Asterisk - The Open Source Telephony Project GIT-master-0644429
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"
Include dependency graph for chan_pjsip.c:

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_prack (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)
 Asterisk core interaction functions. More...
 
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 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 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 char * app_pjsip_hangup = "PJSIPHangup"
 
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_from_function
 
static struct ast_custom_function chan_pjsip_parse_uri_function
 
static struct ast_sip_session_supplement chan_pjsip_prack_supplement
 
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.

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 3446 of file chan_pjsip.c.

◆ __unreg_module()

static void __unreg_module ( void  )
static

Definition at line 3446 of file chan_pjsip.c.

◆ answer()

static int answer ( void *  data)
static

Definition at line 687 of file chan_pjsip.c.

688{
689 struct answer_data *ans_data = data;
690 pj_status_t status = PJ_SUCCESS;
691 pjsip_tx_data *packet = NULL;
692 struct ast_sip_session *session = ans_data->session;
694
695 if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
696 ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
697 session->inv_session->cause,
698 pjsip_get_status_text(session->inv_session->cause)->ptr);
699 SCOPE_EXIT_RTN_VALUE(0, "Disconnected\n");
700 }
701
702 pjsip_dlg_inc_lock(session->inv_session->dlg);
703 if (session->inv_session->invite_tsx) {
704 status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
705 } else {
706 ast_log(LOG_ERROR,"Cannot answer '%s' because there is no associated SIP transaction\n",
707 ast_channel_name(session->channel));
708 }
709 pjsip_dlg_dec_lock(session->inv_session->dlg);
710
711 if (status == PJ_SUCCESS && packet) {
713 }
714
715 if (status != PJ_SUCCESS) {
716 char err[PJ_ERR_MSG_SIZE];
717
718 pj_strerror(status, err, sizeof(err));
719 ast_log(LOG_WARNING,"Cannot answer '%s': %s\n",
720 ast_channel_name(session->channel), err);
721 /*
722 * Return this value so we can distinguish between this
723 * failure and the threadpool synchronous push failing.
724 */
725 SCOPE_EXIT_RTN_VALUE(-2, "pjproject failure\n");
726 }
728}
jack_status_t status
Definition: app_jack.c:146
static struct ast_mansession session
#define ast_log
Definition: astobj2.c:42
const char * ast_channel_name(const struct ast_channel *chan)
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
#define SCOPE_ENTER_TASK(level, indent,...)
#define LOG_ERROR
#define LOG_WARNING
const char * ast_sip_session_get_name(const struct ast_sip_session *session)
Get the channel or endpoint name associated with the session.
void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
Send a SIP response.
#define NULL
Definition: resample.c:96
struct ast_sip_session * session
Definition: chan_pjsip.c:683
unsigned long indent
Definition: chan_pjsip.c:684
A structure describing a SIP session.

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

Referenced by add_bundle_groups(), add_sdp_streams(), ast_dns_resolver_set_result(), ast_raw_answer_with_stream_topology(), ast_search_dns(), ast_stun_request(), ast_unreal_answer(), chan_pjsip_answer(), dns_parse_answer(), dns_parse_answer_ex(), dump_answer(), ebl_callback(), enum_callback(), parse_naptr(), parse_srv(), pbx_builtin_incomplete(), session_inv_on_rx_offer(), srv_callback(), stun_monitor_request(), tds_log(), txt_callback(), verify_mock_cdr_record(), and zapateller_exec().

◆ AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM ( void  )

Definition at line 3446 of file chan_pjsip.c.

◆ call()

static int call ( void *  data)
static

Definition at line 2395 of file chan_pjsip.c.

2396{
2397 struct ast_sip_channel_pvt *channel = data;
2398 struct ast_sip_session *session = channel->session;
2399 pjsip_tx_data *tdata;
2400 int res = 0;
2401 SCOPE_ENTER(1, "%s Topology: %s\n",
2403 ast_str_tmp(256, ast_stream_topology_to_str(channel->session->pending_media_state->topology, &STR_TMP))
2404 );
2405
2406
2408
2409 if (res) {
2410 ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
2411 ast_queue_hangup(session->channel);
2412 } else {
2416 }
2417 ao2_ref(channel, -1);
2418 SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
2419}
#define ao2_ref(o, delta)
Reference/unreference an object and return the old refcount.
Definition: astobj2.h:459
static void update_initial_connected_line(struct ast_sip_session *session)
Definition: chan_pjsip.c:2370
static void set_channel_on_rtp_instance(const struct ast_sip_session *session, const char *channel_id)
Definition: chan_pjsip.c:494
int ast_queue_hangup(struct ast_channel *chan)
Queue a hangup frame.
Definition: channel.c:1169
const char * ast_channel_uniqueid(const struct ast_channel *chan)
void ast_set_hangupsource(struct ast_channel *chan, const char *source, int force)
Set the source of the hangup in this channel and it's bridge.
Definition: channel.c:2518
#define SCOPE_ENTER(level,...)
void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
Send a SIP request.
int ast_sip_session_create_invite(struct ast_sip_session *session, pjsip_tx_data **tdata)
Creates an INVITE request.
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 ast_str_tmp(init_len, __expr)
Provides a temporary ast_str and returns a copy of its buffer.
Definition: strings.h:1189
A structure which contains a channel implementation and session.
struct ast_channel * channel

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, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, session, set_channel_on_rtp_instance(), and update_initial_connected_line().

Referenced by ast_call(), can_ring_entry(), chan_pjsip_call(), close_rtp_connection(), close_udptl_connection(), configure_local_rtp(), find_call(), native_start(), onAlerting(), onCallCleared(), onCallEstablished(), onModeChanged(), onNewCallCreated(), onOutgoingCall(), onProgress(), ooh323_onReceivedDigit(), ooh323_onReceivedSetup(), ooh323_set_read_format(), ooh323_set_write_format(), ooh323c_set_capability_for_call(), ooh323c_start_call_thread(), ooh323c_start_receive_channel(), ooh323c_start_transmit_channel(), ooh323c_start_transmit_datachannel(), ooh323c_stop_call_thread(), ooh323c_stop_transmit_channel(), ooh323c_stop_transmit_datachannel(), setup_rtp_connection(), setup_rtp_remote(), setup_udptl_connection(), and update_our_aliases().

◆ call_pickup_incoming_request()

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

Definition at line 3030 of file chan_pjsip.c.

3031{
3032 struct ast_features_pickup_config *pickup_cfg;
3033 struct ast_channel *chan;
3034
3035 /* Check for a to-tag to determine if this is a reinvite */
3036 if (rdata->msg_info.to->tag.slen) {
3037 /* We don't care about reinvites */
3038 return 0;
3039 }
3040
3041 pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
3042 if (!pickup_cfg) {
3043 ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n");
3044 return 0;
3045 }
3046
3047 if (strcmp(session->exten, pickup_cfg->pickupexten)) {
3048 ao2_ref(pickup_cfg, -1);
3049 return 0;
3050 }
3051 ao2_ref(pickup_cfg, -1);
3052
3053 /* We can't directly use session->channel because the pickup operation will cause a masquerade to occur,
3054 * changing the channel pointer in session to a different channel. To ensure we work on the right channel
3055 * we store a pointer locally before we begin and keep a reference so it remains valid no matter what.
3056 */
3057 chan = ast_channel_ref(session->channel);
3058 if (ast_pickup_call(chan)) {
3060 } else {
3062 }
3063 /* A hangup always occurs because the pickup operation will have either failed resulting in the call
3064 * needing to be hung up OR the pickup operation was a success and the channel we now have is actually
3065 * the channel that was replaced, which should be hung up since it is literally in limbo not connected
3066 * to anything at all.
3067 */
3068 ast_hangup(chan);
3069 ast_channel_unref(chan);
3070
3071 return 1;
3072}
#define AST_CAUSE_CALL_REJECTED
Definition: causes.h:111
#define AST_CAUSE_NORMAL_CLEARING
Definition: causes.h:106
void ast_hangup(struct ast_channel *chan)
Hang up a channel.
Definition: channel.c:2560
#define ast_channel_ref(c)
Increase channel reference count.
Definition: channel.h:2993
#define ast_channel_unref(c)
Decrease channel reference count.
Definition: channel.h:3004
void ast_channel_hangupcause_set(struct ast_channel *chan, int value)
struct ast_features_pickup_config * ast_get_chan_features_pickup_config(struct ast_channel *chan)
Get the pickup configuration options for a channel.
int ast_pickup_call(struct ast_channel *chan)
Pickup a call.
Definition: pickup.c:199
Main Channel structure associated with a channel.
Configuration relating to call pickup.

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(), LOG_ERROR, ast_features_pickup_config::pickupexten, and session.

◆ 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 1122 of file chan_pjsip.c.

1123{
1124 RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
1125
1126 hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
1127 if (hold_uid) {
1128 /* Device is already on hold. Nothing to do. */
1129 return 0;
1130 }
1131
1132 /* Device wasn't in hold list already. Create a new one. */
1133 hold_uid = ao2_alloc_options(strlen(chan_uid) + 1, NULL,
1135 if (!hold_uid) {
1136 return -1;
1137 }
1138
1139 ast_copy_string(hold_uid, chan_uid, strlen(chan_uid) + 1);
1140
1141 if (ao2_link(pjsip_uids_onhold, hold_uid) == 0) {
1142 return -1;
1143 }
1144
1145 return 0;
1146}
#define ao2_link(container, obj)
Add an object to a container.
Definition: astobj2.h:1532
@ AO2_ALLOC_OPT_LOCK_NOLOCK
Definition: astobj2.h:367
#define ao2_cleanup(obj)
Definition: astobj2.h:1934
#define ao2_find(container, arg, flags)
Definition: astobj2.h:1736
#define ao2_alloc_options(data_size, destructor_fn, options)
Definition: astobj2.h:404
@ OBJ_SEARCH_KEY
The arg parameter is a search key, but is not an object.
Definition: astobj2.h:1101
static struct ao2_container * pjsip_uids_onhold
Definition: chan_pjsip.c:1112
void ast_copy_string(char *dst, const char *src, size_t size)
Size-limited null-terminating string copy.
Definition: strings.h:425
#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:941

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

Referenced by chan_pjsip_indicate().

◆ 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 731 of file chan_pjsip.c.

732{
733 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
734 struct ast_sip_session *session;
735 struct answer_data ans_data = { 0, };
736 int res;
737 SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
738
739 if (ast_channel_state(ast) == AST_STATE_UP) {
740 SCOPE_EXIT_RTN_VALUE(0, "Already up\n");
741 return 0;
742 }
743
745 session = ao2_bump(channel->session);
746
747 /* the answer task needs to be pushed synchronously otherwise a race condition
748 can occur between this thread and bridging (specifically when native bridging
749 attempts to do direct media) */
751 ans_data.session = session;
752 ans_data.indent = ast_trace_get_indent();
753 res = ast_sip_push_task_wait_serializer(session->serializer, answer, &ans_data);
754 if (res) {
755 if (res == -1) {
756 ast_log(LOG_ERROR,"Cannot answer '%s': Unable to push answer task to the threadpool.\n",
757 ast_channel_name(session->channel));
758 }
759 ao2_ref(session, -1);
760 ast_channel_lock(ast);
761 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
762 }
763 ao2_ref(session, -1);
764 ast_channel_lock(ast);
765
767}
#define ao2_bump(obj)
Bump refcount on an AO2 object by one, returning the object.
Definition: astobj2.h:480
static int answer(void *data)
Definition: chan_pjsip.c:687
void * ast_channel_tech_pvt(const struct ast_channel *chan)
#define ast_channel_lock(chan)
Definition: channel.h:2968
#define ast_channel_unlock(chan)
Definition: channel.h:2969
ast_channel_state
ast_channel states
Definition: channelstate.h:35
@ AST_STATE_UP
Definition: channelstate.h:42
int ast_setstate(struct ast_channel *chan, enum ast_channel_state)
Change the state of a channel.
Definition: channel.c:7408
#define ast_trace_get_indent()
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:2179
struct ast_sip_session * session
Pointer to session.

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, answer_data::indent, LOG_ERROR, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, answer_data::session, ast_sip_channel_pvt::session, and session.

◆ 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 2422 of file chan_pjsip.c.

2423{
2424 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2425 SCOPE_ENTER(1, "%s Topology: %s\n", ast_sip_session_get_name(channel->session),
2427
2428 ao2_ref(channel, +1);
2429 if (ast_sip_push_task(channel->session->serializer, call, channel)) {
2430 ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest);
2431 ao2_cleanup(channel);
2432 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
2433 }
2434
2435 SCOPE_EXIT_RTN_VALUE(0, "'call' task pushed\n");
2436}
static int call(void *data)
Definition: chan_pjsip.c:2395
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:2099
struct ast_stream_topology * topology
The media stream topology.
struct ast_sip_session_media_state * pending_media_state
struct ast_taskprocessor * serializer

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.

◆ 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 770 of file chan_pjsip.c.

772{
773 const char *target_context;
774 int exists;
775 int dsp_features;
776
777 dsp_features = ast_dsp_get_features(session->dsp);
778 dsp_features &= ~DSP_FEATURE_FAX_DETECT;
779 if (dsp_features) {
780 ast_dsp_set_features(session->dsp, dsp_features);
781 } else {
782 ast_dsp_free(session->dsp);
783 session->dsp = NULL;
784 }
785
786 /* If already executing in the fax extension don't do anything */
787 if (!strcmp(ast_channel_exten(ast), "fax")) {
788 return f;
789 }
790
791 target_context = ast_channel_context(ast);
792
793 /*
794 * We need to unlock the channel here because ast_exists_extension has the
795 * potential to start and stop an autoservice on the channel. Such action
796 * is prone to deadlock if the channel is locked.
797 *
798 * ast_async_goto() has its own restriction on not holding the channel lock.
799 */
801 ast_frfree(f);
802 f = &ast_null_frame;
803 exists = ast_exists_extension(ast, target_context, "fax", 1,
804 S_COR(ast_channel_caller(ast)->id.number.valid,
805 ast_channel_caller(ast)->id.number.str, NULL));
806 if (exists) {
807 ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
808 ast_channel_name(ast));
809 pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
810 if (ast_async_goto(ast, target_context, "fax", 1)) {
811 ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
812 ast_channel_name(ast), target_context);
813 }
814 } else {
815 ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
816 ast_channel_name(ast), target_context);
817 }
818
819 /* It's possible for a masquerade to have occurred when doing the ast_async_goto resulting in
820 * the channel on the session having changed. Since we need to return with the original channel
821 * locked we lock the channel that was passed in and not session->channel.
822 */
823 ast_channel_lock(ast);
824
825 return f;
826}
const char * ast_channel_context(const struct ast_channel *chan)
struct ast_party_caller * ast_channel_caller(struct ast_channel *chan)
const char * ast_channel_exten(const struct ast_channel *chan)
void ast_dsp_free(struct ast_dsp *dsp)
Definition: dsp.c:1783
int ast_dsp_get_features(struct ast_dsp *dsp)
Get features.
Definition: dsp.c:1777
void ast_dsp_set_features(struct ast_dsp *dsp, int features)
Select feature set.
Definition: dsp.c:1768
static int exists(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
Definition: func_logic.c:157
#define ast_frfree(fr)
struct ast_frame ast_null_frame
Definition: main/frame.c:79
#define ast_verb(level,...)
#define LOG_NOTICE
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:4175
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.
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:6969
#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:87
Number structure.
Definition: app_followme.c:154

References ast_async_goto(), ast_channel_caller(), ast_channel_context(), ast_channel_exten(), ast_channel_lock, 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, exists(), LOG_ERROR, LOG_NOTICE, NULL, pbx_builtin_setvar_helper(), S_COR, and session.

