Asterisk - The Open Source Telephony Project GIT-master-f36a736
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 3445 of file chan_pjsip.c.

◆ __unreg_module()

static void __unreg_module ( void  )
static

Definition at line 3445 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 3445 of file chan_pjsip.c.

◆ call()

static int call ( void *  data)
static

Definition at line 2394 of file chan_pjsip.c.

2395{
2396 struct ast_sip_channel_pvt *channel = data;
2397 struct ast_sip_session *session = channel->session;
2398 pjsip_tx_data *tdata;
2399 int res = 0;
2400 SCOPE_ENTER(1, "%s Topology: %s\n",
2402 ast_str_tmp(256, ast_stream_topology_to_str(channel->session->pending_media_state->topology, &STR_TMP))
2403 );
2404
2405
2407
2408 if (res) {
2409 ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
2410 ast_queue_hangup(session->channel);
2411 } else {
2415 }
2416 ao2_ref(channel, -1);
2417 SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
2418}
#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:2369
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:939
#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 3029 of file chan_pjsip.c.

3030{
3031 struct ast_features_pickup_config *pickup_cfg;
3032 struct ast_channel *chan;
3033
3034 /* Check for a to-tag to determine if this is a reinvite */
3035 if (rdata->msg_info.to->tag.slen) {
3036 /* We don't care about reinvites */
3037 return 0;
3038 }
3039
3040 pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
3041 if (!pickup_cfg) {
3042 ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n");
3043 return 0;
3044 }
3045
3046 if (strcmp(session->exten, pickup_cfg->pickupexten)) {
3047 ao2_ref(pickup_cfg, -1);
3048 return 0;
3049 }
3050 ao2_ref(pickup_cfg, -1);
3051
3052 /* We can't directly use session->channel because the pickup operation will cause a masquerade to occur,
3053 * changing the channel pointer in session to a different channel. To ensure we work on the right channel
3054 * we store a pointer locally before we begin and keep a reference so it remains valid no matter what.
3055 */
3056 chan = ast_channel_ref(session->channel);
3057 if (ast_pickup_call(chan)) {
3059 } else {
3061 }
3062 /* A hangup always occurs because the pickup operation will have either failed resulting in the call
3063 * needing to be hung up OR the pickup operation was a success and the channel we now have is actually
3064 * the channel that was replaced, which should be hung up since it is literally in limbo not connected
3065 * to anything at all.
3066 */
3067 ast_hangup(chan);
3068 ast_channel_unref(chan);
3069
3070 return 1;
3071}
#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 2421 of file chan_pjsip.c.

2422{
2423 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2424 SCOPE_ENTER(1, "%s Topology: %s\n", ast_sip_session_get_name(channel->session),
2426
2427 ao2_ref(channel, +1);
2428 if (ast_sip_push_task(channel->session->serializer, call, channel)) {
2429 ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest);
2430 ao2_cleanup(channel);
2431 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
2432 }
2433
2434 SCOPE_EXIT_RTN_VALUE(0, "'call' task pushed\n");
2435}
static int call(void *data)
Definition: chan_pjsip.c:2394
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:958

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 2186 of file chan_pjsip.c.

2187{
2188 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
2189 struct ast_sip_session_media *media;
2190
2192
2193 switch (channel->session->dtmf) {
2195 if (!media || !media->rtp) {
2196 return 0;
2197 }
2198
2200 break;
2201 case AST_SIP_DTMF_AUTO:
2202 if (!media || !media->rtp) {
2203 return 0;
2204 }
2205
2207 return -1;
2208 }
2209
2211 break;
2213 if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) {
2214 return 0;
2215 }
2217 break;
2218 case AST_SIP_DTMF_NONE:
2219 break;
2221 return -1;
2222 default:
2223 break;
2224 }
2225
2226 return 0;
2227}
char digit
@ AST_MEDIA_TYPE_AUDIO
Definition: codec.h:32
@ AST_SIP_DTMF_NONE
Definition: res_pjsip.h:542
@ AST_SIP_DTMF_AUTO_INFO
Definition: res_pjsip.h:553
@ AST_SIP_DTMF_AUTO
Definition: res_pjsip.h:551
@ AST_SIP_DTMF_INBAND
Definition: res_pjsip.h:547
@ AST_SIP_DTMF_RFC_4733
Definition: res_pjsip.h:545
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 2298 of file chan_pjsip.c.

