session, topology);
1811  break;
1813  break;
1815  break;
1816  case -1:
1817  res = -1;
1818  break;
1819  default:
1820  ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
1821  res = -1;
1822  break;
1823  }
1824 
1825  if (response_code) {
1826  struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
1827 
1828  if (!ind_data) {
1829  SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc indicate data\n", ast_channel_name(ast));
1830  }
1831 
1832  if (ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1833  ast_log(LOG_ERROR, "%s: Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1834  ast_channel_name(ast), response_code, ast_sorcery_object_get_id(channel->session->endpoint));
1835  ao2_cleanup(ind_data);
1836  res = -1;
1837  }
1838  }
1839 
1840  SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_channel_name(ast));
1841 }
static int remote_send_hold(void *data)
Update local hold state to be held.
Definition: chan_pjsip.c:1500
struct ast_format * ast_format_vp8
Built-in cached vp8 format.
Definition: format_cache.c:191
#define SCOPE_EXIT_LOG_RTN_VALUE(__value, __log_level,...)
Definition: logger.h:927
enum ast_sip_session_t38state t38state
#define ast_channel_lock(chan)
Definition: channel.h:2890
static int indicate(void *data)
Definition: chan_pjsip.c:1333
struct ast_sip_endpoint * endpoint
static int transmit_info_with_vidupdate(void *data)
Send SIP INFO with video update request.
Definition: chan_pjsip.c:1351
void ast_sip_session_unsuspend(struct ast_sip_session *session)
Request the session serializer be unsuspended.
static int remote_send_unhold(void *data)
Update local hold state to be unheld.
Definition: chan_pjsip.c:1506
void * ast_channel_tech_pvt(const struct ast_channel *chan)
void ast_sip_session_suspend(struct ast_sip_session *session)
Request and wait for the session serializer to be suspended.
#define LOG_WARNING
Definition: logger.h:274
union ast_frame::@257 data
static int chan_pjsip_add_hold(const char *chan_uid)
Add a channel ID to the list of PJSIP channels on hold.
Definition: chan_pjsip.c:1117
int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
Send a frame out over RTP.
Definition: rtp_engine.c:568
static struct indicate_data * indicate_data_alloc(struct ast_sip_session *session, int condition, int response_code, const void *frame_data, size_t datalen)
Definition: chan_pjsip.c:1308
enum ast_control_t38 request_response
#define ast_trace(level,...)
Definition: logger.h:876
ast_channel_state
ast_channel states
Definition: channelstate.h:35
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
#define ast_assert(a)
Definition: utils.h:650
Definition: muted.c:95
#define NULL
Definition: resample.c:96
struct pjsip_inv_session * inv_session
void ast_moh_stop(struct ast_channel *chan)
Turn off music on hold on a given channel.
Definition: channel.c:7739
unsigned int inband_progress
Definition: res_pjsip.h:863
static void chan_pjsip_remove_hold(const char *chan_uid)
Remove a channel ID from the list of PJSIP channels on hold.
Definition: chan_pjsip.c:1148
static int update_connected_line_information(void *data)
Update connected line information.
Definition: chan_pjsip.c:1429
struct ast_sip_session_media_state * active_media_state
enum ast_format_cmp_res ast_format_cap_iscompatible_format(const struct ast_format_cap *cap, const struct ast_format *format)
Find if ast_format is within the capabilities of the ast_format_cap object.
Definition: format_cap.c:583
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
#define ao2_bump(obj)
Definition: astobj2.h:491
#define ast_str_tmp(init_len, __expr)
Definition: strings.h:1136
#define ast_log
Definition: astobj2.c:42
const char * ast_stream_topology_to_str(const struct ast_stream_topology *topology, struct ast_str **buf)
Get a string representing the topology for debugging/display purposes.
Definition: stream.c:936
int ast_devstate_changed(enum ast_device_state state, enum ast_devstate_cache cachable, const char *fmt,...)
Tells Asterisk the State for Device is changed.
Definition: devicestate.c:510
size_t datalen
Definition: chan_pjsip.c:1297
#define ao2_ref(o, delta)
Definition: astobj2.h:464
char * ast_frame_subclass2str(struct ast_frame *f, char *subclass, size_t slen, char *moreinfo, size_t mlen)
Copy the discription of a frame&#39;s subclass into the provided string.
Definition: main/frame.c:406
const char * ast_sorcery_object_get_id(const void *object)
Get the unique identifier of a sorcery object.
Definition: sorcery.c:2309
struct ast_taskprocessor * serializer
#define ast_test_suite_event_notify(s, f,...)
Definition: test.h:196
const char * ast_channel_uniqueid(const struct ast_channel *chan)
#define LOG_ERROR
Definition: logger.h:285
int ast_sip_push_task(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Pushes a task to SIP servants.
Definition: res_pjsip.c:5023
struct ast_format * ast_format_vp9
Built-in cached vp9 format.
Definition: format_cache.c:196
int ast_moh_start(struct ast_channel *chan, const char *mclass, const char *interpclass)
Turn on music on hold on a given channel.
Definition: channel.c:7729
struct ast_format * ast_format_h264
Built-in cached h264 format.
Definition: format_cache.c:176
#define ast_channel_unlock(chan)
Definition: channel.h:2891
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
A structure containing SIP session media information.
struct ast_format_cap * ast_channel_nativeformats(const struct ast_channel *chan)
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
struct ast_format * ast_format_h265
Built-in cached h265 format.
Definition: format_cache.c:181
#define AST_VECTOR_GET(vec, idx)
Get an element from a vector.
Definition: vector.h:682
enum ast_media_type type
Media type of this session media.
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
struct ast_rtp_instance * rtp
RTP instance itself.
const char * ast_channel_name(const struct ast_channel *chan)
static int handle_topology_request_change(struct ast_sip_session *session, const struct ast_stream_topology *proposed)
Definition: chan_pjsip.c:1592
int ast_channel_get_device_name(struct ast_channel *chan, char *device_name, size_t name_buffer_length)
Get a device name given its channel structure.
Definition: channel.c:10554
Data structure associated with a single frame of data.
enum ast_frame_type frametype
int ast_devstate_changed_literal(enum ast_device_state state, enum ast_devstate_cache cachable, const char *device)
Tells Asterisk the State for Device is changed.
Definition: devicestate.c:471
#define AST_VECTOR_SIZE(vec)
Get the number of elements in a vector.
Definition: vector.h:611
unsigned int moh_passthrough

◆ chan_pjsip_new()

static struct ast_channel* chan_pjsip_new ( struct ast_sip_session session,
int  state,
const char *  exten,
const char *  title,
const struct ast_assigned_ids assignedids,
const struct ast_channel requestor,
const char *  cid_name 
)
static

Function called to create a new PJSIP Asterisk channel.

Definition at line 547 of file chan_pjsip.c.

References ast_sip_endpoint::accountcode, AO2_ALLOC_OPT_LOCK_NOLOCK, ao2_alloc_options, ao2_cleanup, ao2_ref, AST_ADSI_UNAVAILABLE, ast_atomic_fetchadd_int(), ast_channel_adsicpe_set(), ast_channel_alloc_with_endpoint, ast_channel_caller(), ast_channel_callgroup_set(), ast_channel_dialed(), ast_channel_named_callgroups_set(), ast_channel_named_pickupgroups_set(), ast_channel_nativeformats_set(), ast_channel_pickupgroup_set(), ast_channel_priority_set(), ast_channel_rings_set(), ast_channel_set_rawreadformat(), ast_channel_set_rawwriteformat(), ast_channel_set_readformat(), ast_channel_set_stream_topology(), ast_channel_set_writeformat(), ast_channel_stage_snapshot(), ast_channel_stage_snapshot_done(), ast_channel_tech_pvt_set(), ast_channel_tech_set(), ast_channel_uniqueid(), ast_channel_unlock, ast_channel_zone_set(), ast_format_cap_alloc, ast_format_cap_append_from_cap(), ast_format_cap_empty(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_format_cap_get_best_by_type(), ast_format_cap_get_format(), ast_get_encoded_str(), ast_get_indication_zone(), ast_hangup(), ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_UNKNOWN, ast_party_id_copy(), ast_sip_channel_pvt_alloc(), ast_sip_session_get_name(), ast_sorcery_object_get_id(), AST_STATE_RING, ast_strdup, ast_stream_topology_clone(), ast_stream_topology_free(), ast_stream_topology_get_count(), ast_stream_topology_get_formats(), ast_strlen_zero, buf, ast_sip_endpoint_pickup_configuration::callgroup, rtp_direct_media_data::chan, chan_idx, chan_pjsip_pvt_dtor(), ast_sip_endpoint::channel_vars, ast_sip_endpoint_media_configuration::codecs, compatible_formats_exist(), ast_sip_endpoint::context, ast_sip_session::endpoint, ast_sip_session::id, ast_sip_endpoint::language, LOG_ERROR, ast_sip_endpoint::media, ast_variable::name, ast_party_id::name, ast_sip_endpoint_pickup_configuration::named_callgroups, ast_sip_endpoint_pickup_configuration::named_pickupgroups, ast_variable::next, NULL, ast_party_id::number, ast_party_dialed::number, pbx_builtin_setvar_helper(), ast_sip_session::pending_media_state, ast_sip_endpoint::persistent, ast_sip_endpoint::pickup, ast_sip_endpoint_pickup_configuration::pickupgroup, RAII_VAR, S_COR, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, set_channel_on_rtp_instance(), ast_party_name::str, ast_party_number::str, ast_party_dialed::str, ast_sip_session_media_state::topology, ast_sip_endpoint_media_configuration::topology, ast_party_name::valid, ast_party_number::valid, ast_variable::value, var, and ast_sip_endpoint::zone.

Referenced by chan_pjsip_incoming_request(), and chan_pjsip_request_with_stream_topology().