Referenced by chan_pjsip_read_stream().

◆ 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 1179 of file chan_pjsip.c.

1180{
1181 RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
1183 RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
1184 struct ast_devstate_aggregate aggregate;
1185 int num, inuse = 0;
1186
1187 if (!endpoint) {
1188 return AST_DEVICE_INVALID;
1189 }
1190
1191 endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
1192 ast_endpoint_get_resource(endpoint->persistent));
1193
1194 if (!endpoint_snapshot) {
1195 return AST_DEVICE_INVALID;
1196 }
1197
1198 if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
1200 } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
1202 }
1203
1204 if (!endpoint_snapshot->num_channels) {
1205 return state;
1206 }
1207
1208 ast_devstate_aggregate_init(&aggregate);
1209
1210 for (num = 0; num < endpoint_snapshot->num_channels; num++) {
1211 struct ast_channel_snapshot *snapshot;
1212
1213 snapshot = ast_channel_snapshot_get_latest(endpoint_snapshot->channel_ids[num]);
1214 if (!snapshot) {
1215 continue;
1216 }
1217
1218 if (chan_pjsip_get_hold(snapshot->base->uniqueid)) {
1220 } else {
1221 ast_devstate_aggregate_add(&aggregate, ast_state_chan2dev(snapshot->state));
1222 }
1223
1224 if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
1225 (snapshot->state == AST_STATE_BUSY)) {
1226 inuse++;
1227 }
1228
1229 ao2_ref(snapshot, -1);
1230 }
1231
1232 if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
1234 } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
1236 }
1237
1238 return state;
1239}
enum cc_state state
Definition: ccss.c:393
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:1166
@ AST_STATE_RING
Definition: channelstate.h:40
@ AST_STATE_BUSY
Definition: channelstate.h:43
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
void ast_devstate_aggregate_init(struct ast_devstate_aggregate *agg)
Initialize aggregate device state.
Definition: devicestate.c:630
enum ast_device_state ast_devstate_aggregate_result(struct ast_devstate_aggregate *agg)
Get the aggregate device state result.
Definition: devicestate.c:663
enum ast_device_state ast_state_chan2dev(enum ast_channel_state chanstate)
Convert channel state to devicestate.
Definition: devicestate.c:242
ast_device_state
Device States.
Definition: devicestate.h:52
@ AST_DEVICE_UNKNOWN
Definition: devicestate.h:53
@ AST_DEVICE_ONHOLD
Definition: devicestate.h:61
@ AST_DEVICE_INVALID
Definition: devicestate.h:57
@ AST_DEVICE_BUSY
Definition: devicestate.h:56
@ AST_DEVICE_NOT_INUSE
Definition: devicestate.h:54
@ AST_DEVICE_UNAVAILABLE
Definition: devicestate.h:58
@ AST_ENDPOINT_OFFLINE
Definition: endpoints.h:55
@ AST_ENDPOINT_ONLINE
Definition: endpoints.h:57
const char * ast_endpoint_get_tech(const struct ast_endpoint *endpoint)
Gets the technology of the given endpoint.
const char * ast_endpoint_get_resource(const struct ast_endpoint *endpoint)
Gets the resource name of the given endpoint.
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_endpoint_snapshot * ast_endpoint_latest_snapshot(const char *tech, const char *resource)
Retrieve the most recent snapshot for the endpoint with the given name.
struct ast_sorcery * ast_sip_get_sorcery(void)
Get a pointer to the SIP sorcery structure.
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:1853
const ast_string_field uniqueid
Structure representing a snapshot of channel state.
struct ast_channel_snapshot_base * base
enum ast_channel_state state
You shouldn't care about the contents of this struct.
Definition: devicestate.h:228
A snapshot of an endpoint's state.
An entity with which Asterisk communicates.
Definition: res_pjsip.h:961

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.

◆ 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 2187 of file chan_pjsip.c.

2188{
2189 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
2190 struct ast_sip_session_media *media;
2191
2193
2194 switch (channel->session->dtmf) {
2196 if (!media || !media->rtp) {
2197 return 0;
2198 }
2199
2201 break;
2202 case AST_SIP_DTMF_AUTO:
2203 if (!media || !media->rtp) {
2204 return 0;
2205 }
2206
2208 return -1;
2209 }
2210
2212 break;
2214 if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) {
2215 return 0;
2216 }
2218 break;
2219 case AST_SIP_DTMF_NONE:
2220 break;
2222 return -1;
2223 default:
2224 break;
2225 }
2226
2227 return 0;
2228}
char digit
@ AST_MEDIA_TYPE_AUDIO
Definition: codec.h:32
@ AST_SIP_DTMF_NONE
Definition: res_pjsip.h:545
@ AST_SIP_DTMF_AUTO_INFO
Definition: res_pjsip.h:556
@ AST_SIP_DTMF_AUTO
Definition: res_pjsip.h:554
@ AST_SIP_DTMF_INBAND
Definition: res_pjsip.h:550
@ AST_SIP_DTMF_RFC_4733
Definition: res_pjsip.h:548
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:2313
@ AST_RTP_DTMF_MODE_INBAND
Definition: rtp_engine.h:157
@ AST_RTP_DTMF_MODE_NONE
Definition: rtp_engine.h:153
int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit)
Begin sending a DTMF digit.
Definition: rtp_engine.c:2257
struct ast_sip_session_media * default_session[AST_MEDIA_TYPE_END]
Default media sessions for each type.
A structure containing SIP session media information.
struct ast_rtp_instance * rtp
RTP instance itself.
struct ast_sip_session_media_state * active_media_state
enum ast_sip_dtmf_mode dtmf

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, digit, ast_sip_session::dtmf, ast_sip_session_media::rtp, and ast_sip_channel_pvt::session.

◆ 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 2299 of file chan_pjsip.c.

2300{
2301 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2302 struct ast_sip_session_media *media;
2303
2304 if (!channel || !channel->session) {
2305 /* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */
2306 ast_debug(3, "Channel %s disappeared while calling digit_end\n", ast_channel_name(ast));
2307 return -1;
2308 }
2309
2311
2312 switch (channel->session->dtmf) {
2314 {
2315 if (!media || !media->rtp) {
2316 return 0;
2317 }
2318
2320 ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
2322 break;
2323 }
2324 /* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */
2325 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));
2326 }
2327
2328 case AST_SIP_DTMF_INFO:
2329 {
2330 struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
2331
2332 if (!dtmf_data) {
2333 return -1;
2334 }
2335
2336 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
2337 ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
2338 ao2_cleanup(dtmf_data);
2339 return -1;
2340 }
2341 break;
2342 }
2344 if (!media || !media->rtp) {
2345 return 0;
2346 }
2347
2349 break;
2350 case AST_SIP_DTMF_AUTO:
2351 if (!media || !media->rtp) {
2352 return 0;
2353 }
2354
2356 return -1;
2357 }
2358
2360 break;
2361 case AST_SIP_DTMF_NONE:
2362 break;
2364 return -1;
2365 }
2366
2367 return 0;
2368}
static int transmit_info_dtmf(void *data)
Definition: chan_pjsip.c:2255
static struct info_dtmf_data * info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
Definition: chan_pjsip.c:2242
#define ast_debug(level,...)
Log a DEBUG message.
@ AST_SIP_DTMF_INFO
Definition: res_pjsip.h:552
int ast_rtp_instance_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
Definition: rtp_engine.c:2285
unsigned int duration
Definition: chan_pjsip.c:2233

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, digit, ast_sip_session::dtmf, info_dtmf_data::duration, info_dtmf_data_alloc(), LOG_WARNING, ast_sip_session_media::rtp, ast_sip_session::serializer, ast_sip_channel_pvt::session, and transmit_info_dtmf().

◆ 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 1050 of file chan_pjsip.c.

1051{
1052 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
1053
1054 if (channel->session->channel != oldchan) {
1055 return -1;
1056 }
1057
1058 /*
1059 * The masquerade has suspended the channel's session
1060 * serializer so we can safely change it outside of
1061 * the serializer thread.
1062 */
1063 channel->session->channel = newchan;
1064
1066
1067 return 0;
1068}

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

◆ 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 252 of file chan_pjsip.c.

253{
254 SCOPE_ENTER(1, "%s Native formats %s\n", ast_channel_name(chan),
258}
static PGresult * result
Definition: cel_pgsql.c:84
struct ast_format_cap * ast_channel_nativeformats(const struct ast_channel *chan)
@ AST_MEDIA_TYPE_UNKNOWN
Definition: codec.h:31
#define AST_FORMAT_CAP_NAMES_LEN
Definition: format_cap.h:324
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
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:734
#define SCOPE_EXIT_RTN(...)

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, result, SCOPE_ENTER, and SCOPE_EXIT_RTN.

◆ 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 1166 of file chan_pjsip.c.

1167{
1168 RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
1169
1170 hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
1171 if (!hold_uid) {
1172 return 0;
1173 }
1174
1175 return 1;
1176}

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

Referenced by chan_pjsip_devicestate().

◆ 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 179 of file chan_pjsip.c.

180{
181 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
182 struct ast_sip_endpoint *endpoint;
183 struct ast_datastore *datastore;
184 struct ast_sip_session_media *media;
185
186 if (!channel || !channel->session) {
188 }
189
190 /* XXX Getting the first RTP instance for direct media related stuff seems just
191 * absolutely wrong. But the native RTP bridge knows no other method than single-stream
192 * for direct media. So this is the best we can do.
193 */
195 if (!media || !media->rtp) {
197 }
198
199 datastore = ast_sip_session_get_datastore(channel->session, "t38");
200 if (datastore) {
201 ao2_ref(datastore, -1);
203 }
204
205 endpoint = channel->session->endpoint;
206
207 *instance = media->rtp;
208 ao2_ref(*instance, +1);
209
210 ast_assert(endpoint != NULL);
213 }
214
215 if (endpoint->media.direct_media.enabled) {
217 }
218
220}
@ AST_SIP_MEDIA_ENCRYPT_NONE
Definition: res_pjsip.h:647
struct ast_datastore * ast_sip_session_get_datastore(struct ast_sip_session *session, const char *name)
Retrieve a session datastore.
@ AST_RTP_GLUE_RESULT_LOCAL
Definition: rtp_engine.h:167
@ AST_RTP_GLUE_RESULT_REMOTE
Definition: rtp_engine.h:165
@ AST_RTP_GLUE_RESULT_FORBID
Definition: rtp_engine.h:163
Structure for a data store object.
Definition: datastore.h:64
struct ast_sip_media_rtp_configuration rtp
Definition: res_pjsip.h:913
struct ast_sip_direct_media_configuration direct_media
Definition: res_pjsip.h:915
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:996
enum ast_sip_session_media_encryption encryption
Definition: res_pjsip.h:856
struct ast_sip_endpoint * endpoint
#define ast_assert(a)
Definition: utils.h:739

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_endpoint_media_configuration::rtp, ast_sip_session_media::rtp, and ast_sip_channel_pvt::session.

◆ chan_pjsip_get_uniqueid()

static const char * chan_pjsip_get_uniqueid ( struct ast_channel ast)
static

Definition at line 1283 of file chan_pjsip.c.

1284{
1285 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1287
1288 if (!channel || !uniqueid) {
1289 return "";
1290 }
1291
1292 ast_copy_pj_str(uniqueid, &channel->session->inv_session->dlg->call_id->id, UNIQUEID_BUFSIZE);
1293
1294 return uniqueid;
1295}
#define UNIQUEID_BUFSIZE
Definition: chan_pjsip.c:76
static struct ast_threadstorage uniqueid_threadbuf
Definition: chan_pjsip.c:75
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:2201
struct pjsip_inv_session * inv_session
void * ast_threadstorage_get(struct ast_threadstorage *ts, size_t init_size)
Retrieve thread storage.

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.

◆ 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 223 of file chan_pjsip.c.

224{
225 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
226 struct ast_sip_endpoint *endpoint;
227 struct ast_sip_session_media *media;
228
229 if (!channel || !channel->session) {
231 }
232
234 if (!media || !media->rtp) {
236 }
237
238 endpoint = channel->session->endpoint;
239
240 *instance = media->rtp;
241 ao2_ref(*instance, +1);
242
243 ast_assert(endpoint != NULL);
246 }
247
249}
@ AST_MEDIA_TYPE_VIDEO
Definition: codec.h:33

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_endpoint_media_configuration::rtp, ast_sip_session_media::rtp, and ast_sip_channel_pvt::session.

◆ 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 2562 of file chan_pjsip.c.

2563{
2564 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2565 int cause;
2566 struct hangup_data *h_data;
2567 SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
2568
2569 if (!channel || !channel->session) {
2570 SCOPE_EXIT_RTN_VALUE(-1, "No channel or session\n");
2571 }
2572
2574 h_data = hangup_data_alloc(cause, ast);
2575
2576 if (!h_data) {
2577 goto failure;
2578 }
2579
2580 if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
2581 ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
2582 goto failure;
2583 }
2584
2585 SCOPE_EXIT_RTN_VALUE(0, "Cause: %d\n", cause);
2586
2587failure:
2588 /* Go ahead and do our cleanup of the session and channel even if we're not going
2589 * to be able to send our SIP request/response
2590 */
2591 clear_session_and_channel(channel->session, ast);
2592 ao2_cleanup(channel);
2593 ao2_cleanup(h_data);
2594
2595 SCOPE_EXIT_RTN_VALUE(-1, "Cause: %d\n", cause);
2596}
static int hangup_cause2sip(int cause)
Internal function which translates from Asterisk cause codes to SIP response codes.
Definition: chan_pjsip.c:2439
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:2513
static int hangup(void *data)
Definition: chan_pjsip.c:2520
static struct hangup_data * hangup_data_alloc(int cause, struct ast_channel *chan)
Definition: chan_pjsip.c:2498
int ast_channel_hangupcause(const struct ast_channel *chan)

References ao2_cleanup, ast_channel_hangupcause(), ast_channel_name(), ast_channel_tech_pvt(), ast_log, ast_sip_push_task(), hangup_data::cause, 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.