2299{
2300 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2301 struct ast_sip_session_media *media;
2302
2303 if (!channel || !channel->session) {
2304 /* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */
2305 ast_debug(3, "Channel %s disappeared while calling digit_end\n", ast_channel_name(ast));
2306 return -1;
2307 }
2308
2310
2311 switch (channel->session->dtmf) {
2313 {
2314 if (!media || !media->rtp) {
2315 return 0;
2316 }
2317
2319 ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
2321 break;
2322 }
2323 /* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */
2324 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));
2325 }
2326
2327 case AST_SIP_DTMF_INFO:
2328 {
2329 struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
2330
2331 if (!dtmf_data) {
2332 return -1;
2333 }
2334
2335 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
2336 ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
2337 ao2_cleanup(dtmf_data);
2338 return -1;
2339 }
2340 break;
2341 }
2343 if (!media || !media->rtp) {
2344 return 0;
2345 }
2346
2348 break;
2349 case AST_SIP_DTMF_AUTO:
2350 if (!media || !media->rtp) {
2351 return 0;
2352 }
2353
2355 return -1;
2356 }
2357
2359 break;
2360 case AST_SIP_DTMF_NONE:
2361 break;
2363 return -1;
2364 }
2365
2366 return 0;
2367}
static int transmit_info_dtmf(void *data)
Definition: chan_pjsip.c:2254
static struct info_dtmf_data * info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
Definition: chan_pjsip.c:2241
#define ast_debug(level,...)
Log a DEBUG message.
@ AST_SIP_DTMF_INFO
Definition: res_pjsip.h:549
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:2232

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:644
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:910
struct ast_sip_direct_media_configuration direct_media
Definition: res_pjsip.h:912
struct ast_sip_endpoint_media_configuration media
Definition: res_pjsip.h:991
enum ast_sip_session_media_encryption encryption
Definition: res_pjsip.h:853
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 2561 of file chan_pjsip.c.

2562{
2563 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2564 int cause;
2565 struct hangup_data *h_data;
2566 SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
2567
2568 if (!channel || !channel->session) {
2569 SCOPE_EXIT_RTN_VALUE(-1, "No channel or session\n");
2570 }
2571
2573 h_data = hangup_data_alloc(cause, ast);
2574
2575 if (!h_data) {
2576 goto failure;
2577 }
2578
2579 if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
2580 ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
2581 goto failure;
2582 }
2583
2584 SCOPE_EXIT_RTN_VALUE(0, "Cause: %d\n", cause);
2585
2586failure:
2587 /* Go ahead and do our cleanup of the session and channel even if we're not going
2588 * to be able to send our SIP request/response
2589 */
2590 clear_session_and_channel(channel->session, ast);
2591 ao2_cleanup(channel);
2592 ao2_cleanup(h_data);
2593
2594 SCOPE_EXIT_RTN_VALUE(-1, "Cause: %d\n", cause);
2595}
static int hangup_cause2sip(int cause)
Internal function which translates from Asterisk cause codes to SIP response codes.
Definition: chan_pjsip.c:2438
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:2512
static int hangup(void *data)
Definition: chan_pjsip.c:2519
static struct hangup_data * hangup_data_alloc(int cause, struct ast_channel *chan)
Definition: chan_pjsip.c:2497
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 3207 of file chan_pjsip.c.

3208{
3210
3211 if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
3212 if (session->endpoint->media.direct_media.enabled && session->channel) {
3213 ast_trace(-1, "%s: Queueing SRCCHANGE\n", ast_sip_session_get_name(session));
3215 }
3216 }
3218}
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 3220 of file chan_pjsip.c.

3221{
3223
3224 if (pj_strcmp2(&rdata->msg_info.msg->line.req.method.name, "PRACK") == 0 &&
3225 pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE) {
3226
3227 session->early_confirmed = 1;
3228 }
3230}

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 2971 of file chan_pjsip.c.