548 {
549  struct ast_channel *chan;
550  struct ast_format_cap *caps;
551  RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
553  struct ast_variable *var;
554  struct ast_stream_topology *topology;
555  SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
556 
557  if (!(pvt = ao2_alloc_options(sizeof(*pvt), chan_pjsip_pvt_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK))) {
558  SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt\n");
559  }
560 
562  S_COR(session->id.number.valid, session->id.number.str, ""),
563  S_COR(session->id.name.valid, session->id.name.str, ""),
564  session->endpoint->accountcode,
565  exten, session->endpoint->context,
566  assignedids, requestor, 0,
567  session->endpoint->persistent, "PJSIP/%s-%08x",
569  (unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
570  if (!chan) {
571  SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
572  }
573 
575 
576  if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
577  ast_channel_unlock(chan);
578  ast_hangup(chan);
579  SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt channel\n");
580  }
581 
582  ast_channel_tech_pvt_set(chan, channel);
583 
587  if (!caps) {
588  ast_channel_unlock(chan);
589  ast_hangup(chan);
590  SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create caps\n");
591  }
593  topology = ast_stream_topology_clone(session->endpoint->media.topology);
594  } else {
597  }
598 
599  if (!topology || !caps) {
600  ao2_cleanup(caps);
601  ast_stream_topology_free(topology);
602  ast_channel_unlock(chan);
603  ast_hangup(chan);
604  SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't get caps or clone topology\n");
605  }
606 
608 
609  ast_channel_nativeformats_set(chan, caps);
610  ast_channel_set_stream_topology(chan, topology);
611 
612  if (!ast_format_cap_empty(caps)) {
613  struct ast_format *fmt;
614 
616  if (!fmt) {
617  /* Since our capabilities aren't empty, this will succeed */
618  fmt = ast_format_cap_get_format(caps, 0);
619  }
620  ast_channel_set_writeformat(chan, fmt);
622  ast_channel_set_readformat(chan, fmt);
624  ao2_ref(fmt, -1);
625  }
626 
627  ao2_ref(caps, -1);
628 
629  if (state == AST_STATE_RING) {
630  ast_channel_rings_set(chan, 1);
631  }
632 
634 
635  ast_party_id_copy(&ast_channel_caller(chan)->id, &session->id);
636  ast_party_id_copy(&ast_channel_caller(chan)->ani, &session->id);
637 
638  if (!ast_strlen_zero(exten)) {
639  /* Set provided DNID on the new channel. */
641  }
642 
643  ast_channel_priority_set(chan, 1);
644 
647 
650 
651  if (!ast_strlen_zero(session->endpoint->language)) {
652  ast_channel_language_set(chan, session->endpoint->language);
653  }
654 
655  if (!ast_strlen_zero(session->endpoint->zone)) {
656  struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
657  if (!zone) {
658  ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
659  }
660  ast_channel_zone_set(chan, zone);
661  }
662 
663  for (var = session->endpoint->channel_vars; var; var = var->next) {
664  char buf[512];
666  var->value, buf, sizeof(buf)));
667  }
668 
670  ast_channel_unlock(chan);
671 
673 
674  SCOPE_EXIT_RTN_VALUE(chan);
675 }
struct ast_party_caller * ast_channel_caller(struct ast_channel *chan)
struct ast_sip_endpoint_pickup_configuration pickup
Definition: res_pjsip.h:851
struct ast_variable * next
static unsigned int chan_idx
Definition: chan_pjsip.c:80
void ast_channel_pickupgroup_set(struct ast_channel *chan, ast_group_t value)
static char exten[AST_MAX_EXTENSION]
Definition: chan_alsa.c:118
Main Channel structure associated with a channel.
struct ast_sip_endpoint * endpoint
char * str
Subscriber phone number (Malloced)
Definition: channel.h:292
char * str
Subscriber phone number (Malloced)
Definition: channel.h:387
static void chan_pjsip_pvt_dtor(void *obj)
Definition: chan_pjsip.c:82
struct ast_sip_session_media_state * pending_media_state
#define ast_channel_alloc_with_endpoint(needqueue, state, cid_num, cid_name, acctcode, exten, context, assignedids, requestor, amaflag, endpoint,...)
Definition: channel.h:1263
char * ast_get_encoded_str(const char *stream, char *result, size_t result_len)
Decode a stream of encoded control or extended ASCII characters.
Definition: main/app.c:2925
void ast_channel_set_writeformat(struct ast_channel *chan, struct ast_format *format)
struct ast_format_cap * ast_stream_topology_get_formats(struct ast_stream_topology *topology)
Create a format capabilities structure representing the topology.
Definition: stream.c:930
struct ast_party_name name
Subscriber name.
Definition: channel.h:341
char buf[BUFSIZE]
Definition: eagi_proxy.c:66
void ast_channel_set_rawwriteformat(struct ast_channel *chan, struct ast_format *format)
void ast_party_id_copy(struct ast_party_id *dest, const struct ast_party_id *src)
Copy the source party id information to the destination party id.
Definition: channel.c:1751
Structure for variables, used for configurations and for channel variables.
#define var
Definition: ast_expr2f.c:614
struct ast_format_cap * codecs
Definition: res_pjsip.h:770
const ast_string_field context
Definition: res_pjsip.h:815
void ast_channel_callgroup_set(struct ast_channel *chan, ast_group_t value)
char * str
Subscriber name (Malloced)
Definition: channel.h:265
A structure which contains a channel implementation and session.
Definition of a media format.
Definition: format.c:43
void ast_channel_named_pickupgroups_set(struct ast_channel *chan, struct ast_namedgroups *value)
struct ast_format * ast_format_cap_get_best_by_type(const struct ast_format_cap *cap, enum ast_media_type type)
Get the most preferred format for a particular media type.
Definition: format_cap.c:417
#define ao2_alloc_options(data_size, destructor_fn, options)
Definition: astobj2.h:406
Definition: muted.c:95
struct ast_party_dialed::@240 number
Dialed/Called number.
#define ast_strdup(str)
A wrapper for strdup()
Definition: astmm.h:243
#define NULL
Definition: resample.c:96
void ast_channel_zone_set(struct ast_channel *chan, struct ast_tone_zone *value)
int ast_atomic_fetchadd_int(volatile int *p, int v)
Atomically add v to *p and return the previous value of *p.
Definition: lock.h:755
void ast_channel_tech_set(struct ast_channel *chan, const struct ast_channel_tech *value)
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
#define ast_log
Definition: astobj2.c:42
void ast_channel_set_rawreadformat(struct ast_channel *chan, struct ast_format *format)
static void set_channel_on_rtp_instance(const struct ast_sip_session *session, const char *channel_id)
Definition: chan_pjsip.c:494
struct ast_namedgroups * named_pickupgroups
Definition: res_pjsip.h:663
void ast_channel_rings_set(struct ast_channel *chan, int value)
static int compatible_formats_exist(struct ast_stream_topology *top, struct ast_format_cap *cap)
Determine if a topology is compatible with format capabilities.
Definition: chan_pjsip.c:526
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition: utils.h:851
A set of tones for a given locale.
Definition: indications.h:74
void ast_channel_nativeformats_set(struct ast_channel *chan, struct ast_format_cap *value)
void ast_channel_stage_snapshot_done(struct ast_channel *chan)
Clear flag to indicate channel snapshot is being staged, and publish snapshot.
const ast_string_field language
Definition: res_pjsip.h:827
struct ast_sip_channel_pvt * ast_sip_channel_pvt_alloc(void *pvt, struct ast_sip_session *session)
Allocate a new SIP channel pvt structure.
#define ao2_ref(o, delta)
Definition: astobj2.h:464
void ast_channel_set_readformat(struct ast_channel *chan, struct ast_format *format)
#define S_COR(a, b, c)
returns the equivalent of logic or for strings, with an additional boolean check: second one if not e...
Definition: strings.h:85
const ast_string_field accountcode
Definition: res_pjsip.h:835
const char * ast_sorcery_object_get_id(const void *object)
Get the unique identifier of a sorcery object.
Definition: sorcery.c:2309
struct ast_namedgroups * named_callgroups
Definition: res_pjsip.h:661
#define ast_format_cap_alloc(flags)
Definition: format_cap.h:52
void ast_channel_adsicpe_set(struct ast_channel *chan, enum ast_channel_adsicpe value)
const ast_string_field zone
Definition: res_pjsip.h:825
const char * ast_channel_uniqueid(const struct ast_channel *chan)
void ast_channel_stage_snapshot(struct ast_channel *chan)
Set flag to indicate channel snapshot is being staged.
#define LOG_ERROR
Definition: logger.h:285
struct ast_stream_topology * ast_channel_set_stream_topology(struct ast_channel *chan, struct ast_stream_topology *topology)
Set the topology of streams on a channel.
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
Format capabilities structure, holds formats + preference order + etc.
Definition: format_cap.c:54
void ast_channel_named_callgroups_set(struct ast_channel *chan, struct ast_namedgroups *value)
struct ast_tone_zone * ast_get_indication_zone(const char *country)
locate ast_tone_zone
Definition: indications.c:433
The PJSIP channel driver pvt, stored in the ast_sip_channel_pvt data structure.
Definition: chan_pjsip.h:42
struct ast_party_dialed * ast_channel_dialed(struct ast_channel *chan)
#define ast_strlen_zero(a)
Definition: muted.c:73
#define ast_channel_unlock(chan)
Definition: channel.h:2891
void ast_hangup(struct ast_channel *chan)
Hang up a channel.
Definition: channel.c:2534
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
int ast_stream_topology_get_count(const struct ast_stream_topology *topology)
Get the number of streams in a topology.
Definition: stream.c:765
int pbx_builtin_setvar_helper(struct ast_channel *chan, const char *name, const char *value)
Add a variable to the channel variable stack, removing the most recently set value for the same name...
struct ast_stream_topology * ast_stream_topology_clone(const struct ast_stream_topology *topology)
Create a deep clone of an existing stream topology.
Definition: stream.c:667
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
struct ast_endpoint * persistent
Definition: res_pjsip.h:865
struct ast_stream_topology * topology
Definition: res_pjsip.h:772
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
struct ast_stream_topology * topology
The media stream topology.
int ast_format_cap_empty(const struct ast_format_cap *cap)
Determine if a format cap has no formats in it.
Definition: format_cap.c:746
int ast_format_cap_append_from_cap(struct ast_format_cap *dst, const struct ast_format_cap *src, enum ast_media_type type)
Append the formats of provided type in src to dst.
Definition: format_cap.c:269
unsigned char valid
TRUE if the name information is valid/present.
Definition: channel.h:280
struct ast_variable * channel_vars
Definition: res_pjsip.h:875
struct ast_format * ast_format_cap_get_format(const struct ast_format_cap *cap, int position)
Get the format at a specific index.
Definition: format_cap.c:400
void ast_stream_topology_free(struct ast_stream_topology *topology)
Unreference and destroy a stream topology.
Definition: stream.c:743
void ast_channel_priority_set(struct ast_channel *chan, int value)
void ast_channel_tech_pvt_set(struct ast_channel *chan, void *value)
struct ast_party_id id
unsigned char valid
TRUE if the number information is valid/present.
Definition: channel.h:298
struct ast_channel_tech chan_pjsip_tech
PBX interface structure for channel registration.
Definition: chan_pjsip.c:109
struct ast_party_number number
Subscriber phone number.
Definition: channel.h:343

◆ chan_pjsip_pvt_dtor()

static void chan_pjsip_pvt_dtor ( void *  obj)
static

◆ chan_pjsip_queryoption()

static int chan_pjsip_queryoption ( struct ast_channel ast,
int  option,
void *  data,
int *  datalen 
)
static

Function called to query options on a channel.