◆ chan_pjsip_incoming_ack()

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

Definition at line 3208 of file chan_pjsip.c.

3209{
3211
3212 if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
3213 if (session->endpoint->media.direct_media.enabled && session->channel) {
3214 ast_trace(-1, "%s: Queueing SRCCHANGE\n", ast_sip_session_get_name(session));
3216 }
3217 }
3219}
int ast_queue_control(struct ast_channel *chan, enum ast_control_frame_type control)
Queue a control frame without payload.
Definition: channel.c:1250
#define ast_trace(level,...)
@ AST_CONTROL_SRCCHANGE

References AST_CONTROL_SRCCHANGE, ast_queue_control(), ast_sip_session_get_name(), ast_trace, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, and session.

◆ chan_pjsip_incoming_prack()

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

Definition at line 3221 of file chan_pjsip.c.

3222{
3224
3225 if (pj_strcmp2(&rdata->msg_info.msg->line.req.method.name, "PRACK") == 0 &&
3226 pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE) {
3227
3228 session->early_confirmed = 1;
3229 }
3231}

References ast_sip_session_get_name(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, and session.

◆ 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 2972 of file chan_pjsip.c.

2973{
2974 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
2975 struct transport_info_data *transport_data;
2976 pjsip_tx_data *packet = NULL;
2978
2979 if (session->channel) {
2980 SCOPE_EXIT_RTN_VALUE(0, "%s: No channel\n", ast_sip_session_get_name(session));
2981 }
2982
2983 /* Check for a to-tag to determine if this is a reinvite */
2984 if (rdata->msg_info.to->tag.slen) {
2985 /* Weird case. We've received a reinvite but we don't have a channel. The most
2986 * typical case for this happening is that a blind transfer fails, and so the
2987 * transferer attempts to reinvite himself back into the call. We already got
2988 * rid of that channel, and the other side of the call is unrecoverable.
2989 *
2990 * We treat this as a failure, so our best bet is to just hang this call
2991 * up and not create a new channel. Clearing defer_terminate here ensures that
2992 * calling ast_sip_session_terminate() can result in a BYE being sent ASAP.
2993 */
2994 session->defer_terminate = 0;
2996 SCOPE_EXIT_RTN_VALUE(-1, "%s: We have a To tag but no channel. Terminating session\n", ast_sip_session_get_name(session));
2997 }
2998
2999 datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
3000 if (!datastore) {
3001 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info datastore\n", ast_sip_session_get_name(session));
3002 }
3003
3004 transport_data = ast_calloc(1, sizeof(*transport_data));
3005 if (!transport_data) {
3006 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info\n", ast_sip_session_get_name(session));
3007 }
3008 pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
3009 pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
3010 datastore->data = transport_data;
3012
3013 if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL, NULL))) {
3014 if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS
3015 && packet) {
3017 }
3018
3019 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n",
3021 }
3022
3024
3025 /* channel gets created on incoming request, but we wait to call start
3026 so other supplements have a chance to run */
3028}
#define ast_calloc(num, len)
A wrapper for calloc()
Definition: astmm.h:202
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:269
static void set_sipdomain_variable(struct ast_sip_session *session)
Definition: chan_pjsip.c:2959
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 SCOPE_EXIT_LOG_RTN_VALUE(__value, __log_level,...)
int ast_sip_session_add_datastore(struct ast_sip_session *session, struct ast_datastore *datastore)
Add a datastore to a SIP session.
struct ast_datastore * ast_sip_session_alloc_datastore(const struct ast_datastore_info *info, const char *uid)
Alternative for ast_datastore_alloc()
void ast_sip_session_terminate(struct ast_sip_session *session, int response)
Terminate a session and, if possible, send the provided response code.
Transport information stored in transport_info datastore.
Definition: chan_pjsip.h:30
pj_sockaddr local_addr
Our address that received the request.
Definition: chan_pjsip.h:34
pj_sockaddr remote_addr
The address that sent the request.
Definition: chan_pjsip.h:32

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(), 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, session, set_sipdomain_variable(), and transport_info.

◆ 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 3151 of file chan_pjsip.c.

3152{
3153 struct pjsip_status_line status = rdata->msg_info.msg->line.status;
3154 SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
3155
3156 if (!session->channel) {
3157 SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
3158 }
3159
3160 switch (status.code) {
3161 case 180: {
3162 pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
3163 if (sdp && sdp->body.ptr) {
3164 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
3165 session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
3167 } else {
3168 ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
3170 }
3171
3172 ast_channel_lock(session->channel);
3173 if (ast_channel_state(session->channel) != AST_STATE_UP) {
3175 }
3176 ast_channel_unlock(session->channel);
3177 break;
3178 }
3179 case 183:
3180 if (session->endpoint->ignore_183_without_sdp) {
3181 pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
3182 if (sdp && sdp->body.ptr) {
3183 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
3184 ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS with SDP\n", ast_sip_session_get_name(session),
3185 (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
3186 session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
3188 }
3189 } else {
3190 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
3191 ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS without SDP\n", ast_sip_session_get_name(session),
3192 (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
3194 }
3195 break;
3196 case 200:
3197 ast_trace(-1, "%s: Queueing ANSWER\n", ast_sip_session_get_name(session));
3199 break;
3200 default:
3201 ast_trace(-1, "%s: Not queueing anything\n", ast_sip_session_get_name(session));
3202 break;
3203 }
3204
3206}
@ AST_STATE_RINGING
Definition: channelstate.h:41
@ AST_CONTROL_PROGRESS
@ AST_CONTROL_ANSWER
@ AST_CONTROL_RINGING

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, SCOPE_ENTER, SCOPE_EXIT_RTN, session, and status.

◆ 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 3121 of file chan_pjsip.c.

3122{
3123 struct pjsip_status_line status = rdata->msg_info.msg->line.status;
3124 struct ast_control_pvt_cause_code *cause_code;
3125 int data_size = sizeof(*cause_code);
3126 SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
3127
3128 if (!session->channel) {
3129 SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
3130 }
3131
3132 /* Build and send the tech-specific cause information */
3133 /* size of the string making up the cause code is "SIP " number + " " + reason length */
3134 data_size += 4 + 4 + pj_strlen(&status.reason);
3135 cause_code = ast_alloca(data_size);
3136 memset(cause_code, 0, data_size);
3137
3139
3140 snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %d %.*s", status.code,
3141 (int) pj_strlen(&status.reason), pj_strbuf(&status.reason));
3142
3143 cause_code->ast_cause = ast_sip_hangup_sip2cause(status.code);
3144 ast_queue_control_data(session->channel, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
3145 ast_channel_hangupcause_hash_set(session->channel, cause_code, data_size);
3146
3148}
#define ast_alloca(size)
call __builtin_alloca to ensure we get gcc builtin semantics
Definition: astmm.h:288
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:1257
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:4365
#define AST_CHANNEL_NAME
Definition: channel.h:173
@ AST_CONTROL_PVT_CAUSE_CODE
const int ast_sip_hangup_sip2cause(int cause)
Convert SIP hangup causes to Asterisk hangup causes.
Definition: res_pjsip.c:3531
char chan_name[AST_CHANNEL_NAME]

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_hangup_sip2cause(), ast_sip_session_get_name(), ast_control_pvt_cause_code::chan_name, ast_control_pvt_cause_code::code, SCOPE_ENTER, SCOPE_EXIT_RTN, session, and status.

◆ 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 1625 of file chan_pjsip.c.

1626{
1627 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1628 struct ast_sip_session_media *media;
1629 int response_code = 0;
1630 int res = 0;
1631 char *device_buf;
1632 size_t device_buf_size;
1633 int i;
1634 const struct ast_stream_topology *topology;
1635 struct ast_frame f = {
1637 .subclass = {
1638 .integer = condition
1639 },
1640 .datalen = datalen,
1641 .data.ptr = (void *)data,
1642 };
1643 char condition_name[256];
1644 unsigned int duration;
1645 char digit;
1646 struct info_dtmf_data *dtmf_data;
1647
1648 SCOPE_ENTER(3, "%s: Indicated %s\n", ast_channel_name(ast),
1649 ast_frame_subclass2str(&f, condition_name, sizeof(condition_name), NULL, 0));
1650
1651 switch (condition) {
1653 if (ast_channel_state(ast) == AST_STATE_RING) {
1654 if (channel->session->endpoint->inband_progress ||
1655 (channel->session->inv_session && channel->session->inv_session->neg &&
1656 pjmedia_sdp_neg_get_state(channel->session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE)) {
1657 res = -1;
1659 response_code = 180;
1660 } else {
1661 response_code = 183;
1662 }
1663 } else {
1664 response_code = 180;
1665 }
1666 } else {
1667 res = -1;
1668 }
1670 break;
1671 case AST_CONTROL_BUSY:
1672 if (ast_channel_state(ast) != AST_STATE_UP) {
1673 response_code = 486;
1674 } else {
1675 res = -1;
1676 }
1677 break;
1679 if (ast_channel_state(ast) != AST_STATE_UP) {
1680 response_code = 503;
1681 } else {
1682 res = -1;
1683 }
1684 break;
1686 if (ast_channel_state(ast) != AST_STATE_UP) {
1687 response_code = 484;
1688 } else {
1689 res = -1;
1690 }
1691 break;
1693 if (ast_channel_state(ast) != AST_STATE_UP) {
1694 response_code = 100;
1695 } else {
1696 res = -1;
1697 }
1698 break;
1700 if (ast_channel_state(ast) != AST_STATE_UP) {
1701 response_code = 183;
1702 } else {
1703 res = -1;
1704 }
1706 break;
1707 case AST_CONTROL_FLASH:
1708 duration = 300;
1709 digit = '!';
1710 dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
1711
1712 if (!dtmf_data) {
1713 res = -1;
1714 break;
1715 }
1716
1717 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
1718 ast_log(LOG_WARNING, "Error sending FLASH via INFO on channel %s\n", ast_channel_name(ast));
1719 ao2_ref(dtmf_data, -1); /* dtmf_data can't be null here */
1720 res = -1;
1721 }
1722 break;
1724 for (i = 0; i < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions); ++i) {
1725 media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, i);
1726 if (!media || media->type != AST_MEDIA_TYPE_VIDEO) {
1727 continue;
1728 }
1729 if (media->rtp) {
1730 /* FIXME: Only use this for VP8. Additional work would have to be done to
1731 * fully support other video codecs */
1732
1736 (channel->session->endpoint->media.webrtc &&
1738 /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
1739 * RTP engine would provide a way to externally write/schedule RTCP
1740 * packets */
1741 struct ast_frame fr;
1743 fr.subclass.integer = AST_CONTROL_VIDUPDATE;
1744 res = ast_rtp_instance_write(media->rtp, &fr);
1745 } else {
1746 ao2_ref(channel->session, +1);
1748 ao2_cleanup(channel->session);
1749 }
1750 }
1751 ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Success");
1752 } else {
1753 ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Failure");
1754 res = -1;
1755 }
1756 }
1757 /* XXX If there were no video streams, then this should set
1758 * res to -1
1759 */
1760 break;
1762 ao2_ref(channel->session, +1);
1764 ao2_cleanup(channel->session);
1765 }
1766 break;
1768 break;
1770 res = -1;
1771 break;
1773 ast_assert(datalen == sizeof(int));
1774 if (*(int *) data) {
1775 /*
1776 * Masquerade is beginning:
1777 * Wait for session serializer to get suspended.
1778 */
1779 ast_channel_unlock(ast);
1781 ast_channel_lock(ast);
1782 } else {
1783 /*
1784 * Masquerade is complete:
1785 * Unsuspend the session serializer.
1786 */
1788 }
1789 break;
1790 case AST_CONTROL_HOLD:
1792 device_buf_size = strlen(ast_channel_name(ast)) + 1;
1793 device_buf = alloca(device_buf_size);
1794 ast_channel_get_device_name(ast, device_buf, device_buf_size);
1796 if (!channel->session->moh_passthrough) {
1797 ast_moh_start(ast, data, NULL);
1798 } else {
1800 ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n",
1802 ao2_ref(channel->session, -1);
1803 }
1804 }
1805 break;
1806 case AST_CONTROL_UNHOLD:
1808 device_buf_size = strlen(ast_channel_name(ast)) + 1;
1809 device_buf = alloca(device_buf_size);
1810 ast_channel_get_device_name(ast, device_buf, device_buf_size);
1812 if (!channel->session->moh_passthrough) {
1813 ast_moh_stop(ast);
1814 } else {
1816 ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n",
1818 ao2_ref(channel->session, -1);
1819 }
1820 }
1821 break;
1823 break;
1825 break;
1827 if (ast_channel_state(ast) != AST_STATE_UP) {
1828 response_code = 181;
1829 } else {
1830 res = -1;
1831 }
1832 break;
1834 res = 0;
1835
1836 if (channel->session->t38state == T38_PEER_REINVITE) {
1837 const struct ast_control_t38_parameters *parameters = data;
1838
1839 if (parameters->request_response == AST_T38_REQUEST_PARMS) {
1841 }
1842 }
1843
1844 break;
1846 topology = data;
1847 ast_trace(-1, "%s: New topology: %s\n", ast_channel_name(ast),
1848 ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
1849 res = handle_topology_request_change(channel->session, topology);
1850 break;
1852 break;
1854 break;
1855 case -1:
1856 res = -1;
1857 break;
1858 default:
1859 ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
1860 res = -1;
1861 break;
1862 }
1863
1864 if (response_code) {
1865 struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
1866
1867 if (!ind_data) {
1868 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc indicate data\n", ast_channel_name(ast));
1869 }
1870
1871 if (ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1872 ast_log(LOG_ERROR, "%s: Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1874 ao2_cleanup(ind_data);
1875 res = -1;
1876 }
1877 }
1878
1879 SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_channel_name(ast));
1880}
static int update_connected_line_information(void *data)
Update connected line information.
Definition: chan_pjsip.c:1434
static int remote_send_unhold(void *data)
Update local hold state to be unheld.
Definition: chan_pjsip.c:1506
static int handle_topology_request_change(struct ast_sip_session *session, const struct ast_stream_topology *proposed)
Definition: chan_pjsip.c:1601
static int indicate(void *data)
Definition: chan_pjsip.c:1338
static int remote_send_hold(void *data)
Update local hold state to be held.
Definition: chan_pjsip.c:1500
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:1313
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:1153
static int transmit_info_with_vidupdate(void *data)
Send SIP INFO with video update request.
Definition: chan_pjsip.c:1356
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:1122
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:10518
@ AST_DEVSTATE_CACHABLE
Definition: devicestate.h:70
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
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
@ AST_FORMAT_CMP_NOT_EQUAL
Definition: format.h:38
struct ast_format * ast_format_h264
Built-in cached h264 format.
Definition: format_cache.c:176
struct ast_format * ast_format_h265
Built-in cached h265 format.
Definition: format_cache.c:181
struct ast_format * ast_format_vp9
Built-in cached vp9 format.
Definition: format_cache.c:196
struct ast_format * ast_format_vp8
Built-in cached vp8 format.
Definition: format_cache.c:191
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:581
@ AST_T38_REQUEST_PARMS
char * ast_frame_subclass2str(struct ast_frame *f, char *subclass, size_t slen, char *moreinfo, size_t mlen)
Copy the discription of a frame's subclass into the provided string.
Definition: main/frame.c:406
@ AST_FRAME_CONTROL
@ AST_CONTROL_SRCUPDATE
@ AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED
@ AST_CONTROL_BUSY
@ AST_CONTROL_UNHOLD
@ AST_CONTROL_VIDUPDATE
@ AST_CONTROL_STREAM_TOPOLOGY_REQUEST_CHANGE
@ AST_CONTROL_PROCEEDING
@ AST_CONTROL_REDIRECTING
@ AST_CONTROL_T38_PARAMETERS
@ AST_CONTROL_CONGESTION
@ AST_CONTROL_HOLD
@ AST_CONTROL_STREAM_TOPOLOGY_CHANGED
@ AST_CONTROL_CONNECTED_LINE
@ AST_CONTROL_FLASH
@ AST_CONTROL_INCOMPLETE
@ AST_CONTROL_MASQUERADE_NOTIFY
@ AST_CONTROL_UPDATE_RTP_PEER
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:7788
void ast_moh_stop(struct ast_channel *chan)
Turn off music on hold on a given channel.
Definition: channel.c:7798
unsigned int ast_sip_get_allow_sending_180_after_183(void)
Retrieve the global setting 'allow_sending_180_after_183'.
@ T38_PEER_REINVITE
void ast_sip_session_unsuspend(struct ast_sip_session *session)
Request the session serializer be unsuspended.
void ast_sip_session_suspend(struct ast_sip_session *session)
Request and wait for the session serializer to be suspended.
int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
Send a frame out over RTP.
Definition: rtp_engine.c:590
const char * ast_sorcery_object_get_id(const void *object)
Get the unique identifier of a sorcery object.
Definition: sorcery.c:2317
enum ast_control_t38 request_response
Data structure associated with a single frame of data.
union ast_frame::@226 data
enum ast_frame_type frametype
unsigned int inband_progress
Definition: res_pjsip.h:1018
struct ast_sip_session_media_state::@263 sessions
Mapping of stream to media sessions.
enum ast_media_type type
Media type of this session media.
enum ast_sip_session_t38state t38state
unsigned int moh_passthrough
size_t datalen
Definition: chan_pjsip.c:1302
#define ast_test_suite_event_notify(s, f,...)
Definition: test.h:189
#define AST_VECTOR_SIZE(vec)
Get the number of elements in a vector.
Definition: vector.h:609
#define AST_VECTOR_GET(vec, idx)
Get an element from a vector.
Definition: vector.h:680