2972{
2973 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
2974 struct transport_info_data *transport_data;
2975 pjsip_tx_data *packet = NULL;
2977
2978 if (session->channel) {
2979 SCOPE_EXIT_RTN_VALUE(0, "%s: No channel\n", ast_sip_session_get_name(session));
2980 }
2981
2982 /* Check for a to-tag to determine if this is a reinvite */
2983 if (rdata->msg_info.to->tag.slen) {
2984 /* Weird case. We've received a reinvite but we don't have a channel. The most
2985 * typical case for this happening is that a blind transfer fails, and so the
2986 * transferer attempts to reinvite himself back into the call. We already got
2987 * rid of that channel, and the other side of the call is unrecoverable.
2988 *
2989 * We treat this as a failure, so our best bet is to just hang this call
2990 * up and not create a new channel. Clearing defer_terminate here ensures that
2991 * calling ast_sip_session_terminate() can result in a BYE being sent ASAP.
2992 */
2993 session->defer_terminate = 0;
2995 SCOPE_EXIT_RTN_VALUE(-1, "%s: We have a To tag but no channel. Terminating session\n", ast_sip_session_get_name(session));
2996 }
2997
2998 datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
2999 if (!datastore) {
3000 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info datastore\n", ast_sip_session_get_name(session));
3001 }
3002
3003 transport_data = ast_calloc(1, sizeof(*transport_data));
3004 if (!transport_data) {
3005 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info\n", ast_sip_session_get_name(session));
3006 }
3007 pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
3008 pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
3009 datastore->data = transport_data;
3011
3012 if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL, NULL))) {
3013 if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS
3014 && packet) {
3016 }
3017
3018 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n",
3020 }
3021
3023
3024 /* channel gets created on incoming request, but we wait to call start
3025 so other supplements have a chance to run */
3027}
#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:2958
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 3150 of file chan_pjsip.c.

3151{
3152 struct pjsip_status_line status = rdata->msg_info.msg->line.status;
3153 SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
3154
3155 if (!session->channel) {
3156 SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
3157 }
3158
3159 switch (status.code) {
3160 case 180: {
3161 pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
3162 if (sdp && sdp->body.ptr) {
3163 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
3164 session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
3166 } else {
3167 ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
3169 }
3170
3171 ast_channel_lock(session->channel);
3172 if (ast_channel_state(session->channel) != AST_STATE_UP) {
3174 }
3175 ast_channel_unlock(session->channel);
3176 break;
3177 }
3178 case 183:
3179 if (session->endpoint->ignore_183_without_sdp) {
3180 pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
3181 if (sdp && sdp->body.ptr) {
3182 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
3183 ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS with SDP\n", ast_sip_session_get_name(session),
3184 (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
3185 session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
3187 }
3188 } else {
3189 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
3190 ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS without SDP\n", ast_sip_session_get_name(session),
3191 (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
3193 }
3194 break;
3195 case 200:
3196 ast_trace(-1, "%s: Queueing ANSWER\n", ast_sip_session_get_name(session));
3198 break;
3199 default:
3200 ast_trace(-1, "%s: Not queueing anything\n", ast_sip_session_get_name(session));
3201 break;
3202 }
3203
3205}
@ 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 3120 of file chan_pjsip.c.