Definition at line 1237 of file chan_pjsip.c.

References ast_channel_tech_pvt(), AST_OPTION_T38_STATE, ast_sip_t38_configuration::enabled, ast_sip_session::endpoint, ast_sip_endpoint::media, ast_sip_channel_pvt::session, state, ast_sip_endpoint_media_configuration::t38, T38_ENABLED, T38_LOCAL_REINVITE, T38_PEER_REINVITE, T38_REJECTED, T38_STATE_NEGOTIATED, T38_STATE_NEGOTIATING, T38_STATE_REJECTED, T38_STATE_UNAVAILABLE, T38_STATE_UNKNOWN, and ast_sip_session::t38state.

Referenced by chan_pjsip_pvt_dtor().

1238 {
1240  int res = -1;
1242 
1243  if (!channel) {
1244  return -1;
1245  }
1246 
1247  switch (option) {
1248  case AST_OPTION_T38_STATE:
1249  if (channel->session->endpoint->media.t38.enabled) {
1250  switch (channel->session->t38state) {
1251  case T38_LOCAL_REINVITE:
1252  case T38_PEER_REINVITE:
1253  state = T38_STATE_NEGOTIATING;
1254  break;
1255  case T38_ENABLED:
1256  state = T38_STATE_NEGOTIATED;
1257  break;
1258  case T38_REJECTED:
1259  state = T38_STATE_REJECTED;
1260  break;
1261  default:
1262  state = T38_STATE_UNKNOWN;
1263  break;
1264  }
1265  }
1266 
1267  *((enum ast_t38_state *) data) = state;
1268  res = 0;
1269 
1270  break;
1271  default:
1272  break;
1273  }
1274 
1275  return res;
1276 }
enum sip_cc_notify_state state
Definition: chan_sip.c:957
enum ast_sip_session_t38state t38state
#define T38_ENABLED
Definition: chan_ooh323.c:102
struct ast_sip_endpoint * endpoint
void * ast_channel_tech_pvt(const struct ast_channel *chan)
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
ast_t38_state
Possible T38 states on channels.
Definition: channel.h:879
Definition: muted.c:95
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
struct ast_sip_t38_configuration t38
Definition: res_pjsip.h:768
#define AST_OPTION_T38_STATE

◆ chan_pjsip_read_stream()

static struct ast_frame * chan_pjsip_read_stream ( struct ast_channel ast)
static

Function called by core to read any waiting frames.

Note
The channel is already locked.

Definition at line 838 of file chan_pjsip.c.

References ast_sip_session::active_media_state, ao2_ref, ast_channel_fdno(), ast_channel_get_up_time(), ast_channel_is_bridged(), ast_channel_name(), ast_channel_nativeformats(), ast_channel_nativeformats_set(), ast_channel_rawwriteformat(), ast_channel_readformat(), ast_channel_set_unbridged_nolock(), ast_channel_tech_pvt(), ast_channel_writeformat(), ast_debug, ast_dsp_free(), ast_dsp_get_features(), ast_dsp_process(), ast_dsp_set_features(), AST_EXTENDED_FDS, ast_format_cap_alloc, ast_format_cap_append, ast_format_cap_append_from_cap(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_format_cap_iscompatible_format(), ast_format_cap_remove_by_type(), ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, ast_format_get_name(), AST_FRAME_DTMF, AST_FRAME_VOICE, ast_frfree, AST_LIST_NEXT, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_UNKNOWN, ast_null_frame, ast_set_read_format_path(), ast_set_write_format_path(), AST_VECTOR_GET_ADDR, AST_VECTOR_SIZE, ast_sip_endpoint::asymmetric_rtp_codec, chan_pjsip_cng_tone_detected(), ast_sip_session_media_state::default_session, ast_sip_session::dsp, DSP_FEATURE_FAX_DETECT, ast_sip_session::endpoint, ast_sip_endpoint::faxdetect_timeout, ast_frame_subclass::format, ast_frame::frametype, ast_frame_subclass::integer, is_compatible_format(), NULL, ast_sip_session_media_read_callback_state::read_callback, ast_sip_session_media_read_callback_state::session, ast_sip_channel_pvt::session, ast_frame::subclass, and ast_sip_session_media::type.

Referenced by chan_pjsip_pvt_dtor().

839 {
841  struct ast_sip_session *session = channel->session;
842  struct ast_sip_session_media_read_callback_state *callback_state;
843  struct ast_frame *f;
844  int fdno = ast_channel_fdno(ast) - AST_EXTENDED_FDS;
845  struct ast_frame *cur;
846 
847  if (fdno >= AST_VECTOR_SIZE(&session->active_media_state->read_callbacks)) {
848  return &ast_null_frame;
849  }
850 
851  callback_state = AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, fdno);
852  f = callback_state->read_callback(session, callback_state->session);
853 
854  if (!f) {
855  return f;
856  }
857 
858  for (cur = f; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
859  if (cur->frametype == AST_FRAME_VOICE) {
860  break;
861  }
862  }
863 
864  if (!cur || callback_state->session != session->active_media_state->default_session[callback_state->session->type]) {
865  return f;
866  }
867 
868  session = channel->session;
869 
870  /*
871  * Asymmetric RTP only has one native format set at a time.
872  * Therefore we need to update the native format to the current
873  * raw read format BEFORE the native format check
874  */
875  if (!session->endpoint->asymmetric_rtp_codec &&
877  is_compatible_format(session, cur)) {
878  struct ast_format_cap *caps;
879 
880  /* For maximum compatibility we ensure that the formats match that of the received media */
881  ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
884 
886  if (caps) {
889  ast_format_cap_append(caps, cur->subclass.format, 0);
891  ao2_ref(caps, -1);
892  }
893 
896 
897  if (ast_channel_is_bridged(ast)) {
899  }
900  }
901 
904  ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n",
906  ast_frfree(f);
907  return &ast_null_frame;
908  }
909 
910  if (session->dsp) {
911  int dsp_features;
912 
913  dsp_features = ast_dsp_get_features(session->dsp);
914  if ((dsp_features & DSP_FEATURE_FAX_DETECT)
915  && session->endpoint->faxdetect_timeout
916  && session->endpoint->faxdetect_timeout <= ast_channel_get_up_time(ast)) {
917  dsp_features &= ~DSP_FEATURE_FAX_DETECT;
918  if (dsp_features) {
919  ast_dsp_set_features(session->dsp, dsp_features);
920  } else {
921  ast_dsp_free(session->dsp);
922  session->dsp = NULL;
923  }
924  ast_debug(3, "Channel driver fax CNG detection timeout on %s\n",
925  ast_channel_name(ast));
926  }
927  }
928  if (session->dsp) {
929  f = ast_dsp_process(ast, session->dsp, f);
930  if (f && (f->frametype == AST_FRAME_DTMF)) {
931  if (f->subclass.integer == 'f') {
932  ast_debug(3, "Channel driver fax CNG detected on %s\n",
933  ast_channel_name(ast));
934  f = chan_pjsip_cng_tone_detected(ast, session, f);
935  /* When chan_pjsip_cng_tone_detected returns it is possible for the
936  * channel pointed to by ast and by session->channel to differ due to a
937  * masquerade. It's best not to touch things after this.
938  */
939  } else {
940  ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
941  ast_channel_name(ast));
942  }
943  }
944  }
945 
946  return f;
947 }
struct ast_frame * ast_dsp_process(struct ast_channel *chan, struct ast_dsp *dsp, struct ast_frame *inf)
Return AST_FRAME_NULL frames when there is silence, AST_FRAME_BUSY on busies, and call progress...
Definition: dsp.c:1484
struct ast_sip_endpoint * endpoint
static int is_compatible_format(struct ast_sip_session *session, struct ast_frame *f)
Determine if the given frame is in a format we&#39;ve negotiated.
Definition: chan_pjsip.c:824
void * ast_channel_tech_pvt(const struct ast_channel *chan)
void ast_dsp_free(struct ast_dsp *dsp)
Definition: dsp.c:1744
unsigned int asymmetric_rtp_codec
Definition: res_pjsip.h:891
#define AST_LIST_NEXT(elm, field)
Returns the next entry in the list after the given entry.
Definition: linkedlists.h:438
struct ast_sip_session_media * default_session[AST_MEDIA_TYPE_END]
Default media sessions for each type.
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
int ast_channel_get_up_time(struct ast_channel *chan)
Obtain how long it has been since the channel was answered.
Definition: channel.c:2840
Definition: muted.c:95
const char * ast_format_get_name(const struct ast_format *format)
Get the name associated with a format.
Definition: format.c:334
#define NULL
Definition: resample.c:96
#define AST_FRAME_DTMF
A structure describing a SIP session.
struct ast_frame_subclass subclass
static struct ast_frame * chan_pjsip_cng_tone_detected(struct ast_channel *ast, struct ast_sip_session *session, struct ast_frame *f)
Internal helper function called when CNG tone is detected.
Definition: chan_pjsip.c:765
int ast_set_read_format_path(struct ast_channel *chan, struct ast_format *raw_format, struct ast_format *core_format)
Set specific read path on channel.
Definition: channel.c:5474
struct ast_sip_session_media_state * active_media_state
struct ast_format * ast_channel_readformat(struct ast_channel *chan)
enum ast_format_cmp_res ast_format_cap_iscompatible_format(const struct ast_format_cap *cap, const struct ast_format *format)
Find if ast_format is within the capabilities of the ast_format_cap object.
Definition: format_cap.c:583
struct ast_dsp * dsp
#define ast_debug(level,...)
Log a DEBUG message.
Definition: logger.h:444
int ast_dsp_get_features(struct ast_dsp *dsp)
Get features.
Definition: dsp.c:1738
enum ast_format_cmp_res ast_format_cmp(const struct ast_format *format1, const struct ast_format *format2)
Compare two formats.
Definition: format.c:201
#define AST_VECTOR_GET_ADDR(vec, idx)
Get an address of element in a vector.
Definition: vector.h:670
void ast_channel_nativeformats_set(struct ast_channel *chan, struct ast_format_cap *value)
static struct ast_mansession session
#define ao2_ref(o, delta)
Definition: astobj2.h:464
#define DSP_FEATURE_FAX_DETECT
Definition: dsp.h:29
ast_sip_session_media_read_cb read_callback
The callback to invoke.
struct ast_sip_session_media * session
The media session.
#define ast_format_cap_append(cap, format, framing)
Definition: format_cap.h:103
void ast_channel_set_unbridged_nolock(struct ast_channel *chan, int value)
Variant of ast_channel_set_unbridged. Use this if the channel is already locked prior to calling...
Structure which contains read callback information.
#define ast_format_cap_alloc(flags)
Definition: format_cap.h:52
void ast_format_cap_remove_by_type(struct ast_format_cap *cap, enum ast_media_type type)
Remove all formats matching a specific format type.
Definition: format_cap.c:525
int ast_channel_fdno(const struct ast_channel *chan)
#define AST_EXTENDED_FDS
Definition: channel.h:196
Format capabilities structure, holds formats + preference order + etc.
Definition: format_cap.c:54
unsigned int faxdetect_timeout
Definition: res_pjsip.h:885
int ast_channel_is_bridged(const struct ast_channel *chan)
Determine if a channel is in a bridge.
Definition: channel.c:10603
int ast_set_write_format_path(struct ast_channel *chan, struct ast_format *core_format, struct ast_format *raw_format)
Set specific write path on channel.
Definition: channel.c:5510
void ast_dsp_set_features(struct ast_dsp *dsp, int features)
Select feature set.
Definition: dsp.c:1729
struct ast_format_cap * ast_channel_nativeformats(const struct ast_channel *chan)
struct ast_frame ast_null_frame
Definition: main/frame.c:79
enum ast_media_type type
Media type of this session media.
const char * ast_channel_name(const struct ast_channel *chan)
#define ast_frfree(fr)
Data structure associated with a single frame of data.
enum ast_frame_type frametype
int ast_format_cap_append_from_cap(struct ast_format_cap *dst, const struct ast_format_cap *src, enum ast_media_type type)
Append the formats of provided type in src to dst.
Definition: format_cap.c:269
struct ast_format * format
struct ast_format * ast_channel_writeformat(struct ast_channel *chan)
struct ast_format * ast_channel_rawwriteformat(struct ast_channel *chan)
#define AST_VECTOR_SIZE(vec)
Get the number of elements in a vector.
Definition: vector.h:611