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_FLASH, 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_get_allow_sending_180_after_183(), 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(), indicate_data::condition, ast_frame::data, indicate_data::datalen, ast_frame::datalen, digit, info_dtmf_data::duration, ast_sip_session::endpoint, ast_frame::frametype, handle_topology_request_change(), ast_sip_endpoint::inband_progress, indicate(), indicate_data_alloc(), info_dtmf_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, indicate_data::response_code, 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_sip_session_media_state::sessions, ast_frame::subclass, T38_PEER_REINVITE, ast_sip_session::t38state, transmit_info_dtmf(), transmit_info_with_vidupdate(), ast_sip_session_media::type, update_connected_line_information(), and ast_sip_endpoint_media_configuration::webrtc.

◆ 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.

548{
549 struct ast_channel *chan;
550 struct ast_format_cap *caps;
551 RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
552 struct ast_sip_channel_pvt *channel;
553 struct ast_variable *var;
554 struct ast_stream_topology *topology;
555 struct ast_channel_initializers initializers = {
557 .tenantid = session->endpoint->tenantid,
558 };
560
562 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt\n");
563 }
564
566 S_COR(session->id.number.valid, session->id.number.str, ""),
567 S_COR(session->id.name.valid, session->id.name.str, ""),
568 session->endpoint->accountcode,
569 exten, session->endpoint->context,
570 assignedids, requestor, 0,
571 session->endpoint->persistent, &initializers, "PJSIP/%s-%08x",
573 (unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
574 if (!chan) {
575 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
576 }
577
579
580 if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
581 ast_channel_unlock(chan);
582 ast_hangup(chan);
583 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt channel\n");
584 }
585
586 ast_channel_tech_pvt_set(chan, channel);
587
588 if (!ast_stream_topology_get_count(session->pending_media_state->topology) ||
589 !compatible_formats_exist(session->pending_media_state->topology, session->endpoint->media.codecs)) {
591 if (!caps) {
592 ast_channel_unlock(chan);
593 ast_hangup(chan);
594 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create caps\n");
595 }
596 ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
597 topology = ast_stream_topology_clone(session->endpoint->media.topology);
598 } else {
599 caps = ast_stream_topology_get_formats(session->pending_media_state->topology);
600 topology = ast_stream_topology_clone(session->pending_media_state->topology);
601 }
602
603 if (!topology || !caps) {
604 ao2_cleanup(caps);
605 ast_stream_topology_free(topology);
606 ast_channel_unlock(chan);
607 ast_hangup(chan);
608 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't get caps or clone topology\n");
609 }
610
612
614 ast_channel_set_stream_topology(chan, topology);
615
616 if (!ast_format_cap_empty(caps)) {
617 struct ast_format *fmt;
618
620 if (!fmt) {
621 /* Since our capabilities aren't empty, this will succeed */
622 fmt = ast_format_cap_get_format(caps, 0);
623 }
628 ao2_ref(fmt, -1);
629 }
630
631 ao2_ref(caps, -1);
632
633 if (state == AST_STATE_RING) {
634 ast_channel_rings_set(chan, 1);
635 }
636
638
641 ast_channel_caller(chan)->ani2 = session->ani2;
642
643 if (!ast_strlen_zero(exten)) {
644 /* Set provided DNID on the new channel. */
645 ast_channel_dialed(chan)->number.str = ast_strdup(exten);
646 }
647
649
650 ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
651 ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup);
652
653 ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups);
654 ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups);
655
656 if (!ast_strlen_zero(session->endpoint->language)) {
657 ast_channel_language_set(chan, session->endpoint->language);
658 }
659
660 if (!ast_strlen_zero(session->endpoint->zone)) {
661 struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
662 if (!zone) {
663 ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
664 }
665 ast_channel_zone_set(chan, zone);
666 }
667
668 for (var = session->endpoint->channel_vars; var; var = var->next) {
669 char buf[512];
671 var->value, buf, sizeof(buf)));
672 }
673
675 ast_channel_unlock(chan);
676
678
680}
#define var
Definition: ast_expr2f.c:605
#define ast_strdup(str)
A wrapper for strdup()
Definition: astmm.h:241
static void chan_pjsip_pvt_dtor(void *obj)
Definition: chan_pjsip.c:82
struct ast_channel_tech chan_pjsip_tech
PBX interface structure for channel registration.
Definition: chan_pjsip.c:109
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
static unsigned int chan_idx
Definition: chan_pjsip.c:80
void ast_channel_rings_set(struct ast_channel *chan, int value)
void ast_channel_named_pickupgroups_set(struct ast_channel *chan, struct ast_namedgroups *value)
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.
void ast_channel_nativeformats_set(struct ast_channel *chan, struct ast_format_cap *value)
@ AST_ADSI_UNAVAILABLE
Definition: channel.h:891
#define AST_CHANNEL_INITIALIZERS_VERSION
struct ABI version
Definition: channel.h:620
void ast_channel_set_rawreadformat(struct ast_channel *chan, struct ast_format *format)
void ast_channel_tech_pvt_set(struct ast_channel *chan, void *value)
void ast_channel_callgroup_set(struct ast_channel *chan, ast_group_t value)
void ast_channel_set_rawwriteformat(struct ast_channel *chan, struct ast_format *format)
struct ast_party_dialed * ast_channel_dialed(struct ast_channel *chan)
#define ast_channel_alloc_with_initializers(needqueue, state, cid_num, cid_name, acctcode, exten, context, assignedids, requestor, amaflag, endpoint, initializers,...)
Definition: channel.h:1307
void ast_channel_named_callgroups_set(struct ast_channel *chan, struct ast_namedgroups *value)
void ast_channel_set_readformat(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:1784
void ast_channel_zone_set(struct ast_channel *chan, struct ast_tone_zone *value)
void ast_channel_priority_set(struct ast_channel *chan, int value)
void ast_channel_pickupgroup_set(struct ast_channel *chan, ast_group_t value)
void ast_channel_adsicpe_set(struct ast_channel *chan, enum ast_channel_adsicpe value)
void ast_channel_tech_set(struct ast_channel *chan, const struct ast_channel_tech *value)
void ast_channel_set_writeformat(struct ast_channel *chan, struct ast_format *format)
char buf[BUFSIZE]
Definition: eagi_proxy.c:66
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:744
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
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
@ AST_FORMAT_CAP_FLAG_DEFAULT
Definition: format_cap.h:38
#define ast_format_cap_alloc(flags)
Allocate a new ast_format_cap structure.
Definition: format_cap.h:49
void ast_channel_stage_snapshot_done(struct ast_channel *chan)
Clear flag to indicate channel snapshot is being staged, and publish snapshot.
void ast_channel_stage_snapshot(struct ast_channel *chan)
Set flag to indicate channel snapshot is being staged.
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:3162
struct ast_tone_zone * ast_get_indication_zone(const char *country)
locate ast_tone_zone
Definition: indications.c:439
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:757
struct ast_sip_channel_pvt * ast_sip_channel_pvt_alloc(void *pvt, struct ast_sip_session *session)
Allocate a new SIP channel pvt structure.
int ast_stream_topology_get_count(const struct ast_stream_topology *topology)
Get the number of streams in a topology.
Definition: stream.c:765
void ast_stream_topology_free(struct ast_stream_topology *topology)
Unreference and destroy a stream topology.
Definition: stream.c:743
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_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
static force_inline int attribute_pure ast_strlen_zero(const char *s)
Definition: strings.h:65
Helper struct for initializing additional channel information on channel creation.
Definition: channel.h:615
uint32_t version
struct ABI version
Definition: channel.h:625
Format capabilities structure, holds formats + preference order + etc.
Definition: format_cap.c:54
Definition of a media format.
Definition: format.c:43
int ani2
Automatic Number Identification 2 (Info Digits)
Definition: channel.h:435
struct ast_party_dialed::@208 number
Dialed/Called number.
char * str
Subscriber phone number (Malloced)
Definition: channel.h:388
A set of tones for a given locale.
Definition: indications.h:74
Structure for variables, used for configurations and for channel variables.
The PJSIP channel driver pvt, stored in the ast_sip_channel_pvt data structure.
Definition: chan_pjsip.h:42

References ast_party_caller::ani2, 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_initializers, ast_channel_caller(), ast_channel_callgroup_set(), ast_channel_dialed(), AST_CHANNEL_INITIALIZERS_VERSION, 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, chan_idx, chan_pjsip_pvt_dtor(), chan_pjsip_tech, compatible_formats_exist(), LOG_ERROR, NULL, ast_party_dialed::number, pbx_builtin_setvar_helper(), RAII_VAR, S_COR, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, session, set_channel_on_rtp_instance(), ast_party_dialed::str, var, and ast_channel_initializers::version.

Referenced by chan_pjsip_incoming_request(), and chan_pjsip_request_with_stream_topology().

◆ chan_pjsip_pvt_dtor()

static void chan_pjsip_pvt_dtor ( void *  obj)
static

Definition at line 82 of file chan_pjsip.c.

83{
84}

Referenced by chan_pjsip_new().

◆ 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 1242 of file chan_pjsip.c.

1243{
1244 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
1245 int res = -1;
1247
1248 if (!channel) {
1249 return -1;
1250 }
1251
1252 switch (option) {
1254 if (channel->session->endpoint->media.t38.enabled) {
1255 switch (channel->session->t38state) {
1256 case T38_LOCAL_REINVITE:
1257 case T38_PEER_REINVITE:
1259 break;
1260 case T38_ENABLED:
1262 break;
1263 case T38_REJECTED:
1265 break;
1266 default:
1268 break;
1269 }
1270 }
1271
1272 *((enum ast_t38_state *) data) = state;
1273 res = 0;
1274
1275 break;
1276 default:
1277 break;
1278 }
1279
1280 return res;
1281}
ast_t38_state
Possible T38 states on channels.
Definition: channel.h:898
@ T38_STATE_UNAVAILABLE
Definition: channel.h:899
@ T38_STATE_UNKNOWN
Definition: channel.h:900
@ T38_STATE_REJECTED
Definition: channel.h:902
@ T38_STATE_NEGOTIATED
Definition: channel.h:903
@ T38_STATE_NEGOTIATING
Definition: channel.h:901
#define AST_OPTION_T38_STATE
@ T38_LOCAL_REINVITE
@ T38_ENABLED
@ T38_REJECTED
struct ast_sip_t38_configuration t38
Definition: res_pjsip.h:917

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.

◆ 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 843 of file chan_pjsip.c.