3121{
3122 struct pjsip_status_line status = rdata->msg_info.msg->line.status;
3123 struct ast_control_pvt_cause_code *cause_code;
3124 int data_size = sizeof(*cause_code);
3125 SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
3126
3127 if (!session->channel) {
3128 SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
3129 }
3130
3131 /* Build and send the tech-specific cause information */
3132 /* size of the string making up the cause code is "SIP " number + " " + reason length */
3133 data_size += 4 + 4 + pj_strlen(&status.reason);
3134 cause_code = ast_alloca(data_size);
3135 memset(cause_code, 0, data_size);
3136
3138
3139 snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %d %.*s", status.code,
3140 (int) pj_strlen(&status.reason), pj_strbuf(&status.reason));
3141
3142 cause_code->ast_cause = ast_sip_hangup_sip2cause(status.code);
3143 ast_queue_control_data(session->channel, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
3144 ast_channel_hangupcause_hash_set(session->channel, cause_code, data_size);
3145
3147}
#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
1737 /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
1738 * RTP engine would provide a way to externally write/schedule RTCP
1739 * packets */
1740 struct ast_frame fr;
1742 fr.subclass.integer = AST_CONTROL_VIDUPDATE;
1743 res = ast_rtp_instance_write(media->rtp, &fr);
1744 } else {
1745 ao2_ref(channel->session, +1);
1747 ao2_cleanup(channel->session);
1748 }
1749 }
1750 ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Success");
1751 } else {
1752 ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Failure");
1753 res = -1;
1754 }
1755 }
1756 /* XXX If there were no video streams, then this should set
1757 * res to -1
1758 */
1759 break;
1761 ao2_ref(channel->session, +1);
1763 ao2_cleanup(channel->session);
1764 }
1765 break;
1767 break;
1769 res = -1;
1770 break;
1772 ast_assert(datalen == sizeof(int));
1773 if (*(int *) data) {
1774 /*
1775 * Masquerade is beginning:
1776 * Wait for session serializer to get suspended.
1777 */
1778 ast_channel_unlock(ast);
1780 ast_channel_lock(ast);
1781 } else {
1782 /*
1783 * Masquerade is complete:
1784 * Unsuspend the session serializer.
1785 */
1787 }
1788 break;
1789 case AST_CONTROL_HOLD:
1791 device_buf_size = strlen(ast_channel_name(ast)) + 1;
1792 device_buf = alloca(device_buf_size);
1793 ast_channel_get_device_name(ast, device_buf, device_buf_size);
1795 if (!channel->session->moh_passthrough) {
1796 ast_moh_start(ast, data, NULL);
1797 } else {
1799 ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n",
1801 ao2_ref(channel->session, -1);
1802 }
1803 }
1804 break;
1805 case AST_CONTROL_UNHOLD:
1807 device_buf_size = strlen(ast_channel_name(ast)) + 1;
1808 device_buf = alloca(device_buf_size);
1809 ast_channel_get_device_name(ast, device_buf, device_buf_size);
1811 if (!channel->session->moh_passthrough) {
1812 ast_moh_stop(ast);
1813 } else {
1815 ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n",
1817 ao2_ref(channel->session, -1);
1818 }
1819 }
1820 break;
1822 break;
1824 break;
1826 if (ast_channel_state(ast) != AST_STATE_UP) {
1827 response_code = 181;
1828 } else {
1829 res = -1;
1830 }
1831 break;
1833 res = 0;
1834
1835 if (channel->session->t38state == T38_PEER_REINVITE) {
1836 const struct ast_control_t38_parameters *parameters = data;
1837
1838 if (parameters->request_response == AST_T38_REQUEST_PARMS) {
1840 }
1841 }
1842
1843 break;
1845 topology = data;
1846 ast_trace(-1, "%s: New topology: %s\n", ast_channel_name(ast),
1847 ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
1848 res = handle_topology_request_change(channel->session, topology);
1849 break;
1851 break;
1853 break;
1854 case -1:
1855 res = -1;
1856 break;
1857 default:
1858 ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
1859 res = -1;
1860 break;
1861 }
1862
1863 if (response_code) {
1864 struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
1865
1866 if (!ind_data) {
1867 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc indicate data\n", ast_channel_name(ast));
1868 }
1869
1870 if (ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1871 ast_log(LOG_ERROR, "%s: Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1873 ao2_cleanup(ind_data);
1874 res = -1;
1875 }
1876 }
1877
1878 SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_channel_name(ast));
1879}
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:1013
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_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, and update_connected_line_information().

◆ 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:768
void ast_stream_topology_free(struct ast_stream_topology *topology)
Unreference and destroy a stream topology.
Definition: stream.c:746
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:933
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:670
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:914

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 2739 of file chan_pjsip.c.

2740{
2741 struct ast_stream_topology *topology;
2742 struct ast_channel *chan;
2743
2745 if (!topology) {
2746 return NULL;
2747 }
2748
2749 chan = chan_pjsip_request_with_stream_topology(type, topology, assignedids, requestor, data, cause);
2750
2751 ast_stream_topology_free(topology);
2752
2753 return chan;
2754}
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:2712
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:851
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 2712 of file chan_pjsip.c.

2713{
2714 struct request_data req_data;
2716 SCOPE_ENTER(1, "%s Topology: %s\n", data,
2718
2719 req_data.topology = topology;
2720 req_data.dest = data;
2721 /* Default failure value in case ast_sip_push_task_wait_servant() itself fails. */
2722 req_data.cause = AST_CAUSE_FAILURE;
2723
2724 if (ast_sip_push_task_wait_servant(NULL, request, &req_data)) {
2725 *cause = req_data.cause;
2726 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't push task\n");
2727 }
2728
2729 session = req_data.session;
2730
2731 if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) {
2732 /* Session needs to be terminated prematurely */
2733 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
2734 }
2735
2736 SCOPE_EXIT_RTN_VALUE(session->channel, "Channel: %s\n", ast_channel_name(session->channel));
2737}
#define AST_CAUSE_FAILURE
Definition: causes.h:150
static int request(void *obj)
Definition: chan_pjsip.c:2604
@ 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:2599

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 2882 of file chan_pjsip.c.

2883{
2884 struct ast_msg_data *msg;
2885 int rc;
2886 struct ast_msg_data_attribute attrs[] =
2887 {
2888 {
2890 .value = (char *)text,
2891 }
2892 };
2893
2895 if (!msg) {
2896 return -1;
2897 }
2898 rc = chan_pjsip_sendtext_data(ast, msg);
2899 ast_free(msg);
2900
2901 return rc;
2902}
char * text
Definition: app_queue.c:1668
#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:2860
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 2860 of file chan_pjsip.c.