◆ chan_pjsip_remove_hold()

static void chan_pjsip_remove_hold ( const char *  chan_uid)
static

Remove a channel ID from the list of PJSIP channels on hold.

Parameters
chan_uid- Unique ID of the channel being taken out of the hold list

Definition at line 1148 of file chan_pjsip.c.

References ao2_find, OBJ_NODATA, OBJ_SEARCH_KEY, and OBJ_UNLINK.

Referenced by chan_pjsip_indicate(), and chan_pjsip_session_end().

1149 {
1151 }
The arg parameter is a search key, but is not an object.
Definition: astobj2.h:1105
static struct ao2_container * pjsip_uids_onhold
Definition: chan_pjsip.c:1107
#define ao2_find(container, arg, flags)
Definition: astobj2.h:1756

◆ chan_pjsip_request()

static struct ast_channel * chan_pjsip_request ( const char *  type,
struct ast_format_cap cap,
const struct ast_assigned_ids assignedids,
const struct ast_channel requestor,
const char *  data,
int *  cause 
)
static

Definition at line 2683 of file chan_pjsip.c.

References ast_stream_topology_create_from_format_cap(), ast_stream_topology_free(), rtp_direct_media_data::chan, chan_pjsip_request_with_stream_topology(), and NULL.

Referenced by chan_pjsip_pvt_dtor().

2684 {
2685  struct ast_stream_topology *topology;
2686  struct ast_channel *chan;
2687 
2689  if (!topology) {
2690  return NULL;
2691  }
2692 
2693  chan = chan_pjsip_request_with_stream_topology(type, topology, assignedids, requestor, data, cause);
2694 
2695  ast_stream_topology_free(topology);
2696 
2697  return chan;
2698 }
static const char type[]
Definition: chan_ooh323.c:109
Main Channel structure associated with a channel.
struct ast_stream_topology * ast_stream_topology_create_from_format_cap(struct ast_format_cap *cap)
A helper function that, given a format capabilities structure, creates a topology and separates the m...
Definition: stream.c:848
#define NULL
Definition: resample.c:96
const char * data
static struct ast_channel * chan_pjsip_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
Function called by core to create a new outgoing PJSIP session.
Definition: chan_pjsip.c:2656
void ast_stream_topology_free(struct ast_stream_topology *topology)
Unreference and destroy a stream topology.
Definition: stream.c:743

◆ chan_pjsip_request_with_stream_topology()

static struct ast_channel * chan_pjsip_request_with_stream_topology ( const char *  type,
struct ast_stream_topology topology,
const struct ast_assigned_ids assignedids,
const struct ast_channel requestor,
const char *  data,
int *  cause 
)
static

Function called by core to create a new outgoing PJSIP session.

Definition at line 2656 of file chan_pjsip.c.

References ao2_cleanup, AST_CAUSE_FAILURE, ast_channel_name(), ast_sip_push_task_wait_servant(), AST_STATE_DOWN, ast_str_tmp, ast_stream_topology_to_str(), request_data::cause, chan_pjsip_new(), ast_sip_session::channel, request_data::dest, NULL, RAII_VAR, request(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, request_data::session, and request_data::topology.

Referenced by chan_pjsip_pvt_dtor(), and chan_pjsip_request().

2657 {
2658  struct request_data req_data;
2660  SCOPE_ENTER(1, "%s Topology: %s\n", data,
2661  ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
2662 
2663  req_data.topology = topology;
2664  req_data.dest = data;
2665  /* Default failure value in case ast_sip_push_task_wait_servant() itself fails. */
2666  req_data.cause = AST_CAUSE_FAILURE;
2667 
2668  if (ast_sip_push_task_wait_servant(NULL, request, &req_data)) {
2669  *cause = req_data.cause;
2670  SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't push task\n");
2671  }
2672 
2673  session = req_data.session;
2674 
2675  if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) {
2676  /* Session needs to be terminated prematurely */
2677  SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
2678  }
2679 
2680  SCOPE_EXIT_RTN_VALUE(session->channel, "Channel: %s\n", ast_channel_name(session->channel));
2681 }
static struct ast_channel * chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name)
Function called to create a new PJSIP Asterisk channel.
Definition: chan_pjsip.c:547
#define NULL
Definition: resample.c:96
int ast_sip_push_task_wait_servant(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Push a task to SIP servants and wait for it to complete.
Definition: res_pjsip.c:5089
A structure describing a SIP session.
#define ast_str_tmp(init_len, __expr)
Definition: strings.h:1136
const char * ast_stream_topology_to_str(const struct ast_stream_topology *topology, struct ast_str **buf)
Get a string representing the topology for debugging/display purposes.
Definition: stream.c:936
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition: utils.h:851
static struct ast_mansession session
#define AST_CAUSE_FAILURE
Definition: causes.h:149
struct ast_stream_topology * topology
Definition: chan_pjsip.c:2543
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
static int request(void *obj)
Definition: chan_pjsip.c:2548
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
const char * ast_channel_name(const struct ast_channel *chan)

◆ chan_pjsip_sendtext()

static int chan_pjsip_sendtext ( struct ast_channel ast,
const char *  text 
)
static

Definition at line 2826 of file chan_pjsip.c.

References ARRAY_LEN, ast_free, ast_msg_data_alloc(), AST_MSG_DATA_ATTR_BODY, AST_MSG_DATA_SOURCE_TYPE_UNKNOWN, chan_pjsip_sendtext_data(), and ast_msg_data_attribute::type.

Referenced by chan_pjsip_pvt_dtor().

2827 {
2828  struct ast_msg_data *msg;
2829  int rc;
2830  struct ast_msg_data_attribute attrs[] =
2831  {
2832  {
2834  .value = (char *)text,
2835  }
2836  };
2837 
2839  if (!msg) {
2840  return -1;
2841  }
2842  rc = chan_pjsip_sendtext_data(ast, msg);
2843  ast_free(msg);
2844 
2845  return rc;
2846 }
#define ARRAY_LEN(a)
Definition: isdn_lib.c:42
enum ast_msg_data_attribute_type type
Definition: message.h:463
Structure used to transport a message through the frame core.
Definition: message.c:1369
struct ast_msg_data * ast_msg_data_alloc(enum ast_msg_data_source_type source, struct ast_msg_data_attribute attributes[], size_t count)
Allocates an ast_msg_data structure.
Definition: message.c:1381
char * text
Definition: app_queue.c:1511
static int chan_pjsip_sendtext_data(struct ast_channel *ast, struct ast_msg_data *msg)
Function called by core to send text on PJSIP session.
Definition: chan_pjsip.c:2804
#define ast_free(a)
Definition: astmm.h:182

◆ chan_pjsip_sendtext_data()

static int chan_pjsip_sendtext_data ( struct ast_channel ast,
struct ast_msg_data msg 
)
static

Function called by core to send text on PJSIP session.

Definition at line 2804 of file chan_pjsip.c.

References ao2_ref, ast_channel_name(), ast_channel_tech_pvt(), ast_debug, AST_MSG_DATA_ATTR_BODY, AST_MSG_DATA_ATTR_FROM, AST_MSG_DATA_ATTR_TO, ast_msg_data_get_attribute(), ast_sip_push_task(), sendtext(), sendtext_data_create(), ast_sip_session::serializer, and ast_sip_channel_pvt::session.

Referenced by chan_pjsip_pvt_dtor(), and chan_pjsip_sendtext().