844{
845 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
846 struct ast_sip_session *session = channel->session;
847 struct ast_sip_session_media_read_callback_state *callback_state;
848 struct ast_frame *f;
849 int fdno = ast_channel_fdno(ast) - AST_EXTENDED_FDS;
850 struct ast_frame *cur;
851
852 if (fdno >= AST_VECTOR_SIZE(&session->active_media_state->read_callbacks)) {
853 return &ast_null_frame;
854 }
855
856 callback_state = AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, fdno);
857 f = callback_state->read_callback(session, callback_state->session);
858
859 if (!f) {
860 return f;
861 }
862
863 for (cur = f; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
864 if (cur->frametype == AST_FRAME_VOICE) {
865 break;
866 }
867 }
868
869 if (!cur || callback_state->session != session->active_media_state->default_session[callback_state->session->type]) {
870 return f;
871 }
872
873 session = channel->session;
874
875 /*
876 * Asymmetric RTP only has one native format set at a time.
877 * Therefore we need to update the native format to the current
878 * raw read format BEFORE the native format check
879 */
880 if (!session->endpoint->asymmetric_rtp_codec &&
883 struct ast_format_cap *caps;
884
885 /* For maximum compatibility we ensure that the formats match that of the received media */
886 ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
889
891 if (caps) {
896 ao2_ref(caps, -1);
897 }
898
901
902 if (ast_channel_is_bridged(ast)) {
904 }
905 }
906
909 ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n",
911 ast_frfree(f);
912 return &ast_null_frame;
913 }
914
915 if (session->dsp) {
916 int dsp_features;
917
918 dsp_features = ast_dsp_get_features(session->dsp);
919 if ((dsp_features & DSP_FEATURE_FAX_DETECT)
920 && session->endpoint->faxdetect_timeout
921 && session->endpoint->faxdetect_timeout <= ast_channel_get_up_time(ast)) {
922 dsp_features &= ~DSP_FEATURE_FAX_DETECT;
923 if (dsp_features) {
924 ast_dsp_set_features(session->dsp, dsp_features);
925 } else {
926 ast_dsp_free(session->dsp);
927 session->dsp = NULL;
928 }
929 ast_debug(3, "Channel driver fax CNG detection timeout on %s\n",
930 ast_channel_name(ast));
931 }
932 }
933 if (session->dsp) {
934 f = ast_dsp_process(ast, session->dsp, f);
935 if (f && (f->frametype == AST_FRAME_DTMF)) {
936 if (f->subclass.integer == 'f') {
937 ast_debug(3, "Channel driver fax CNG detected on %s\n",
938 ast_channel_name(ast));
940 /* When chan_pjsip_cng_tone_detected returns it is possible for the
941 * channel pointed to by ast and by session->channel to differ due to a
942 * masquerade. It's best not to touch things after this.
943 */
944 } else {
945 ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
946 ast_channel_name(ast));
947 }
948 }
949 }
950
951 return f;
952}
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.
Definition: chan_pjsip.c:829
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:770
#define AST_EXTENDED_FDS
Definition: channel.h:197
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.
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:5507
int ast_channel_fdno(const struct ast_channel *chan)
struct ast_format * ast_channel_rawwriteformat(struct ast_channel *chan)
int ast_channel_is_bridged(const struct ast_channel *chan)
Determine if a channel is in a bridge.
Definition: channel.c:10567
struct ast_format * ast_channel_writeformat(struct ast_channel *chan)
int ast_channel_get_up_time(struct ast_channel *chan)
Obtain how long it has been since the channel was answered.
Definition: channel.c:2864
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:5543
struct ast_format * ast_channel_readformat(struct ast_channel *chan)
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:1499
#define DSP_FEATURE_FAX_DETECT
Definition: dsp.h:29
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
const char * ast_format_get_name(const struct ast_format *format)
Get the name associated with a format.
Definition: format.c:334
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:523
#define ast_format_cap_append(cap, format, framing)
Add format capability to capabilities structure.
Definition: format_cap.h:99
#define AST_FRAME_DTMF
@ AST_FRAME_VOICE
#define AST_LIST_NEXT(elm, field)
Returns the next entry in the list after the given entry.
Definition: linkedlists.h:439
struct ast_format * format
struct ast_frame_subclass subclass
Structure which contains read callback information.
ast_sip_session_media_read_cb read_callback
The callback to invoke.
struct ast_sip_session_media * session
The media session.
#define AST_VECTOR_GET_ADDR(vec, idx)
Get an address of element in a vector.
Definition: vector.h:668

References 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, chan_pjsip_cng_tone_detected(), ast_sip_session::channel, DSP_FEATURE_FAX_DETECT, 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, session, ast_frame::subclass, and ast_sip_session_media::type.

◆ 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 1153 of file chan_pjsip.c.

1154{
1156}
@ OBJ_NODATA
Definition: astobj2.h:1044
@ OBJ_UNLINK
Definition: astobj2.h:1039

References ao2_find, OBJ_NODATA, OBJ_SEARCH_KEY, OBJ_UNLINK, and pjsip_uids_onhold.

Referenced by chan_pjsip_indicate(), and chan_pjsip_session_end().

◆ 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

Asterisk core interaction functions.

Definition at line 2740 of file chan_pjsip.c.

2741{
2742 struct ast_stream_topology *topology;
2743 struct ast_channel *chan;
2744
2746 if (!topology) {
2747 return NULL;
2748 }
2749
2750 chan = chan_pjsip_request_with_stream_topology(type, topology, assignedids, requestor, data, cause);
2751
2752 ast_stream_topology_free(topology);
2753
2754 return chan;
2755}
static const char type[]
Definition: chan_ooh323.c:109
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:2713
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
const char * data

References ast_stream_topology_create_from_format_cap(), ast_stream_topology_free(), chan_pjsip_request_with_stream_topology(), ast_channel::data, NULL, and type.

◆ 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 2713 of file chan_pjsip.c.

2714{
2715 struct request_data req_data;
2717 SCOPE_ENTER(1, "%s Topology: %s\n", data,
2719
2720 req_data.topology = topology;
2721 req_data.dest = data;
2722 /* Default failure value in case ast_sip_push_task_wait_servant() itself fails. */
2723 req_data.cause = AST_CAUSE_FAILURE;
2724
2725 if (ast_sip_push_task_wait_servant(NULL, request, &req_data)) {
2726 *cause = req_data.cause;
2727 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't push task\n");
2728 }
2729
2730 session = req_data.session;
2731
2732 if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) {
2733 /* Session needs to be terminated prematurely */
2734 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
2735 }
2736
2737 SCOPE_EXIT_RTN_VALUE(session->channel, "Channel: %s\n", ast_channel_name(session->channel));
2738}
#define AST_CAUSE_FAILURE
Definition: causes.h:150
static int request(void *obj)
Definition: chan_pjsip.c:2605
@ AST_STATE_DOWN
Definition: channelstate.h:36
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:2165
struct ast_stream_topology * topology
Definition: chan_pjsip.c:2600

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(), request_data::dest, NULL, RAII_VAR, request(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, request_data::session, session, and request_data::topology.

Referenced by chan_pjsip_request().

◆ chan_pjsip_sendtext()

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

Definition at line 2883 of file chan_pjsip.c.

2884{
2885 struct ast_msg_data *msg;
2886 int rc;
2887 struct ast_msg_data_attribute attrs[] =
2888 {
2889 {
2891 .value = (char *)text,
2892 }
2893 };
2894
2896 if (!msg) {
2897 return -1;
2898 }
2899 rc = chan_pjsip_sendtext_data(ast, msg);
2900 ast_free(msg);
2901
2902 return rc;
2903}
char * text
Definition: app_queue.c:1639
#define ast_free(a)
Definition: astmm.h:180
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:2861
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.
@ AST_MSG_DATA_ATTR_BODY
Definition: message.h:458
@ AST_MSG_DATA_SOURCE_TYPE_UNKNOWN
Definition: message.h:447
enum ast_msg_data_attribute_type type
Definition: message.h:463
Structure used to transport a message through the frame core.
#define ARRAY_LEN(a)
Definition: utils.h:666

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

◆ 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 2861 of file chan_pjsip.c.

2862{
2863 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2864 struct sendtext_data *data = sendtext_data_create(ast, msg);
2865
2866 ast_debug(1, "Sending MESSAGE from '%s' to '%s:%s': %s\n",
2869 ast_channel_name(ast),
2871
2872 if (!data) {
2873 return -1;
2874 }
2875
2876 if (ast_sip_push_task(channel->session->serializer, sendtext, data)) {
2877 ao2_ref(data, -1);
2878 return -1;
2879 }
2880 return 0;
2881}
static int sendtext(void *obj)
Definition: chan_pjsip.c:2790
static struct sendtext_data * sendtext_data_create(struct ast_channel *chan, struct ast_msg_data *msg)
Definition: chan_pjsip.c:2769
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.
@ AST_MSG_DATA_ATTR_TO
Definition: message.h:455
@ AST_MSG_DATA_ATTR_FROM
Definition: message.h:456
struct ast_msg_data * msg
Definition: chan_pjsip.c:2759

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_data::msg, sendtext(), sendtext_data_create(), ast_sip_session::serializer, and ast_sip_channel_pvt::session.

Referenced by chan_pjsip_sendtext().

◆ chan_pjsip_session_begin()

static void chan_pjsip_session_begin ( struct ast_sip_session session)
static

SIP session interaction functions.

Definition at line 2905 of file chan_pjsip.c.

2906{
2907 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
2909
2910 if (session->endpoint->media.direct_media.glare_mitigation ==
2912 SCOPE_EXIT_RTN("Direct media no glare mitigation\n");
2913 }
2914
2916 "direct_media_glare_mitigation");
2917
2918 if (!datastore) {
2919 SCOPE_EXIT_RTN("Couldn't create datastore\n");
2920 }
2921
2924}
static struct ast_datastore_info direct_media_mitigation_info
Definition: chan_pjsip.c:274
@ AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE
Definition: res_pjsip.h:632

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(), direct_media_mitigation_info, NULL, RAII_VAR, SCOPE_ENTER, SCOPE_EXIT_RTN, and 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 2927 of file chan_pjsip.c.

2928{
2930
2931 if (!session->channel) {
2932 SCOPE_EXIT_RTN("No channel\n");
2933 }
2934
2935
2936 if (session->active_media_state &&
2937 session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
2938 struct ast_sip_session_media *media =
2939 session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
2940 if (media->rtp) {
2942 }
2943 }
2944
2946
2947 ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
2948 if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
2949 int cause = ast_sip_hangup_sip2cause(session->inv_session->cause);
2950
2951 ast_queue_hangup_with_cause(session->channel, cause);
2952 } else {
2953 ast_queue_hangup(session->channel);
2954 }
2955
2957}
int ast_queue_hangup_with_cause(struct ast_channel *chan, int cause)
Queue a hangup frame with hangupcause set.
Definition: channel.c:1185
void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
Set standard statistics from an RTP instance on a channel.
Definition: rtp_engine.c:2691

References ast_channel_hangupcause(), ast_channel_name(), ast_channel_uniqueid(), AST_MEDIA_TYPE_AUDIO, ast_queue_hangup(), ast_queue_hangup_with_cause(), ast_rtp_instance_set_stats_vars(), ast_set_hangupsource(), ast_sip_hangup_sip2cause(), ast_sip_session_get_name(), chan_pjsip_remove_hold(), ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN, and session.

◆ 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.

454{
455 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
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
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}
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
static int send_direct_media_request(void *data)
Definition: chan_pjsip.c:396
static int cdata(void *userdata, int state, const char *cdata, size_t len)
struct ast_rtp_instance * vrtp
Definition: chan_pjsip.c:361
struct ast_channel * chan
Definition: chan_pjsip.c:359
struct ast_rtp_instance * rtp
Definition: chan_pjsip.c:360
struct ast_format_cap * cap
Definition: chan_pjsip.c:362

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, rtp_direct_media_data::cap, cdata(), rtp_direct_media_data::chan, ast_sip_session::channel, LOG_ERROR, rtp_direct_media_data::rtp, rtp_direct_media_data_create(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, send_direct_media_request(), session, and rtp_direct_media_data::vrtp.

◆ 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 2168 of file chan_pjsip.c.

2169{
2170 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
2171 struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
2172
2173 if (!trnf_data) {
2174 return -1;
2175 }
2176
2177 if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
2178 ast_log(LOG_WARNING, "Error requesting transfer\n");
2179 ao2_cleanup(trnf_data);
2180 return -1;
2181 }
2182
2183 return 0;
2184}
static int transfer(void *data)
Definition: chan_pjsip.c:2133
static struct transfer_data * transfer_data_alloc(struct ast_sip_session *session, const char *target)
Definition: chan_pjsip.c:1895

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_data::target, transfer(), and transfer_data_alloc().

◆ chan_pjsip_write()

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

Definition at line 1044 of file chan_pjsip.c.

1045{
1046 return chan_pjsip_write_stream(ast, -1, frame);
1047}
static int chan_pjsip_write_stream(struct ast_channel *ast, int stream_num, struct ast_frame *f)
Definition: chan_pjsip.c:954

References chan_pjsip_write_stream().

◆ 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 954 of file chan_pjsip.c.

955{
956 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
957 struct ast_sip_session *session = channel->session;
958 struct ast_sip_session_media *media = NULL;
959 int res = 0;
960
961 /* The core provides a guarantee that the stream will exist when we are called if stream_num is provided */
962 if (stream_num >= 0) {
963 /* What is not guaranteed is that a media session will exist */
966 }
967 }
968
969 switch (frame->frametype) {
970 case AST_FRAME_VOICE:
971 if (!media) {
972 return 0;
973 } else if (media->type != AST_MEDIA_TYPE_AUDIO) {
974 ast_debug(3, "Channel %s stream %d is of type '%s', not audio!\n",
976 return 0;
977 } else if (media == channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO] &&
980 struct ast_str *write_transpath = ast_str_alloca(256);
981 struct ast_str *read_transpath = ast_str_alloca(256);
982
984 "Channel %s asked to send %s frame when native formats are %s (rd:%s->%s;%s wr:%s->%s;%s)\n",
985 ast_channel_name(ast),
986 ast_format_get_name(frame->subclass.format),
993 ast_translate_path_to_str(ast_channel_writetrans(ast), &write_transpath));
994 return 0;
995 } else if (media->write_callback) {
996 res = media->write_callback(session, media, frame);
997
998 }
999 break;
1000 case AST_FRAME_VIDEO:
1001 if (!media) {
1002 return 0;
1003 } else if (media->type != AST_MEDIA_TYPE_VIDEO) {
1004 ast_debug(3, "Channel %s stream %d is of type '%s', not video!\n",
1005 ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
1006 return 0;
1007 } else if (media->write_callback) {
1008 res = media->write_callback(session, media, frame);
1009 }
1010 break;
1011 case AST_FRAME_MODEM:
1012 if (!media) {
1013 return 0;
1014 } else if (media->type != AST_MEDIA_TYPE_IMAGE) {
1015 ast_debug(3, "Channel %s stream %d is of type '%s', not image!\n",
1016 ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
1017 return 0;
1018 } else if (media->write_callback) {
1019 res = media->write_callback(session, media, frame);
1020 }
1021 break;
1022 case AST_FRAME_CNG:
1023 break;
1024 case AST_FRAME_RTCP:
1025 /* We only support writing out feedback */
1026 if (frame->subclass.integer != AST_RTP_RTCP_PSFB || !media) {
1027 return 0;
1028 } else if (media->type != AST_MEDIA_TYPE_VIDEO) {
1029 ast_debug(3, "Channel %s stream %d is of type '%s', not video! Unable to write RTCP feedback.\n",
1030 ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
1031 return 0;
1032 } else if (media->write_callback) {
1033 res = media->write_callback(session, media, frame);
1034 }
1035 break;
1036 default:
1037 ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP\n", frame->frametype);
1038 break;
1039 }
1040
1041 return res;
1042}
struct ast_format * ast_channel_rawreadformat(struct ast_channel *chan)
struct ast_trans_pvt * ast_channel_readtrans(const struct ast_channel *chan)
struct ast_trans_pvt * ast_channel_writetrans(const struct ast_channel *chan)
@ AST_MEDIA_TYPE_IMAGE
Definition: codec.h:34
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:348
@ AST_FRAME_VIDEO
@ AST_FRAME_RTCP
@ AST_FRAME_MODEM
#define AST_RTP_RTCP_PSFB
Definition: rtp_engine.h:329
#define ast_str_alloca(init_len)
Definition: strings.h:848
ast_sip_session_media_write_cb write_callback
The write callback when writing frames.
int stream_num
The stream number to place into any resulting frames.
Support for dynamic strings.
Definition: strings.h:623
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:930

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::channel, 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, session, ast_sip_session_media_state::sessions, ast_sip_session_media::stream_num, ast_frame::subclass, ast_sip_session_media::type, and ast_sip_session_media::write_callback.

Referenced by chan_pjsip_write().

◆ 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 327 of file chan_pjsip.c.

329{
330 int changed = 0, position = -1;
331
332 if (media->rtp) {
333 position = rtp_find_rtcp_fd_position(session, media->rtp);
334 }
335
336 if (rtp) {
338 if (media->rtp) {
339 if (position != -1) {
340 ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, -1);
341 }
343 }
344 } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
346 changed = 1;
347 if (media->rtp) {
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}
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:306
void ast_channel_set_fd(struct ast_channel *chan, int which, int fd)
Definition: channel.c:2445
static int ast_sockaddr_isnull(const struct ast_sockaddr *addr)
Checks if the ast_sockaddr is null. "null" in this sense essentially means uninitialized,...
Definition: netsock2.h:127
static void ast_sockaddr_setnull(struct ast_sockaddr *addr)
Sets address addr to null.
Definition: netsock2.h:138
#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:1286
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:727
@ AST_RTP_PROPERTY_RTCP
Definition: rtp_engine.h:126
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:2368
struct ast_sockaddr direct_media_addr
Direct media address.

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

Referenced by send_direct_media_request().

◆ 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 2513 of file chan_pjsip.c.

2514{
2515 session->channel = NULL;
2518}

References ast_channel_tech_pvt_set(), NULL, session, and set_channel_on_rtp_instance().

Referenced by chan_pjsip_hangup(), and hangup().

◆ 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.

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}
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:653

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().

◆ direct_media_mitigate_glare()

static int direct_media_mitigate_glare ( struct ast_sip_session session)
static

Definition at line 276 of file chan_pjsip.c.

277{
278 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
279
280 if (session->endpoint->media.direct_media.glare_mitigation ==
282 return 0;
283 }
284
285 datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
286 if (!datastore) {
287 return 0;
288 }
289
290 /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
291 ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
292
293 if ((session->endpoint->media.direct_media.glare_mitigation ==
295 session->inv_session->role == PJSIP_ROLE_UAC) ||
296 (session->endpoint->media.direct_media.glare_mitigation ==
298 session->inv_session->role == PJSIP_ROLE_UAS)) {
299 return 1;
300 }
301
302 return 0;
303}
@ AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING
Definition: res_pjsip.h:640
@ AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING
Definition: res_pjsip.h:636
void ast_sip_session_remove_datastore(struct ast_sip_session *session, const char *name)
Remove a session datastore from the session.

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(), NULL, RAII_VAR, and session.

Referenced by send_direct_media_request().

◆ 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 1601 of file chan_pjsip.c.

1603{
1605 int res;
1606 SCOPE_ENTER(1);
1607
1609 if (!refresh_data) {
1610 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create refresh_data\n");
1611 }
1612
1614 if (res) {
1616 }
1617 SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
1618}
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
static void topology_change_refresh_data_free(struct topology_change_refresh_data *refresh_data)
Definition: chan_pjsip.c:1516

References ast_sip_push_task(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, send_topology_change_refresh(), session, topology_change_refresh_data_alloc(), and topology_change_refresh_data_free().

Referenced by chan_pjsip_indicate().

◆ hangup()

static int hangup ( void *  data)
static

Definition at line 2520 of file chan_pjsip.c.

2521{
2522 struct hangup_data *h_data = data;
2523 struct ast_channel *ast = h_data->chan;
2524 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2525 SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
2526
2527 /*
2528 * Before cleaning we have to ensure that channel or its session is not NULL
2529 * we have seen rare case when taskprocessor calls hangup but channel is NULL
2530 * due to SIP session timeout and answer happening at the same time
2531 */
2532 if (channel) {
2533 struct ast_sip_session *session = channel->session;
2534 if (session) {
2535 int cause = h_data->cause;
2536
2537 if (channel->session->active_media_state &&
2538 channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
2539 struct ast_sip_session_media *media =
2541 if (media->rtp) {
2543 }
2544 }
2545
2546 /*
2547 * It's possible that session_terminate might cause the session to be destroyed
2548 * immediately so we need to keep a reference to it so we can NULL session->channel
2549 * afterwards.
2550 */
2554 }
2555 ao2_cleanup(channel);
2556 }
2557 ao2_cleanup(h_data);
2559}
struct ast_channel * chan
Definition: chan_pjsip.c:2488

References ast_sip_session::active_media_state, ao2_bump, ao2_cleanup, ast_channel_name(), ast_channel_tech_pvt(), AST_MEDIA_TYPE_AUDIO, ast_rtp_instance_set_stats_vars(), ast_sip_session_terminate(), hangup_data::cause, hangup_data::chan, ast_sip_session::channel, clear_session_and_channel(), ast_sip_session_media_state::default_session, ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, ast_sip_channel_pvt::session, and session.

Referenced by ast_hangup(), chan_pjsip_hangup(), destroy_conference_bridge(), hangup_data_destroy(), hangup_data_init(), hangup_playback(), manage_calls(), play_on_channel(), playback_final_update(), and sla_stop_ringing_station().

◆ hangup_cause2sip()

static int hangup_cause2sip ( int  cause)
static

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

Definition at line 2439 of file chan_pjsip.c.

2440{
2441 switch (cause) {
2442 case AST_CAUSE_UNALLOCATED: /* 1 */
2443 case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
2444 case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
2445 return 404;
2446 case AST_CAUSE_CONGESTION: /* 34 */
2447 case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
2448 return 503;
2449 case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
2450 return 408;
2451 case AST_CAUSE_NO_ANSWER: /* 19 */
2452 case AST_CAUSE_UNREGISTERED: /* 20 */
2453 return 480;
2454 case AST_CAUSE_CALL_REJECTED: /* 21 */
2455 return 403;
2456 case AST_CAUSE_NUMBER_CHANGED: /* 22 */
2457 return 410;
2458 case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
2459 return 480;
2461 return 484;
2463 return 486;
2464 case AST_CAUSE_FAILURE:
2465 return 500;
2466 case AST_CAUSE_FACILITY_REJECTED: /* 29 */
2467 return 501;
2469 return 503;
2471 return 502;
2472 case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
2473 return 488;
2474 case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
2475 return 500;
2477 default:
2478 ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
2479 return 0;
2480 }
2481
2482 /* Never reached */
2483 return 0;
2484}
#define AST_CAUSE_SWITCH_CONGESTION
Definition: causes.h:123
#define AST_CAUSE_CONGESTION
Definition: causes.h:153
#define AST_CAUSE_UNALLOCATED
Definition: causes.h:98
#define AST_CAUSE_INTERWORKING
Definition: causes.h:146
#define AST_CAUSE_NUMBER_CHANGED
Definition: causes.h:112
#define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL
Definition: causes.h:130
#define AST_CAUSE_INVALID_NUMBER_FORMAT
Definition: causes.h:116
#define AST_CAUSE_CHAN_NOT_IMPLEMENTED
Definition: causes.h:132
#define AST_CAUSE_DESTINATION_OUT_OF_ORDER
Definition: causes.h:115
#define AST_CAUSE_NO_USER_RESPONSE
Definition: causes.h:108
#define AST_CAUSE_NOTDEFINED
Definition: causes.h:155
#define AST_CAUSE_FACILITY_REJECTED
Definition: causes.h:117
#define AST_CAUSE_NORMAL_UNSPECIFIED
Definition: causes.h:119
#define AST_CAUSE_NO_ROUTE_TRANSIT_NET
Definition: causes.h:99
#define AST_CAUSE_NO_ROUTE_DESTINATION
Definition: causes.h:100
#define AST_CAUSE_UNREGISTERED
Definition: causes.h:154
#define AST_CAUSE_NO_ANSWER
Definition: causes.h:109
#define AST_CAUSE_USER_BUSY
Definition: causes.h:107

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().

◆ hangup_data_alloc()

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

Definition at line 2498 of file chan_pjsip.c.

2499{
2500 struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
2501
2502 if (!h_data) {
2503 return NULL;
2504 }
2505
2506 h_data->cause = cause;
2507 h_data->chan = ast_channel_ref(chan);
2508
2509 return h_data;
2510}
#define ao2_alloc(data_size, destructor_fn)
Definition: astobj2.h:409
static void hangup_data_destroy(void *obj)
Definition: chan_pjsip.c:2491

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

Referenced by chan_pjsip_hangup().

◆ hangup_data_destroy()

static void hangup_data_destroy ( void *  obj)
static

Definition at line 2491 of file chan_pjsip.c.

2492{
2493 struct hangup_data *h_data = obj;
2494
2495 h_data->chan = ast_channel_unref(h_data->chan);
2496}

References ast_channel_unref, and hangup_data::chan.

Referenced by hangup_data_alloc().

◆ indicate()

static int indicate ( void *  data)
static

Definition at line 1338 of file chan_pjsip.c.

1339{
1340 pjsip_tx_data *packet = NULL;
1341 struct indicate_data *ind_data = data;
1342 struct ast_sip_session *session = ind_data->session;
1343 int response_code = ind_data->response_code;
1344
1345 if ((session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
1346 (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS)) {
1348 }
1349
1350 ao2_ref(ind_data, -1);
1351
1352 return 0;
1353}
struct ast_sip_session * session
Definition: chan_pjsip.c:1298

References ao2_ref, ast_sip_session_send_response(), NULL, indicate_data::response_code, indicate_data::session, and session.

Referenced by ast_channel_request_stream_topology_change(), ast_channel_stream_topology_changed(), chan_pjsip_indicate(), and indicate_data_internal().

◆ indicate_data_alloc()

static struct indicate_data * indicate_data_alloc ( struct ast_sip_session session,
int  condition,
int  response_code,
const void *  frame_data,
size_t  datalen 
)
static

Definition at line 1313 of file chan_pjsip.c.

1315{
1316 struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy);
1317
1318 if (!ind_data) {
1319 return NULL;
1320 }
1321
1322 ind_data->frame_data = ast_malloc(datalen);
1323 if (!ind_data->frame_data) {
1324 ao2_ref(ind_data, -1);
1325 return NULL;
1326 }
1327
1328 memcpy(ind_data->frame_data, frame_data, datalen);
1329 ind_data->datalen = datalen;
1330 ind_data->condition = condition;
1331 ind_data->response_code = response_code;
1332 ao2_ref(session, +1);
1333 ind_data->session = session;
1334
1335 return ind_data;
1336}
#define ast_malloc(len)
A wrapper for malloc()
Definition: astmm.h:191
static void indicate_data_destroy(void *obj)
Definition: chan_pjsip.c:1305
void * frame_data
Definition: chan_pjsip.c:1301

References ao2_alloc, ao2_ref, ast_malloc, indicate_data::condition, indicate_data::datalen, indicate_data::frame_data, indicate_data_destroy(), NULL, indicate_data::response_code, indicate_data::session, and session.

Referenced by chan_pjsip_indicate().

◆ indicate_data_destroy()

static void indicate_data_destroy ( void *  obj)
static

Definition at line 1305 of file chan_pjsip.c.

1306{
1307 struct indicate_data *ind_data = obj;
1308
1309 ast_free(ind_data->frame_data);
1310 ao2_ref(ind_data->session, -1);
1311}

References ao2_ref, ast_free, indicate_data::frame_data, and indicate_data::session.

Referenced by indicate_data_alloc().

◆ info_dtmf_data_alloc()

static struct info_dtmf_data * info_dtmf_data_alloc ( struct ast_sip_session session,
char  digit,
unsigned int  duration 
)
static

Definition at line 2242 of file chan_pjsip.c.

2243{
2244 struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
2245 if (!dtmf_data) {
2246 return NULL;
2247 }
2248 ao2_ref(session, +1);
2249 dtmf_data->session = session;
2250 dtmf_data->digit = digit;
2251 dtmf_data->duration = duration;
2252 return dtmf_data;
2253}
static void info_dtmf_data_destroy(void *obj)
Definition: chan_pjsip.c:2236
struct ast_sip_session * session
Definition: chan_pjsip.c:2231

References ao2_alloc, ao2_ref, digit, info_dtmf_data::digit, info_dtmf_data::duration, info_dtmf_data_destroy(), NULL, info_dtmf_data::session, and session.

Referenced by chan_pjsip_digit_end(), and chan_pjsip_indicate().

◆ info_dtmf_data_destroy()

static void info_dtmf_data_destroy ( void *  obj)
static

Definition at line 2236 of file chan_pjsip.c.

2237{
2238 struct info_dtmf_data *dtmf_data = obj;
2239 ao2_ref(dtmf_data->session, -1);
2240}

References ao2_ref, and info_dtmf_data::session.

Referenced by info_dtmf_data_alloc().

◆ is_colp_update_allowed()

static int is_colp_update_allowed ( struct ast_sip_session session)
static

Definition at line 1407 of file chan_pjsip.c.

1408{
1409 struct ast_party_id connected_id;
1410 int update_allowed = 0;
1411
1412 if (!session->endpoint->id.send_connected_line
1413 || (!session->endpoint->id.send_pai && !session->endpoint->id.send_rpid)) {
1414 return 0;
1415 }
1416
1417 /*
1418 * Check if privacy allows the update. Check while the channel
1419 * is locked so we can work with the shallow connected_id copy.
1420 */
1421 ast_channel_lock(session->channel);
1422 connected_id = ast_channel_connected_effective_id(session->channel);
1423 if (connected_id.number.valid
1424 && (session->endpoint->id.trust_outbound
1426 update_allowed = 1;
1427 }
1428 ast_channel_unlock(session->channel);
1429
1430 return update_allowed;
1431}
#define AST_PRES_ALLOWED
Definition: callerid.h:432
#define AST_PRES_RESTRICTION
Definition: callerid.h:431
int ast_party_id_presentation(const struct ast_party_id *id)
Determine the overall presentation value for the given party.
Definition: channel.c:1840
struct ast_party_id ast_channel_connected_effective_id(struct ast_channel *chan)
Information needed to identify an endpoint in a call.
Definition: channel.h:340

References ast_channel_connected_effective_id(), ast_channel_lock, ast_channel_unlock, ast_party_id_presentation(), AST_PRES_ALLOWED, AST_PRES_RESTRICTION, ast_party_id::number, session, and ast_party_number::valid.