2805 {
2807  struct sendtext_data *data = sendtext_data_create(ast, msg);
2808 
2809  ast_debug(1, "Sending MESSAGE from '%s' to '%s:%s': %s\n",
2812  ast_channel_name(ast),
2814 
2815  if (!data) {
2816  return -1;
2817  }
2818 
2819  if (ast_sip_push_task(channel->session->serializer, sendtext, data)) {
2820  ao2_ref(data, -1);
2821  return -1;
2822  }
2823  return 0;
2824 }
static struct sendtext_data * sendtext_data_create(struct ast_channel *chan, struct ast_msg_data *msg)
Definition: chan_pjsip.c:2712
void * ast_channel_tech_pvt(const struct ast_channel *chan)
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
#define ast_debug(level,...)
Log a DEBUG message.
Definition: logger.h:444
#define ao2_ref(o, delta)
Definition: astobj2.h:464
struct ast_taskprocessor * serializer
int ast_sip_push_task(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Pushes a task to SIP servants.
Definition: res_pjsip.c:5023
const char * ast_msg_data_get_attribute(struct ast_msg_data *msg, enum ast_msg_data_attribute_type attribute_type)
Get attribute from ast_msg_data.
Definition: message.c:1496
const char * ast_channel_name(const struct ast_channel *chan)
static int sendtext(void *obj)
Definition: chan_pjsip.c:2733

◆ chan_pjsip_session_begin()

static void chan_pjsip_session_begin ( struct ast_sip_session session)
static

SIP session interaction functions.

Definition at line 2939 of file chan_pjsip.c.

References ao2_cleanup, AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE, ast_sip_session_add_datastore(), ast_sip_session_alloc_datastore(), ast_sip_session_get_name(), ast_sip_endpoint_media_configuration::direct_media, ast_sip_session::endpoint, ast_sip_direct_media_configuration::glare_mitigation, ast_sip_endpoint::media, NULL, RAII_VAR, SCOPE_ENTER, and SCOPE_EXIT_RTN.

2940 {
2941  RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
2942  SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
2943 
2944  if (session->endpoint->media.direct_media.glare_mitigation ==
2946  SCOPE_EXIT_RTN("Direct media no glare mitigation\n");
2947  }
2948 
2950  "direct_media_glare_mitigation");
2951 
2952  if (!datastore) {
2953  SCOPE_EXIT_RTN("Couldn't create datastore\n");
2954  }
2955 
2956  ast_sip_session_add_datastore(session, datastore);
2957  SCOPE_EXIT_RTN();
2958 }
struct ast_sip_endpoint * endpoint
Structure for a data store object.
Definition: datastore.h:68
#define NULL
Definition: resample.c:96
static struct ast_datastore_info direct_media_mitigation_info
Definition: chan_pjsip.c:266
struct ast_sip_direct_media_configuration direct_media
Definition: res_pjsip.h:766
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
struct ast_datastore * ast_sip_session_alloc_datastore(const struct ast_datastore_info *info, const char *uid)
Alternative for ast_datastore_alloc()
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition: utils.h:851
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
#define SCOPE_EXIT_RTN(...)
Definition: logger.h:900
enum ast_sip_direct_media_glare_mitigation glare_mitigation
Definition: res_pjsip.h:735
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
int ast_sip_session_add_datastore(struct ast_sip_session *session, struct ast_datastore *datastore)
Add a datastore to a SIP session.

◆ chan_pjsip_session_end()

static void chan_pjsip_session_end ( struct ast_sip_session session)
static

Function called when the session ends.

Definition at line 2961 of file chan_pjsip.c.

References ast_channel_hangupcause(), ast_channel_name(), ast_channel_uniqueid(), ast_queue_hangup(), ast_queue_hangup_with_cause(), ast_set_hangupsource(), ast_sip_session_get_name(), chan_pjsip_remove_hold(), ast_sip_session::channel, hangup_sip2cause(), ast_sip_session::inv_session, SCOPE_ENTER, and SCOPE_EXIT_RTN.

2962 {
2963  SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
2964 
2965  if (!session->channel) {
2966  SCOPE_EXIT_RTN("No channel\n");
2967  }
2968 
2970 
2971  ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
2972  if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
2973  int cause = hangup_sip2cause(session->inv_session->cause);
2974 
2975  ast_queue_hangup_with_cause(session->channel, cause);
2976  } else {
2977  ast_queue_hangup(session->channel);
2978  }
2979 
2980  SCOPE_EXIT_RTN();
2981 }
int ast_queue_hangup(struct ast_channel *chan)
Queue a hangup frame.
Definition: channel.c:1146
struct pjsip_inv_session * inv_session
static void chan_pjsip_remove_hold(const char *chan_uid)
Remove a channel ID from the list of PJSIP channels on hold.
Definition: chan_pjsip.c:1148
int ast_queue_hangup_with_cause(struct ast_channel *chan, int cause)
Queue a hangup frame with hangupcause set.
Definition: channel.c:1162
static int hangup_sip2cause(int cause)
Convert SIP hangup causes to Asterisk hangup causes.
Definition: chan_pjsip.c:2849
void ast_set_hangupsource(struct ast_channel *chan, const char *source, int force)
Set the source of the hangup in this channel and it&#39;s bridge.
Definition: channel.c:2490
struct ast_channel * channel
const char * ast_channel_uniqueid(const struct ast_channel *chan)
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
#define SCOPE_EXIT_RTN(...)
Definition: logger.h:900
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
int ast_channel_hangupcause(const struct ast_channel *chan)
const char * ast_channel_name(const struct ast_channel *chan)

◆ chan_pjsip_set_rtp_peer()

static int chan_pjsip_set_rtp_peer ( struct ast_channel chan,
struct ast_rtp_instance rtp,
struct ast_rtp_instance vrtp,
struct ast_rtp_instance tpeer,
const struct ast_format_cap cap,
int  nat_active 
)
static

Function called by RTP engine to change where the remote party should send media.

Definition at line 448 of file chan_pjsip.c.

References ao2_ref, ast_channel_is_bridged(), ast_channel_name(), ast_channel_tech_pvt(), ast_debug, ast_format_cap_get_names(), AST_FORMAT_CAP_NAMES_LEN, ast_log, ast_sip_push_task(), ast_str_tmp, cdata(), ast_sip_endpoint_media_configuration::direct_media, ast_sip_direct_media_configuration::disable_on_nat, ast_sip_session::endpoint, LOG_ERROR, ast_sip_endpoint::media, rtp_direct_media_data_create(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, send_direct_media_request(), ast_sip_session::serializer, and ast_sip_channel_pvt::session.

454 {
456  struct ast_sip_session *session = channel->session;
458  SCOPE_ENTER(1, "%s %s\n", ast_channel_name(chan),
460 
461  /* Don't try to do any direct media shenanigans on early bridges */
462  if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) {
463  ast_debug(4, "Disregarding setting RTP on %s: channel is not bridged\n", ast_channel_name(chan));
464  SCOPE_EXIT_RTN_VALUE(0, "Channel not bridged\n");
465  }
466 
467  if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
468  ast_debug(4, "Disregarding setting RTP on %s: NAT is active\n", ast_channel_name(chan));
469  SCOPE_EXIT_RTN_VALUE(0, "NAT is active\n");
470  }
471 
472  cdata = rtp_direct_media_data_create(chan, rtp, vrtp, cap, session);
473  if (!cdata) {
475  }
476 
478  ast_log(LOG_ERROR, "Unable to send direct media request for channel %s\n", ast_channel_name(chan));
479  ao2_ref(cdata, -1);
480  }
481 
483 }
struct ast_sip_endpoint * endpoint
void * ast_channel_tech_pvt(const struct ast_channel *chan)
#define AST_FORMAT_CAP_NAMES_LEN
Definition: format_cap.h:326
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
struct ast_sip_direct_media_configuration direct_media
Definition: res_pjsip.h:766
A structure describing a SIP session.
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
#define ast_str_tmp(init_len, __expr)
Definition: strings.h:1136
#define ast_debug(level,...)
Log a DEBUG message.
Definition: logger.h:444
#define ast_log
Definition: astobj2.c:42
static struct ast_mansession session
#define ao2_ref(o, delta)
Definition: astobj2.h:464
static int send_direct_media_request(void *data)
Definition: chan_pjsip.c:396
struct ast_taskprocessor * serializer
#define LOG_ERROR
Definition: logger.h:285
int ast_sip_push_task(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Pushes a task to SIP servants.
Definition: res_pjsip.c:5023
const char * ast_format_cap_get_names(const struct ast_format_cap *cap, struct ast_str **buf)
Get the names of codecs of a set of formats.
Definition: format_cap.c:736
int ast_channel_is_bridged(const struct ast_channel *chan)
Determine if a channel is in a bridge.
Definition: channel.c:10603
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
static int cdata(void *userdata, int state, const char *cdata, size_t len)
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
static struct rtp_direct_media_data * rtp_direct_media_data_create(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, const struct ast_format_cap *cap, struct ast_sip_session *session)
Definition: chan_pjsip.c:377
const char * ast_channel_name(const struct ast_channel *chan)

◆ chan_pjsip_transfer()

static int chan_pjsip_transfer ( struct ast_channel ast,
const char *  target 
)
static

Function called by core for Asterisk initiated transfer.

Definition at line 2120 of file chan_pjsip.c.

References ao2_cleanup, ast_channel_tech_pvt(), ast_log, ast_sip_push_task(), LOG_WARNING, ast_sip_session::serializer, ast_sip_channel_pvt::session, transfer(), and transfer_data_alloc().

Referenced by chan_pjsip_pvt_dtor().

2121 {
2123  struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
2124 
2125  if (!trnf_data) {
2126  return -1;
2127  }
2128 
2129  if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
2130  ast_log(LOG_WARNING, "Error requesting transfer\n");
2131  ao2_cleanup(trnf_data);
2132  return -1;
2133  }
2134 
2135  return 0;
2136 }
static int transfer(void *data)
Definition: chan_pjsip.c:2085
void * ast_channel_tech_pvt(const struct ast_channel *chan)
#define LOG_WARNING
Definition: logger.h:274
static struct transfer_data * transfer_data_alloc(struct ast_sip_session *session, const char *target)
Definition: chan_pjsip.c:1856
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
#define ast_log
Definition: astobj2.c:42
struct ast_taskprocessor * serializer
int ast_sip_push_task(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Pushes a task to SIP servants.
Definition: res_pjsip.c:5023
#define ao2_cleanup(obj)
Definition: astobj2.h:1958

◆ chan_pjsip_write()

static int chan_pjsip_write ( struct ast_channel ast,
struct ast_frame f 
)
static

Definition at line 1039 of file chan_pjsip.c.

References chan_pjsip_write_stream().

Referenced by chan_pjsip_pvt_dtor().

1040 {
1041  return chan_pjsip_write_stream(ast, -1, frame);
1042 }
static int chan_pjsip_write_stream(struct ast_channel *ast, int stream_num, struct ast_frame *f)
Definition: chan_pjsip.c:949

◆ chan_pjsip_write_stream()

static int chan_pjsip_write_stream ( struct ast_channel ast,
int  stream_num,
struct ast_frame f 
)
static

Definition at line 949 of file chan_pjsip.c.

References ast_sip_session::active_media_state, ast_channel_name(), ast_channel_nativeformats(), ast_channel_rawreadformat(), ast_channel_rawwriteformat(), ast_channel_readformat(), ast_channel_readtrans(), ast_channel_tech_pvt(), ast_channel_writeformat(), ast_channel_writetrans(), ast_codec_media_type2str(), ast_debug, ast_format_cap_get_names(), ast_format_cap_iscompatible_format(), AST_FORMAT_CAP_NAMES_LEN, AST_FORMAT_CMP_NOT_EQUAL, ast_format_get_name(), AST_FRAME_CNG, AST_FRAME_MODEM, AST_FRAME_RTCP, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_IMAGE, AST_MEDIA_TYPE_VIDEO, AST_RTP_RTCP_PSFB, ast_str_alloca, ast_translate_path_to_str(), AST_VECTOR_GET, AST_VECTOR_SIZE, ast_sip_session_media_state::default_session, ast_frame_subclass::format, ast_frame::frametype, ast_frame_subclass::integer, LOG_WARNING, NULL, ast_sip_channel_pvt::session, ast_frame::subclass, ast_sip_session_media::type, and ast_sip_session_media::write_callback.

Referenced by chan_pjsip_pvt_dtor(), and chan_pjsip_write().

950 {
952  struct ast_sip_session *session = channel->session;
953  struct ast_sip_session_media *media = NULL;
954  int res = 0;
955 
956  /* The core provides a guarantee that the stream will exist when we are called if stream_num is provided */
957  if (stream_num >= 0) {
958  /* What is not guaranteed is that a media session will exist */
959  if (stream_num < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions)) {
960  media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, stream_num);
961  }
962  }
963 
964  switch (frame->frametype) {
965  case AST_FRAME_VOICE:
966  if (!media) {
967  return 0;
968  } else if (media->type != AST_MEDIA_TYPE_AUDIO) {
969  ast_debug(3, "Channel %s stream %d is of type '%s', not audio!\n",
971  return 0;
972  } else if (media == channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO] &&
975  struct ast_str *write_transpath = ast_str_alloca(256);
976  struct ast_str *read_transpath = ast_str_alloca(256);
977 
979  "Channel %s asked to send %s frame when native formats are %s (rd:%s->%s;%s wr:%s->%s;%s)\n",
980  ast_channel_name(ast),
981  ast_format_get_name(frame->subclass.format),
985  ast_translate_path_to_str(ast_channel_readtrans(ast), &read_transpath),
988  ast_translate_path_to_str(ast_channel_writetrans(ast), &write_transpath));
989  return 0;
990  } else if (media->write_callback) {
991  res = media->write_callback(session, media, frame);
992 
993  }
994  break;
995  case AST_FRAME_VIDEO:
996  if (!media) {
997  return 0;
998  } else if (media->type != AST_MEDIA_TYPE_VIDEO) {
999  ast_debug(3, "Channel %s stream %d is of type '%s', not video!\n",
1000  ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
1001  return 0;
1002  } else if (media->write_callback) {
1003  res = media->write_callback(session, media, frame);
1004  }
1005  break;
1006  case AST_FRAME_MODEM:
1007  if (!media) {
1008  return 0;
1009  } else if (media->type != AST_MEDIA_TYPE_IMAGE) {
1010  ast_debug(3, "Channel %s stream %d is of type '%s', not image!\n",
1011  ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
1012  return 0;
1013  } else if (media->write_callback) {
1014  res = media->write_callback(session, media, frame);
1015  }
1016  break;
1017  case AST_FRAME_CNG:
1018  break;
1019  case AST_FRAME_RTCP:
1020  /* We only support writing out feedback */
1021  if (frame->subclass.integer != AST_RTP_RTCP_PSFB || !media) {
1022  return 0;
1023  } else if (media->type != AST_MEDIA_TYPE_VIDEO) {
1024  ast_debug(3, "Channel %s stream %d is of type '%s', not video! Unable to write RTCP feedback.\n",
1025  ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
1026  return 0;
1027  } else if (media->write_callback) {
1028  res = media->write_callback(session, media, frame);
1029  }
1030  break;
1031  default:
1032  ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP\n", frame->frametype);
1033  break;
1034  }
1035 
1036  return res;
1037 }
void * ast_channel_tech_pvt(const struct ast_channel *chan)
#define LOG_WARNING
Definition: logger.h:274
#define AST_FORMAT_CAP_NAMES_LEN
Definition: format_cap.h:326
const char * ast_codec_media_type2str(enum ast_media_type type)
Conversion function to take a media type and turn it into a string.
Definition: codec.c:347
struct ast_sip_session_media * default_session[AST_MEDIA_TYPE_END]
Default media sessions for each type.
ast_sip_session_media_write_cb write_callback
The write callback when writing frames.
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
#define ast_str_alloca(init_len)
Definition: strings.h:800
const char * ast_format_get_name(const struct ast_format *format)
Get the name associated with a format.
Definition: format.c:334
#define NULL
Definition: resample.c:96
A structure describing a SIP session.
struct ast_trans_pvt * ast_channel_readtrans(const struct ast_channel *chan)
struct ast_sip_session_media_state * active_media_state
struct ast_format * ast_channel_readformat(struct ast_channel *chan)
enum ast_format_cmp_res ast_format_cap_iscompatible_format(const struct ast_format_cap *cap, const struct ast_format *format)
Find if ast_format is within the capabilities of the ast_format_cap object.
Definition: format_cap.c:583
#define ast_debug(level,...)
Log a DEBUG message.
Definition: logger.h:444
#define ast_log
Definition: astobj2.c:42
struct ast_trans_pvt * ast_channel_writetrans(const struct ast_channel *chan)
static struct ast_mansession session
int stream_num
The stream number to place into any resulting frames.
struct ast_format * ast_channel_rawreadformat(struct ast_channel *chan)
The descriptor of a dynamic string XXX storage will be optimized later if needed We use the ts field ...
Definition: strings.h:584
const char * ast_format_cap_get_names(const struct ast_format_cap *cap, struct ast_str **buf)
Get the names of codecs of a set of formats.
Definition: format_cap.c:736
A structure containing SIP session media information.
struct ast_format_cap * ast_channel_nativeformats(const struct ast_channel *chan)
#define AST_VECTOR_GET(vec, idx)
Get an element from a vector.
Definition: vector.h:682
enum ast_media_type type
Media type of this session media.
const char * ast_translate_path_to_str(struct ast_trans_pvt *t, struct ast_str **str)
Puts a string representation of the translation path into outbuf.
Definition: translate.c:922
const char * ast_channel_name(const struct ast_channel *chan)
#define AST_RTP_RTCP_PSFB
Definition: rtp_engine.h:299
struct ast_format * ast_channel_writeformat(struct ast_channel *chan)
struct ast_format * ast_channel_rawwriteformat(struct ast_channel *chan)
#define AST_VECTOR_SIZE(vec)
Get the number of elements in a vector.
Definition: vector.h:611

◆ check_for_rtp_changes()

static int check_for_rtp_changes ( struct ast_channel chan,
struct ast_rtp_instance rtp,
struct ast_sip_session_media media,
struct ast_sip_session session 
)
static
Precondition
chan is locked

Definition at line 319 of file chan_pjsip.c.

References ast_channel_set_fd(), AST_EXTENDED_FDS, ast_rtp_instance_fd(), ast_rtp_instance_get_and_cmp_remote_address, ast_rtp_instance_set_last_rx(), ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_RTCP, ast_sockaddr_isnull(), ast_sockaddr_setnull(), ast_sip_session_media::direct_media_addr, NULL, ast_sip_session_media::rtp, and rtp_find_rtcp_fd_position().

Referenced by send_direct_media_request().

321 {
322  int changed = 0, position = -1;
323 
324  if (media->rtp) {
325  position = rtp_find_rtcp_fd_position(session, media->rtp);
326  }
327 
328  if (rtp) {
330  if (media->rtp) {
331  if (position != -1) {
332  ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, -1);
333  }
335  }
336  } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
338  changed = 1;
339  if (media->rtp) {
340  /* Direct media has ended - reset time of last received RTP packet
341  * to avoid premature RTP timeout. Synchronisation between the
342  * modification of direct_mdedia_addr+last_rx here and reading the
343  * values in res_pjsip_sdp_rtp.c:rtp_check_timeout() is provided
344  * by the channel's lock (which is held while this function is
345  * executed).
346  */
347  ast_rtp_instance_set_last_rx(media->rtp, time(NULL));
349  if (position != -1) {
350  ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, ast_rtp_instance_fd(media->rtp, 1));
351  }
352  }
353  }
354 
355  return changed;
356 }
struct ast_sockaddr direct_media_addr
Direct media address.
#define NULL
Definition: resample.c:96
static void ast_sockaddr_setnull(struct ast_sockaddr *addr)
Sets address addr to null.
Definition: netsock2.h:140
static int ast_sockaddr_isnull(const struct ast_sockaddr *addr)
Checks if the ast_sockaddr is null. "null" in this sense essentially means uninitialized, or having a 0 length.
Definition: netsock2.h:127
#define AST_EXTENDED_FDS
Definition: channel.h:196
int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp)
Get the file descriptor for an RTP session (or RTCP)
Definition: rtp_engine.c:2192
void ast_channel_set_fd(struct ast_channel *chan, int which, int fd)
Definition: channel.c:2417
struct ast_rtp_instance * rtp
RTP instance itself.
void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
Set the value of an RTP instance property.
Definition: rtp_engine.c:705
static int rtp_find_rtcp_fd_position(struct ast_sip_session *session, struct ast_rtp_instance *rtp)
Helper function to find the position for RTCP.
Definition: chan_pjsip.c:298
#define ast_rtp_instance_get_and_cmp_remote_address(instance, address)
Get the address of the remote endpoint that we are sending RTP to, comparing its address to another...
Definition: rtp_engine.h:1228
void ast_rtp_instance_set_last_rx(struct ast_rtp_instance *rtp, time_t time)
Set the last RTP reception time.
Definition: rtp_engine.c:3773

◆ clear_session_and_channel()

static void clear_session_and_channel ( struct ast_sip_session session,
struct ast_channel ast 
)
static

Clear a channel from a session along with its PVT.