Referenced by update_connected_line_information().

◆ is_compatible_format()

static int is_compatible_format ( struct ast_sip_session session,
struct ast_frame f 
)
static

Determine if the given frame is in a format we've negotiated.

Definition at line 829 of file chan_pjsip.c.

830{
831 struct ast_stream_topology *topology = session->active_media_state->topology;
832 struct ast_stream *stream = ast_stream_topology_get_stream(topology, f->stream_num);
833 const struct ast_format_cap *cap = ast_stream_get_formats(stream);
834
836}
struct ast_stream * ast_stream_topology_get_stream(const struct ast_stream_topology *topology, unsigned int position)
Get a specific stream from the topology.
Definition: stream.c:788
const struct ast_format_cap * ast_stream_get_formats(const struct ast_stream *stream)
Get the current negotiated formats of a stream.
Definition: stream.c:330

References ast_format_cap_iscompatible_format(), AST_FORMAT_CMP_NOT_EQUAL, ast_stream_get_formats(), ast_stream_topology_get_stream(), ast_frame_subclass::format, session, ast_frame::stream_num, and ast_frame::subclass.

Referenced by chan_pjsip_read_stream().

◆ load_module()

static int load_module ( void  )
static

Load the module.

Module loading including tests for configuration or dependencies. This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE, or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails tests return AST_MODULE_LOAD_FAILURE. If the module can not load the configuration file or other non-critical problem return AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.

Definition at line 3290 of file chan_pjsip.c.

3291{
3292 struct ao2_container *endpoints;
3293
3296 }
3297
3299
3301
3303 ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
3304 goto end;
3305 }
3306
3308 ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
3309 goto end;
3310 }
3311
3313 ast_log(LOG_ERROR, "Unable to register PJSIP_PARSE_URI dialplan function\n");
3314 goto end;
3315 }
3316
3318 ast_log(LOG_ERROR, "Unable to register PJSIP_PARSE_URI_FROM dialplan function\n");
3319 goto end;
3320 }
3321
3323 ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
3324 goto end;
3325 }
3326
3328 ast_log(LOG_WARNING, "Unable to register PJSIP_DTMF_MODE dialplan function\n");
3329 goto end;
3330 }
3331
3333 ast_log(LOG_WARNING, "Unable to register PJSIP_MOH_PASSTHROUGH dialplan function\n");
3334 goto end;
3335 }
3336
3338 ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n");
3339 goto end;
3340 }
3341
3343 ast_log(LOG_WARNING, "Unable to register PJSIPHangup dialplan application\n");
3344 goto end;
3345 }
3347
3348
3350
3353
3357 ast_log(LOG_ERROR, "Unable to create held channels container\n");
3358 goto end;
3359 }
3360
3365
3367 ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
3368 goto end;
3369 }
3370
3371 /* since endpoints are loaded before the channel driver their device
3372 states get set to 'invalid', so they need to be updated */
3373 if ((endpoints = ast_sip_get_endpoints())) {
3375 ao2_ref(endpoints, -1);
3376 }
3377
3378 return 0;
3379
3380end:
3399
3402
3404}
@ AO2_ALLOC_OPT_LOCK_RWLOCK
Definition: astobj2.h:365
#define ao2_callback(c, flags, cb_fn, arg)
ao2_callback() is a generic function that applies cb_fn() to all objects in a container,...
Definition: astobj2.h:1693
#define ao2_container_alloc_hash(ao2_options, container_options, n_buckets, hash_fn, sort_fn, cmp_fn)
Allocate and initialize a hash container with the desired number of buckets.
Definition: astobj2.h:1303
@ AO2_CONTAINER_ALLOC_OPT_DUPS_REJECT
Reject objects with duplicate keys in container.
Definition: astobj2.h:1188
static int uid_hold_sort_fn(const void *obj_left, const void *obj_right, const int flags)
Definition: chan_pjsip.c:1089
static char * app_pjsip_hangup
Definition: chan_pjsip.c:3278
static int update_devstate(void *obj, void *arg, int flags)
Definition: chan_pjsip.c:3233
static struct ast_sip_session_supplement chan_pjsip_supplement
SIP session supplement structure.
Definition: chan_pjsip.c:143
static struct ast_custom_function chan_pjsip_dial_contacts_function
Definition: chan_pjsip.c:3240
static int uid_hold_hash_fn(const void *obj, const int flags)
Definition: chan_pjsip.c:1071
static struct ast_sip_session_supplement pbx_start_supplement
Definition: chan_pjsip.c:3114
static struct ast_custom_function chan_pjsip_parse_uri_from_function
Definition: chan_pjsip.c:3250
static struct ast_sip_session_supplement chan_pjsip_prack_supplement
Definition: chan_pjsip.c:172
static struct ast_rtp_glue chan_pjsip_rtp_glue
Local glue for interacting with the RTP engine core.
Definition: chan_pjsip.c:486
static pjsip_module refer_callback_module
REFER Callback module, used to attach session data structure to subscription.
Definition: chan_pjsip.c:1952
static const char channel_type[]
Definition: chan_pjsip.c:78
static struct ast_custom_function moh_passthrough_function
Definition: chan_pjsip.c:3267
static struct ast_custom_function media_offer_function
Definition: chan_pjsip.c:3255
static struct ast_sip_session_supplement chan_pjsip_ack_supplement
Definition: chan_pjsip.c:164
static struct ast_custom_function chan_pjsip_parse_uri_function
Definition: chan_pjsip.c:3245
static struct ast_custom_function session_refresh_function
Definition: chan_pjsip.c:3273
static struct ast_sip_session_supplement chan_pjsip_supplement_response
SIP session supplement structure just for responses.
Definition: chan_pjsip.c:155
static struct ast_sip_session_supplement call_pickup_supplement
Definition: chan_pjsip.c:3074
static struct ast_custom_function dtmf_mode_function
Definition: chan_pjsip.c:3261
void ast_channel_unregister(const struct ast_channel_tech *tech)
Unregister a channel technology.
Definition: channel.c:570
int ast_channel_register(const struct ast_channel_tech *tech)
Register a channel technology (a new channel driver) Called by a channel module to register the kind ...
Definition: channel.c:539
int pjsip_channel_cli_register(void)
Registers the channel cli commands.
Definition: cli_commands.c:462
int pjsip_app_hangup(struct ast_channel *chan, const char *data)
PJSIPHangup Dialplan App.
int pjsip_action_hangup(struct mansession *s, const struct message *m)
PJSIPHangup Manager Action.
char * end
Definition: eagi_proxy.c:73
int ast_format_cap_append_by_type(struct ast_format_cap *cap, enum ast_media_type type)
Add all codecs Asterisk knows about for a specific type to the capabilities structure.
Definition: format_cap.c:216
int ast_manager_unregister(const char *action)
Unregister a registered manager command.
Definition: manager.c:7606
static struct ao2_container * endpoints
#define EVENT_FLAG_SYSTEM
Definition: manager.h:75
#define ast_manager_register_xml(action, authority, func)
Register a manager callback using XML documentation to describe the manager.
Definition: manager.h:191
#define EVENT_FLAG_CALL
Definition: manager.h:76
int ast_unregister_application(const char *app)
Unregister an application.
Definition: pbx_app.c:392
@ AST_MODULE_LOAD_DECLINE
Module has failed to load, may be in an inconsistent state.
Definition: module.h:78
#define ast_register_application_xml(app, execute)
Register an application using XML documentation.
Definition: module.h:640
#define ast_custom_function_register(acf)
Register a custom function.
Definition: pbx.h:1558
int ast_custom_function_unregister(struct ast_custom_function *acf)
Unregister a custom function.
void ast_sip_unregister_service(pjsip_module *module)
Definition: res_pjsip.c:133
struct ao2_container * ast_sip_get_endpoints(void)
Retrieve any endpoints available to sorcery.
int ast_sip_register_service(pjsip_module *module)
Register a SIP service in Asterisk.
Definition: res_pjsip.c:117
#define ast_sip_session_register_supplement(supplement)
void ast_sip_session_unregister_supplement(struct ast_sip_session_supplement *supplement)
Unregister a an supplement to SIP session processing.
Definition: pjsip_session.c:63
#define ast_rtp_glue_register(glue)
Definition: rtp_engine.h:905
int ast_rtp_glue_unregister(struct ast_rtp_glue *glue)
Unregister RTP glue.
Definition: rtp_engine.c:408
Generic container type.
struct ast_format_cap * capabilities
Definition: channel.h:652

References AO2_ALLOC_OPT_LOCK_RWLOCK, ao2_callback, ao2_cleanup, ao2_container_alloc_hash, AO2_CONTAINER_ALLOC_OPT_DUPS_REJECT, ao2_ref, app_pjsip_hangup, ast_channel_register(), ast_channel_unregister(), ast_custom_function_register, ast_custom_function_unregister(), ast_format_cap_alloc, ast_format_cap_append_by_type(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_log, ast_manager_register_xml, ast_manager_unregister(), AST_MEDIA_TYPE_AUDIO, AST_MODULE_LOAD_DECLINE, ast_register_application_xml, ast_rtp_glue_register, ast_rtp_glue_unregister(), ast_sip_get_endpoints(), ast_sip_register_service(), ast_sip_session_register_supplement, ast_sip_session_unregister_supplement(), ast_sip_unregister_service(), ast_unregister_application(), call_pickup_supplement, ast_channel_tech::capabilities, chan_pjsip_ack_supplement, chan_pjsip_dial_contacts_function, chan_pjsip_parse_uri_from_function, chan_pjsip_parse_uri_function, chan_pjsip_prack_supplement, chan_pjsip_rtp_glue, chan_pjsip_supplement, chan_pjsip_supplement_response, chan_pjsip_tech, channel_type, dtmf_mode_function, end, endpoints, EVENT_FLAG_CALL, EVENT_FLAG_SYSTEM, LOG_ERROR, LOG_WARNING, media_offer_function, moh_passthrough_function, NULL, OBJ_NODATA, pbx_start_supplement, pjsip_action_hangup(), pjsip_app_hangup(), pjsip_channel_cli_register(), pjsip_uids_onhold, refer_callback_module, session_refresh_function, uid_hold_hash_fn(), uid_hold_sort_fn(), and update_devstate().

◆ on_topology_change_response()

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

Definition at line 1549 of file chan_pjsip.c.

1550{
1551 SCOPE_ENTER(3, "%s: Received response code %d. PT: %s AT: %s\n", ast_sip_session_get_name(session),
1552 rdata->msg_info.msg->line.status.code,
1553 ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)),
1554 ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
1555
1556
1557 if (PJSIP_IS_STATUS_IN_CLASS(rdata->msg_info.msg->line.status.code, 200)) {
1558 /* The topology was changed to something new so give notice to what requested
1559 * it so it queries the channel and updates accordingly.
1560 */
1561 if (session->channel) {
1563 SCOPE_EXIT_RTN_VALUE(0, "%s: Queued topology change frame\n", ast_sip_session_get_name(session));
1564 }
1565 SCOPE_EXIT_RTN_VALUE(0, "%s: No channel? Can't queue topology change frame\n", ast_sip_session_get_name(session));
1566 } else if (300 <= rdata->msg_info.msg->line.status.code) {
1567 /* The topology change failed, so drop the current pending media state */
1568 ast_sip_session_media_state_reset(session->pending_media_state);
1569 SCOPE_EXIT_RTN_VALUE(0, "%s: response code > 300. Resetting pending media state\n", ast_sip_session_get_name(session));
1570 }
1571
1572 SCOPE_EXIT_RTN_VALUE(0, "%s: Nothing to do\n", ast_sip_session_get_name(session));
1573}
void ast_sip_session_media_state_reset(struct ast_sip_session_media_state *media_state)
Reset a media state to a clean state.

References AST_CONTROL_STREAM_TOPOLOGY_CHANGED, ast_queue_control(), ast_sip_session_get_name(), ast_sip_session_media_state_reset(), ast_str_tmp, ast_stream_topology_to_str(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, and session.

Referenced by send_topology_change_refresh().

◆ pbx_start_incoming_request()

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

Definition at line 3080 of file chan_pjsip.c.

3081{
3082 int res;
3084
3085 /* Check for a to-tag to determine if this is a reinvite */
3086 if (rdata->msg_info.to->tag.slen) {
3087 /* We don't care about reinvites */
3088 SCOPE_EXIT_RTN_VALUE(0, "Reinvite\n");
3089 }
3090
3091 res = ast_pbx_start(session->channel);
3092
3093 switch (res) {
3094 case AST_PBX_FAILED:
3095 ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
3097 ast_hangup(session->channel);
3098 break;
3099 case AST_PBX_CALL_LIMIT:
3100 ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
3102 ast_hangup(session->channel);
3103 break;
3104 case AST_PBX_SUCCESS:
3105 default:
3106 break;
3107 }
3108
3109 ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
3110
3111 SCOPE_EXIT_RTN_VALUE((res == AST_PBX_SUCCESS) ? 0 : -1, "RC: %d\n", res);
3112}
@ AST_PBX_FAILED
Definition: pbx.h:372
@ AST_PBX_CALL_LIMIT
Definition: pbx.h:373
@ AST_PBX_SUCCESS
Definition: pbx.h:371
enum ast_pbx_result ast_pbx_start(struct ast_channel *c)
Create a new thread and start the PBX.
Definition: pbx.c:4708

References AST_CAUSE_SWITCH_CONGESTION, ast_channel_hangupcause_set(), ast_channel_name(), ast_debug, ast_hangup(), ast_log, AST_PBX_CALL_LIMIT, AST_PBX_FAILED, ast_pbx_start(), AST_PBX_SUCCESS, ast_sip_session_get_name(), LOG_WARNING, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, and session.

◆ remote_send_hold()

static int remote_send_hold ( void *  data)
static

Update local hold state to be held.

Definition at line 1500 of file chan_pjsip.c.

1501{
1502 return remote_send_hold_refresh(data, 1);
1503}
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.
Definition: chan_pjsip.c:1487

References remote_send_hold_refresh().

Referenced by chan_pjsip_indicate().

◆ remote_send_hold_refresh()

static int remote_send_hold_refresh ( struct ast_sip_session session,
unsigned int  held 
)
static

Update local hold state and send a re-INVITE with the new SDP.

Definition at line 1487 of file chan_pjsip.c.