Definition at line 2465 of file chan_pjsip.c.

References ast_channel_tech_pvt_set(), ast_sip_session::channel, NULL, and set_channel_on_rtp_instance().

Referenced by chan_pjsip_hangup(), and hangup().

2466 {
2467  session->channel = NULL;
2468  set_channel_on_rtp_instance(session, "");
2470 }
#define NULL
Definition: resample.c:96
static void set_channel_on_rtp_instance(const struct ast_sip_session *session, const char *channel_id)
Definition: chan_pjsip.c:494
struct ast_channel * channel
void ast_channel_tech_pvt_set(struct ast_channel *chan, void *value)

◆ compatible_formats_exist()

static int compatible_formats_exist ( struct ast_stream_topology top,
struct ast_format_cap cap 
)
static

Determine if a topology is compatible with format capabilities.

This will return true if ANY formats in the topology are compatible with the format capabilities.

XXX When supporting true multistream, we will need to be sure to mark which streams from top1 are compatible with which streams from top2. Then the ones that are not compatible will need to be marked as "removed" so that they are negotiated as expected.

Parameters
topTopology
capFormat capabilities
Return values
1The topology has at least one compatible format
0The topology has no compatible formats or an error occurred.

Definition at line 526 of file chan_pjsip.c.

References ao2_ref, ast_format_cap_get_names(), ast_format_cap_iscompatible(), AST_FORMAT_CAP_NAMES_LEN, ast_str_tmp, ast_stream_topology_get_formats(), ast_stream_topology_to_str(), SCOPE_ENTER, and SCOPE_EXIT_RTN_VALUE.

Referenced by chan_pjsip_new().

527 {
528  struct ast_format_cap *cap_from_top;
529  int res;
530  SCOPE_ENTER(1, "Topology: %s Formats: %s\n",
533 
534  cap_from_top = ast_stream_topology_get_formats(top);
535 
536  if (!cap_from_top) {
537  SCOPE_EXIT_RTN_VALUE(0, "Topology had no formats\n");
538  }
539 
540  res = ast_format_cap_iscompatible(cap_from_top, cap);
541  ao2_ref(cap_from_top, -1);
542 
543  SCOPE_EXIT_RTN_VALUE(res, "Compatible? %s\n", res ? "yes" : "no");
544 }
struct ast_format_cap * ast_stream_topology_get_formats(struct ast_stream_topology *topology)
Create a format capabilities structure representing the topology.
Definition: stream.c:930
#define AST_FORMAT_CAP_NAMES_LEN
Definition: format_cap.h:326
#define ast_str_tmp(init_len, __expr)
Definition: strings.h:1136
const char * ast_stream_topology_to_str(const struct ast_stream_topology *topology, struct ast_str **buf)
Get a string representing the topology for debugging/display purposes.
Definition: stream.c:936
#define ao2_ref(o, delta)
Definition: astobj2.h:464
Format capabilities structure, holds formats + preference order + etc.
Definition: format_cap.c:54
const char * ast_format_cap_get_names(const struct ast_format_cap *cap, struct ast_str **buf)
Get the names of codecs of a set of formats.
Definition: format_cap.c:736
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
int ast_format_cap_iscompatible(const struct ast_format_cap *cap1, const struct ast_format_cap *cap2)
Determine if any joint capabilities exist between two capabilities structures.
Definition: format_cap.c:655

◆ direct_media_mitigate_glare()

static int direct_media_mitigate_glare ( struct ast_sip_session session)
static

Definition at line 268 of file chan_pjsip.c.

References ao2_cleanup, AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING, AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE, AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING, ast_sip_session_get_datastore(), ast_sip_session_remove_datastore(), ast_sip_endpoint_media_configuration::direct_media, ast_sip_session::endpoint, ast_sip_direct_media_configuration::glare_mitigation, ast_sip_session::inv_session, ast_sip_endpoint::media, NULL, and RAII_VAR.

Referenced by send_direct_media_request().

269 {
270  RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
271 
274  return 0;
275  }
276 
277  datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
278  if (!datastore) {
279  return 0;
280  }
281 
282  /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
283  ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
284 
285  if ((session->endpoint->media.direct_media.glare_mitigation ==
287  session->inv_session->role == PJSIP_ROLE_UAC) ||
290  session->inv_session->role == PJSIP_ROLE_UAS)) {
291  return 1;
292  }
293 
294  return 0;
295 }
struct ast_sip_endpoint * endpoint
Structure for a data store object.
Definition: datastore.h:68
#define NULL
Definition: resample.c:96
struct pjsip_inv_session * inv_session
struct ast_sip_direct_media_configuration direct_media
Definition: res_pjsip.h:766
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:841
void ast_sip_session_remove_datastore(struct ast_sip_session *session, const char *name)
Remove a session datastore from the session.
struct ast_datastore * ast_sip_session_get_datastore(struct ast_sip_session *session, const char *name)
Retrieve a session datastore.
#define RAII_VAR(vartype, varname, initval, dtor)
Declare a variable that will call a destructor function when it goes out of scope.
Definition: utils.h:851
enum ast_sip_direct_media_glare_mitigation glare_mitigation
Definition: res_pjsip.h:735
#define ao2_cleanup(obj)
Definition: astobj2.h:1958

◆ handle_topology_request_change()

static int handle_topology_request_change ( struct ast_sip_session session,
const struct ast_stream_topology proposed 
)
static

Definition at line 1592 of file chan_pjsip.c.

References ast_sip_push_task(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, send_topology_change_refresh(), ast_sip_session::serializer, topology_change_refresh_data_alloc(), and topology_change_refresh_data_free().

Referenced by chan_pjsip_indicate().

1594 {
1596  int res;
1597  SCOPE_ENTER(1);
1598 
1599  refresh_data = topology_change_refresh_data_alloc(session, proposed);
1600  if (!refresh_data) {
1601  SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create refresh_data\n");
1602  }
1603 
1604  res = ast_sip_push_task(session->serializer, send_topology_change_refresh, refresh_data);
1605  if (res) {
1606  topology_change_refresh_data_free(refresh_data);
1607  }
1608  SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
1609 }
static struct topology_change_refresh_data * topology_change_refresh_data_alloc(struct ast_sip_session *session, const struct ast_stream_topology *topology)
Definition: chan_pjsip.c:1524
static int send_topology_change_refresh(void *data)
Definition: chan_pjsip.c:1575
struct ast_taskprocessor * serializer
int ast_sip_push_task(struct ast_taskprocessor *serializer, int(*sip_task)(void *), void *task_data)
Pushes a task to SIP servants.
Definition: res_pjsip.c:5023
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
static void topology_change_refresh_data_free(struct topology_change_refresh_data *refresh_data)
Definition: chan_pjsip.c:1516

◆ hangup()

static int hangup ( void *  data)
static

Definition at line 2472 of file chan_pjsip.c.

References ao2_bump, ao2_cleanup, ast_channel_name(), ast_channel_tech_pvt(), ast_sip_session_terminate(), hangup_data::cause, hangup_data::chan, clear_session_and_channel(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, and ast_sip_channel_pvt::session.

Referenced by ast_hangup(), chan_pjsip_hangup(), hangup_playback(), and manage_calls().

2473 {
2474  struct hangup_data *h_data = data;
2475  struct ast_channel *ast = h_data->chan;
2477  SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
2478 
2479  /*
2480  * Before cleaning we have to ensure that channel or its session is not NULL
2481  * we have seen rare case when taskprocessor calls hangup but channel is NULL
2482  * due to SIP session timeout and answer happening at the same time
2483  */
2484  if (channel) {
2485  struct ast_sip_session *session = channel->session;
2486  if (session) {
2487  int cause = h_data->cause;
2488 
2489  /*
2490  * It's possible that session_terminate might cause the session to be destroyed
2491  * immediately so we need to keep a reference to it so we can NULL session->channel
2492  * afterwards.
2493  */
2494  ast_sip_session_terminate(ao2_bump(session), cause);
2495  clear_session_and_channel(session, ast);
2496  ao2_cleanup(session);
2497  }
2498  ao2_cleanup(channel);
2499  }
2500  ao2_cleanup(h_data);
2502 }
Main Channel structure associated with a channel.
void * ast_channel_tech_pvt(const struct ast_channel *chan)
void ast_sip_session_terminate(struct ast_sip_session *session, int response)
Terminate a session and, if possible, send the provided response code.
A structure which contains a channel implementation and session.
struct ast_sip_session * session
Pointer to session.
Definition: muted.c:95
A structure describing a SIP session.
#define ao2_bump(obj)
Definition: astobj2.h:491
static struct ast_mansession session
struct ast_channel * chan
Definition: chan_pjsip.c:2440
#define SCOPE_ENTER(level,...)
Definition: logger.h:885
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: logger.h:904
#define ao2_cleanup(obj)
Definition: astobj2.h:1958
const char * ast_channel_name(const struct ast_channel *chan)
static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast)
Clear a channel from a session along with its PVT.
Definition: chan_pjsip.c:2465

◆ hangup_cause2sip()

static int hangup_cause2sip ( int  cause)
static

Internal function which translates from Asterisk cause codes to SIP response codes.

Definition at line 2391 of file chan_pjsip.c.

References AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_CALL_REJECTED, AST_CAUSE_CHAN_NOT_IMPLEMENTED, AST_CAUSE_CONGESTION, AST_CAUSE_DESTINATION_OUT_OF_ORDER, AST_CAUSE_FACILITY_REJECTED, AST_CAUSE_FAILURE, AST_CAUSE_INTERWORKING, AST_CAUSE_INVALID_NUMBER_FORMAT, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NO_ROUTE_TRANSIT_NET, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_NORMAL_UNSPECIFIED, AST_CAUSE_NOTDEFINED, AST_CAUSE_NUMBER_CHANGED, AST_CAUSE_SWITCH_CONGESTION, AST_CAUSE_UNALLOCATED, AST_CAUSE_UNREGISTERED, AST_CAUSE_USER_BUSY, and ast_debug.

Referenced by chan_pjsip_hangup().

2392 {
2393  switch (cause) {
2394  case AST_CAUSE_UNALLOCATED: /* 1 */
2395  case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
2396  case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
2397  return 404;
2398  case AST_CAUSE_CONGESTION: /* 34 */
2399  case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
2400  return 503;
2401  case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
2402  return 408;
2403  case AST_CAUSE_NO_ANSWER: /* 19 */
2404  case AST_CAUSE_UNREGISTERED: /* 20 */
2405  return 480;
2406  case AST_CAUSE_CALL_REJECTED: /* 21 */
2407  return 403;
2408  case AST_CAUSE_NUMBER_CHANGED: /* 22 */
2409  return 410;
2410  case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
2411  return 480;
2413  return 484;
2414  case AST_CAUSE_USER_BUSY:
2415  return 486;
2416  case AST_CAUSE_FAILURE:
2417  return 500;
2418  case AST_CAUSE_FACILITY_REJECTED: /* 29 */
2419  return 501;
2421  return 503;
2423  return 502;
2424  case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
2425  return 488;
2426  case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
2427  return 500;
2428  case AST_CAUSE_NOTDEFINED:
2429  default:
2430  ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
2431  return 0;
2432  }
2433 
2434  /* Never reached */
2435  return 0;
2436 }
#define AST_CAUSE_SWITCH_CONGESTION
Definition: causes.h:122
#define AST_CAUSE_UNALLOCATED
Definition: causes.h:97
#define AST_CAUSE_FACILITY_REJECTED
Definition: causes.h:116
#define AST_CAUSE_NO_USER_RESPONSE
Definition: causes.h:107
#define AST_CAUSE_INVALID_NUMBER_FORMAT
Definition: causes.h:115
#define ast_debug(level,...)
Log a DEBUG message.
Definition: logger.h:444
#define AST_CAUSE_NO_ROUTE_TRANSIT_NET
Definition: causes.h:98
#define AST_CAUSE_CHAN_NOT_IMPLEMENTED
Definition: causes.h:131
#define AST_CAUSE_NO_ANSWER
Definition: causes.h:108
#define AST_CAUSE_NOTDEFINED
Definition: causes.h:154
#define AST_CAUSE_DESTINATION_OUT_OF_ORDER
Definition: causes.h:114
#define AST_CAUSE_FAILURE
Definition: causes.h:149
#define AST_CAUSE_NORMAL_UNSPECIFIED
Definition: causes.h:118
#define AST_CAUSE_UNREGISTERED
Definition: causes.h:153
#define AST_CAUSE_NUMBER_CHANGED
Definition: causes.h:111
#define AST_CAUSE_INTERWORKING
Definition: causes.h:145
#define AST_CAUSE_NO_ROUTE_DESTINATION
Definition: causes.h:99
#define AST_CAUSE_USER_BUSY
Definition: causes.h:106
#define AST_CAUSE_CALL_REJECTED
Definition: causes.h:110
#define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL
Definition: causes.h:129
#define AST_CAUSE_CONGESTION
Definition: causes.h:152

◆ hangup_data_alloc()

static struct hangup_data* hangup_data_alloc ( int  cause,
struct ast_channel chan 
)
static

Definition at line 2450 of file chan_pjsip.c.

References ao2_alloc, ast_channel_ref, hangup_data::cause, hangup_data::chan, hangup_data_destroy(), and NULL.

Referenced by chan_pjsip_hangup().

2451 {
2452  struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
2453 
2454  if (!h_data) {
2455  return NULL;
2456  }
2457 
2458  h_data->cause = cause;
2459  h_data->chan = ast_channel_ref(chan);
2460 
2461  return h_data;
2462 }
static void hangup_data_destroy(void *obj)
Definition: chan_pjsip.c:2443
#define NULL
Definition: resample.c:96
struct ast_channel * chan
Definition: chan_pjsip.c:2440
#define ao2_alloc(data_size, destructor_fn)
Definition: astobj2.h:411
#define ast_channel_ref(c)
Increase channel reference count.
Definition: channel.h:2915

◆ hangup_data_destroy()

static void hangup_data_destroy ( void *  obj)
static

Definition at line 2443 of file chan_pjsip.c.

References ast_channel_unref, and hangup_data::chan.

Referenced by hangup_data_alloc().

2444 {
2445  struct hangup_data *h_data = obj;
2446 
2447  h_data->chan = ast_channel_unref(h_data->chan);
2448 }
#define ast_channel_unref(c)
Decrease channel reference count.
Definition: channel.h:2926
struct ast_channel * chan
Definition: chan_pjsip.c:2440

◆ hangup_sip2cause()

static int hangup_sip2cause ( int  cause)
static

Convert SIP hangup causes to Asterisk hangup causes.

Definition at line 2849 of file chan_pjsip.c.

References AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_BUSY, AST_CAUSE_CALL_REJECTED, AST_CAUSE_CONGESTION, AST_CAUSE_DESTINATION_OUT_OF_ORDER, AST_CAUSE_FACILITY_REJECTED, AST_CAUSE_FAILURE, AST_CAUSE_INTERWORKING, AST_CAUSE_INVALID_NUMBER_FORMAT, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_NORMAL, AST_CAUSE_NORMAL_TEMPORARY_FAILURE, AST_CAUSE_NUMBER_CHANGED, AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, AST_CAUSE_UNALLOCATED, and AST_CAUSE_USER_BUSY.

Referenced by chan_pjsip_incoming_response_update_cause(), and chan_pjsip_session_end().

2850 {
2851  /* Possible values taken from causes.h */
2852 
2853  switch(cause) {
2854  case 401: /* Unauthorized */
2855  return AST_CAUSE_CALL_REJECTED;
2856  case 403: /* Not found */
2857  return AST_CAUSE_CALL_REJECTED;
2858  case 404: /* Not found */
2859  return AST_CAUSE_UNALLOCATED;
2860  case 405: /* Method not allowed */
2861  return AST_CAUSE_INTERWORKING;
2862  case 407: /* Proxy authentication required */
2863  return AST_CAUSE_CALL_REJECTED;
2864  case 408: /* No reaction */
2866  case 409: /* Conflict */
2868  case 410: /* Gone */
2869  return AST_CAUSE_NUMBER_CHANGED;
2870  case 411: /* Length required */
2871  return AST_CAUSE_INTERWORKING;
2872  case 413: /* Request entity too large */
2873  return AST_CAUSE_INTERWORKING;
2874  case 414: /* Request URI too large */
2875  return AST_CAUSE_INTERWORKING;
2876  case 415: /* Unsupported media type */
2877  return AST_CAUSE_INTERWORKING;
2878  case 420: /* Bad extension */
2880  case 480: /* No answer */
2881  return AST_CAUSE_NO_ANSWER;
2882  case 481: /* No answer */
2883  return AST_CAUSE_INTERWORKING;
2884  case 482: /* Loop detected */
2885  return AST_CAUSE_INTERWORKING;
2886  case 483: /* Too many hops */
2887  return AST_CAUSE_NO_ANSWER;
2888  case 484: /* Address incomplete */
2890  case 485: /* Ambiguous */
2891  return AST_CAUSE_UNALLOCATED;
2892  case 486: /* Busy everywhere */
2893  return AST_CAUSE_BUSY;
2894  case 487: /* Request terminated */
2895  return AST_CAUSE_INTERWORKING;
2896  case 488: /* No codecs approved */
2898  case 491: /* Request pending */
2899  return AST_CAUSE_INTERWORKING;
2900  case 493: /* Undecipherable */
2901  return AST_CAUSE_INTERWORKING;
2902  case 500: /* Server internal failure */
2903  return AST_CAUSE_FAILURE;
2904  case 501: /* Call rejected */
2906  case 502:
2908  case 503: /* Service unavailable */
2909  return AST_CAUSE_CONGESTION;
2910  case 504: /* Gateway timeout */
2912  case 505: /* SIP version not supported */
2913  return AST_CAUSE_INTERWORKING;
2914  case 600: /* Busy everywhere */
2915  return AST_CAUSE_USER_BUSY;
2916  case 603: /* Decline */
2917  return AST_CAUSE_CALL_REJECTED;
2918  case 604: /* Does not exist anywhere */
2919  return AST_CAUSE_UNALLOCATED;
2920  case 606: /* Not acceptable */
2922  default:
2923  if (cause < 500 && cause >= 400) {
2924  /* 4xx class error that is unknown - someting wrong with our request */
2925  return AST_CAUSE_INTERWORKING;
2926  } else if (cause < 600 && cause >= 500) {
2927  /* 5xx class error - problem in the remote end */
2928  return AST_CAUSE_CONGESTION;
2929  } else if (cause < 700 && cause >= 600) {
2930  /* 6xx - global errors in the 4xx class */
2931  return AST_CAUSE_INTERWORKING;
2932  }
2933  return AST_CAUSE_NORMAL;
2934  }
2935  /* Never reached */
2936  return 0;
2937 }
#define AST_CAUSE_NORMAL_TEMPORARY_FAILURE
Definition: causes.h:121
#define AST_CAUSE_UNALLOCATED
Definition: causes.h:97
#define AST_CAUSE_FACILITY_REJECTED
Definition: causes.h:116
#define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE
Definition: causes.h:142
#define AST_CAUSE_NO_USER_RESPONSE
Definition: causes.h:107
#define AST_CAUSE_INVALID_NUMBER_FORMAT
Definition: causes.h:115
#define AST_CAUSE_NO_ANSWER
Definition: causes.h:108
#define AST_CAUSE_DESTINATION_OUT_OF_ORDER
Definition: causes.h:114
#define AST_CAUSE_NORMAL
Definition: causes.h:150
#define AST_CAUSE_FAILURE
Definition: causes.h:149
#define AST_CAUSE_NUMBER_CHANGED
Definition: causes.h:111
#define AST_CAUSE_INTERWORKING
Definition: causes.h:145
#define AST_CAUSE_NO_ROUTE_DESTINATION
Definition: causes.h:99
#define AST_CAUSE_USER_BUSY
Definition: causes.h:106
#define AST_CAUSE_BUSY