1488{
1489 struct ast_sip_session_media *session_media = session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
1490 if (session_media) {
1491 session_media->locally_held = held;
1492 }
1494 ao2_ref(session, -1);
1495
1496 return 0;
1497}
@ AST_SIP_SESSION_REFRESH_METHOD_INVITE
Definition: res_pjsip.h:625
int ast_sip_session_refresh(struct ast_sip_session *session, ast_sip_session_request_creation_cb on_request_creation, ast_sip_session_sdp_creation_cb on_sdp_creation, ast_sip_session_response_cb on_response, enum ast_sip_session_refresh_method method, int generate_new_sdp, struct ast_sip_session_media_state *media_state)
Send a reinvite or UPDATE on a session.
unsigned int locally_held
Stream is on hold by local side.

References ao2_ref, AST_MEDIA_TYPE_AUDIO, ast_sip_session_refresh(), AST_SIP_SESSION_REFRESH_METHOD_INVITE, ast_sip_session_media::locally_held, NULL, and session.

Referenced by remote_send_hold(), and remote_send_unhold().

◆ remote_send_unhold()

static int remote_send_unhold ( void *  data)
static

Update local hold state to be unheld.

Definition at line 1506 of file chan_pjsip.c.

1507{
1508 return remote_send_hold_refresh(data, 0);
1509}

References remote_send_hold_refresh().

Referenced by chan_pjsip_indicate().

◆ request()

static int request ( void *  obj)
static

Definition at line 2605 of file chan_pjsip.c.

2606{
2607 struct request_data *req_data = obj;
2608 struct ast_sip_session *session = NULL;
2609 char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
2610 struct ast_sip_endpoint *endpoint;
2611
2613 AST_APP_ARG(endpoint);
2614 AST_APP_ARG(aor);
2615 );
2616 SCOPE_ENTER(1, "%s\n",tmp);
2617
2618 if (ast_strlen_zero(tmp)) {
2619 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
2621 SCOPE_EXIT_RTN_VALUE(-1, "Empty destination\n");
2622 }
2623
2625
2627 /* If a request user has been specified extract it from the endpoint name portion */
2628 if ((endpoint_name = strchr(args.endpoint, '@'))) {
2629 request_user = args.endpoint;
2630 *endpoint_name++ = '\0';
2631 } else {
2632 endpoint_name = args.endpoint;
2633 }
2634
2635 if (ast_strlen_zero(endpoint_name)) {
2636 if (request_user) {
2637 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@<endpoint-name>\n",
2638 request_user);
2639 } else {
2640 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
2641 }
2643 SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
2644 }
2645 endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
2646 endpoint_name);
2647 if (!endpoint) {
2648 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
2650 SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
2651 }
2652 } else {
2653 /* First try to find an exact endpoint match, for single (user) or multi-domain (user@domain) */
2654 endpoint_name = args.endpoint;
2655 if (ast_strlen_zero(endpoint_name)) {
2656 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
2658 SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
2659 }
2660 endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
2661 endpoint_name);
2662 if (!endpoint) {
2663 /* It seems it's not a multi-domain endpoint or single endpoint exact match,
2664 * it's possible that it's a SIP trunk with a specified user (user@trunkname),
2665 * so extract the user before @ sign.
2666 */
2667 endpoint_name = strchr(args.endpoint, '@');
2668 if (!endpoint_name) {
2669 /*
2670 * Couldn't find an '@' so it had to be an endpoint
2671 * name that doesn't exist.
2672 */
2673 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n",
2674 args.endpoint);
2676 SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
2677 }
2678 request_user = args.endpoint;
2679 *endpoint_name++ = '\0';
2680
2681 if (ast_strlen_zero(endpoint_name)) {
2682 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@<endpoint-name>\n",
2683 request_user);
2685 SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
2686 }
2687
2688 endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
2689 endpoint_name);
2690 if (!endpoint) {
2691 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
2693 SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
2694 }
2695 }
2696 }
2697
2698 session = ast_sip_session_create_outgoing(endpoint, NULL, args.aor, request_user,
2699 req_data->topology);
2700 ao2_ref(endpoint, -1);
2701 if (!session) {
2702 ast_log(LOG_ERROR, "Failed to create outgoing session to endpoint '%s'\n", endpoint_name);
2704 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create session\n");
2705 }
2706
2707 req_data->session = session;
2708
2710}
#define ast_strdupa(s)
duplicate a string in memory from the stack
Definition: astmm.h:298
static int tmp()
Definition: bt_open.c:389
#define AST_CAUSE_CHANNEL_UNACCEPTABLE
Definition: causes.h:102
#define AST_APP_ARG(name)
Define an application argument.
#define AST_DECLARE_APP_ARGS(name, arglist)
Declare a structure to hold an application's arguments.
#define AST_NONSTANDARD_APP_ARGS(args, parse, sep)
Performs the 'nonstandard' argument separation process for an application.
unsigned int ast_sip_get_disable_multi_domain(void)
Retrieve the system setting 'disable multi domain'.
struct ast_sip_session * ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint, struct ast_sip_contact *contact, const char *location, const char *request_user, struct ast_stream_topology *req_topology)
Create a new outgoing SIP session.
struct ast_sip_session * session
Definition: chan_pjsip.c:2599
const char * dest
Definition: chan_pjsip.c:2601
const char * args

References ao2_ref, args, AST_APP_ARG, AST_CAUSE_CHANNEL_UNACCEPTABLE, AST_CAUSE_NO_ROUTE_DESTINATION, AST_DECLARE_APP_ARGS, ast_log, AST_NONSTANDARD_APP_ARGS, ast_sip_get_disable_multi_domain(), ast_sip_get_sorcery(), ast_sip_session_create_outgoing(), ast_sorcery_retrieve_by_id(), ast_strdupa, ast_strlen_zero(), request_data::cause, request_data::dest, LOG_ERROR, NULL, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, request_data::session, session, tmp(), and request_data::topology.

Referenced by ast_bridge_channel_merge_inhibit(), ast_bridge_merge_inhibit(), ast_http_body_discard(), ast_http_body_read_status(), ast_http_get_contents(), ast_http_request_close_on_completion(), ast_http_send(), ast_parse_digest(), AST_TEST_DEFINE(), bridge_manager_destroy(), bridge_manager_service_req(), bridge_manager_thread(), bridge_merge_inhibit_nolock(), chan_pjsip_request_with_stream_topology(), ewscal_write_event(), get_ewscal_ids_for(), http_request_tracking_init(), http_request_tracking_setup(), httpd_process_request(), parse_ewscal_id(), run_agi(), send_ews_request_and_parse(), setup_env(), xmpp_pubsub_build_node_request(), xmpp_pubsub_build_publish_skeleton(), xmpp_pubsub_delete_node(), xmpp_pubsub_handle_error(), xmpp_pubsub_iq_create(), xmpp_pubsub_publish_device_state(), xmpp_pubsub_publish_mwi(), xmpp_pubsub_purge_nodes(), xmpp_pubsub_request_nodes(), xmpp_pubsub_subscribe(), and xmpp_pubsub_unsubscribe().

◆ rtp_direct_media_data_create()

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 
)
static

Definition at line 377 of file chan_pjsip.c.

380{
382
383 if (!cdata) {
384 return NULL;
385 }
386
387 cdata->chan = ao2_bump(chan);
388 cdata->rtp = ao2_bump(rtp);
389 cdata->vrtp = ao2_bump(vrtp);
390 cdata->cap = ao2_bump((struct ast_format_cap *)cap);
391 cdata->session = ao2_bump(session);
392
393 return cdata;
394}
static void rtp_direct_media_data_destroy(void *data)
Definition: chan_pjsip.c:366

References ao2_alloc, ao2_bump, rtp_direct_media_data::cap, cdata(), rtp_direct_media_data::chan, NULL, rtp_direct_media_data::rtp, rtp_direct_media_data_destroy(), session, and rtp_direct_media_data::vrtp.

Referenced by chan_pjsip_set_rtp_peer().

◆ rtp_direct_media_data_destroy()

static void rtp_direct_media_data_destroy ( void *  data)
static

Definition at line 366 of file chan_pjsip.c.

367{
368 struct rtp_direct_media_data *cdata = data;
369
370 ao2_cleanup(cdata->session);
371 ao2_cleanup(cdata->cap);
372 ao2_cleanup(cdata->vrtp);
373 ao2_cleanup(cdata->rtp);
374 ao2_cleanup(cdata->chan);
375}

References ao2_cleanup, and cdata().

Referenced by rtp_direct_media_data_create().

◆ rtp_find_rtcp_fd_position()

static int rtp_find_rtcp_fd_position ( struct ast_sip_session session,
struct ast_rtp_instance rtp 
)
static

Helper function to find the position for RTCP.

Definition at line 306 of file chan_pjsip.c.

307{
308 int index;
309
310 for (index = 0; index < AST_VECTOR_SIZE(&session->active_media_state->read_callbacks); ++index) {
311 struct ast_sip_session_media_read_callback_state *callback_state =
312 AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, index);
313
314 if (callback_state->fd != ast_rtp_instance_fd(rtp, 1)) {
315 continue;
316 }
317
318 return index;
319 }
320
321 return -1;
322}

References ast_rtp_instance_fd(), AST_VECTOR_GET_ADDR, AST_VECTOR_SIZE, ast_sip_session_media_read_callback_state::fd, and session.

Referenced by check_for_rtp_changes().

◆ send_direct_media_request()

static int send_direct_media_request ( void *  data)
static

Definition at line 396 of file chan_pjsip.c.

397{
398 struct rtp_direct_media_data *cdata = data;
399 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(cdata->chan);
400 struct ast_sip_session *session;
401 int changed = 0;
402 int res = 0;
403
404 /* XXX In an ideal world each media stream would be direct, but for now preserve behavior
405 * and connect only the default media sessions for audio and video.
406 */
407
408 /* The channel needs to be locked when checking for RTP changes.
409 * Otherwise, we could end up destroying an underlying RTCP structure
410 * at the same time that the channel thread is attempting to read RTCP
411 */
412 ast_channel_lock(cdata->chan);
413 session = channel->session;
414 if (session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
415 changed |= check_for_rtp_changes(
416 cdata->chan, cdata->rtp, session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO], session);
417 }
418 if (session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO]) {
419 changed |= check_for_rtp_changes(
420 cdata->chan, cdata->vrtp, session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO], session);
421 }
423
424 if (direct_media_mitigate_glare(cdata->session)) {
425 ast_debug(4, "Disregarding setting RTP on %s: mitigating re-INVITE glare\n", ast_channel_name(cdata->chan));
426 ao2_ref(cdata, -1);
427 return 0;
428 }
429
430 if (cdata->cap && ast_format_cap_count(cdata->cap) &&
431 !ast_format_cap_identical(cdata->session->direct_media_cap, cdata->cap)) {
433 ast_format_cap_append_from_cap(cdata->session->direct_media_cap, cdata->cap, AST_MEDIA_TYPE_UNKNOWN);
434 changed = 1;
435 }
436
437 if (changed) {
438 ast_debug(4, "RTP changed on %s; initiating direct media update\n", ast_channel_name(cdata->chan));
439 res = ast_sip_session_refresh(cdata->session, NULL, NULL, NULL,
440 cdata->session->endpoint->media.direct_media.method, 1, NULL);
441 }
442
443 ao2_ref(cdata, -1);
444 return res;
445}
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)
Definition: chan_pjsip.c:327
static int direct_media_mitigate_glare(struct ast_sip_session *session)
Definition: chan_pjsip.c:276
int ast_format_cap_identical(const struct ast_format_cap *cap1, const struct ast_format_cap *cap2)
Determine if two capabilities structures are identical.
Definition: format_cap.c:687
size_t ast_format_cap_count(const struct ast_format_cap *cap)
Get the number of formats present within the capabilities structure.
Definition: format_cap.c:395

References ao2_ref, ast_channel_lock, ast_channel_name(), ast_channel_tech_pvt(), ast_channel_unlock, ast_debug, ast_format_cap_append_from_cap(), ast_format_cap_count(), ast_format_cap_identical(), ast_format_cap_remove_by_type(), AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_UNKNOWN, AST_MEDIA_TYPE_VIDEO, ast_sip_session_refresh(), cdata(), ast_sip_session::channel, check_for_rtp_changes(), direct_media_mitigate_glare(), NULL, and session.

Referenced by chan_pjsip_set_rtp_peer().

◆ send_topology_change_refresh()

static int send_topology_change_refresh ( void *  data)
static

Definition at line 1575 of file chan_pjsip.c.

1576{
1581 int ret;
1583 ast_str_tmp(256, ast_stream_topology_to_str(refresh_data->media_state->topology, &STR_TMP)));
1584
1585 /* See RFC 6337, especially section 3.2: If the early media SDP was sent reliably, we are allowed
1586 * to send UPDATEs. Only relevant for AST_STATE_RINGING and AST_STATE_RING - if the channel is UP,
1587 * re-INVITES can be sent.
1588 */
1589 if (session->early_confirmed && (state == AST_STATE_RINGING || state == AST_STATE_RING)) {
1591 }
1592
1594 method, 1, refresh_data->media_state);
1595 refresh_data->media_state = NULL;
1597
1599}
static int on_topology_change_response(struct ast_sip_session *session, pjsip_rx_data *rdata)
Definition: chan_pjsip.c:1549
const char * method
Definition: res_pjsip.c:1279
ast_sip_session_refresh_method
Definition: res_pjsip.h:623
@ AST_SIP_SESSION_REFRESH_METHOD_UPDATE
Definition: res_pjsip.h:627
struct ast_sip_session * session

References ast_sip_session_get_name(), ast_sip_session_refresh(), AST_SIP_SESSION_REFRESH_METHOD_INVITE, AST_SIP_SESSION_REFRESH_METHOD_UPDATE, AST_STATE_RING, AST_STATE_RINGING, ast_str_tmp, ast_stream_topology_to_str(), method, NULL, on_topology_change_response(), SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, refresh_data::session, session, and topology_change_refresh_data_free().

Referenced by handle_topology_request_change().

◆ sendtext()

static int sendtext ( void *  obj)
static

Definition at line 2790 of file chan_pjsip.c.

2791{
2792 struct sendtext_data *data = obj;
2793 pjsip_tx_data *tdata;
2794 const char *body_text = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_BODY);
2795 const char *content_type = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_CONTENT_TYPE);
2796 char *sep;
2797 struct ast_sip_body body = {
2798 .type = "text",
2799 .subtype = "plain",
2800 .body_text = body_text,
2801 };
2802
2803 if (!ast_strlen_zero(content_type)) {
2804 sep = strchr(content_type, '/');
2805 if (sep) {
2806 *sep = '\0';
2807 body.type = content_type;
2808 body.subtype = ++sep;
2809 }
2810 }
2811
2812 if (data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
2813 ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
2814 data->session->