2861{
2862 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2863 struct sendtext_data *data = sendtext_data_create(ast, msg);
2864
2865 ast_debug(1, "Sending MESSAGE from '%s' to '%s:%s': %s\n",
2868 ast_channel_name(ast),
2870
2871 if (!data) {
2872 return -1;
2873 }
2874
2875 if (ast_sip_push_task(channel->session->serializer, sendtext, data)) {
2876 ao2_ref(data, -1);
2877 return -1;
2878 }
2879 return 0;
2880}
static int sendtext(void *obj)
Definition: chan_pjsip.c:2789
static struct sendtext_data * sendtext_data_create(struct ast_channel *chan, struct ast_msg_data *msg)
Definition: chan_pjsip.c:2768
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:2758

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 2904 of file chan_pjsip.c.

2905{
2906 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
2908
2909 if (session->endpoint->media.direct_media.glare_mitigation ==
2911 SCOPE_EXIT_RTN("Direct media no glare mitigation\n");
2912 }
2913
2915 "direct_media_glare_mitigation");
2916
2917 if (!datastore) {
2918 SCOPE_EXIT_RTN("Couldn't create datastore\n");
2919 }
2920
2923}
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:629

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 2926 of file chan_pjsip.c.

2927{
2929
2930 if (!session->channel) {
2931 SCOPE_EXIT_RTN("No channel\n");
2932 }
2933
2934
2935 if (session->active_media_state &&
2936 session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
2937 struct ast_sip_session_media *media =
2938 session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
2939 if (media->rtp) {
2941 }
2942 }
2943
2945
2946 ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
2947 if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
2948 int cause = ast_sip_hangup_sip2cause(session->inv_session->cause);
2949
2950 ast_queue_hangup_with_cause(session->channel, cause);
2951 } else {
2952 ast_queue_hangup(session->channel);
2953 }
2954
2956}
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 2167 of file chan_pjsip.c.

2168{
2169 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
2170 struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
2171
2172 if (!trnf_data) {
2173 return -1;
2174 }
2175
2176 if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
2177 ast_log(LOG_WARNING, "Error requesting transfer\n");
2178 ao2_cleanup(trnf_data);
2179 return -1;
2180 }
2181
2182 return 0;
2183}
static int transfer(void *data)
Definition: chan_pjsip.c:2132
static struct transfer_data * transfer_data_alloc(struct ast_sip_session *session, const char *target)
Definition: chan_pjsip.c:1894

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 2512 of file chan_pjsip.c.

2513{
2514 session->channel = NULL;
2517}

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:637
@ AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING
Definition: res_pjsip.h:633
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 2519 of file chan_pjsip.c.

2520{
2521 struct hangup_data *h_data = data;
2522 struct ast_channel *ast = h_data->chan;
2523 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
2524 SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
2525
2526 /*
2527 * Before cleaning we have to ensure that channel or its session is not NULL
2528 * we have seen rare case when taskprocessor calls hangup but channel is NULL
2529 * due to SIP session timeout and answer happening at the same time
2530 */
2531 if (channel) {
2532 struct ast_sip_session *session = channel->session;
2533 if (session) {
2534 int cause = h_data->cause;
2535
2536 if (channel->session->active_media_state &&
2537 channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
2538 struct ast_sip_session_media *media =
2540 if (media->rtp) {
2542 }
2543 }
2544
2545 /*
2546 * It's possible that session_terminate might cause the session to be destroyed
2547 * immediately so we need to keep a reference to it so we can NULL session->channel
2548 * afterwards.
2549 */
2553 }
2554 ao2_cleanup(channel);
2555 }
2556 ao2_cleanup(h_data);
2558}
struct ast_channel * chan
Definition: chan_pjsip.c:2487

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 2438 of file chan_pjsip.c.

2439{
2440 switch (cause) {
2441 case AST_CAUSE_UNALLOCATED: /* 1 */
2442 case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
2443 case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
2444 return 404;
2445 case AST_CAUSE_CONGESTION: /* 34 */
2446 case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
2447 return 503;
2448 case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
2449 return 408;
2450 case AST_CAUSE_NO_ANSWER: /* 19 */
2451 case AST_CAUSE_UNREGISTERED: /* 20 */
2452 return 480;
2453 case AST_CAUSE_CALL_REJECTED: /* 21 */
2454 return 403;
2455 case AST_CAUSE_NUMBER_CHANGED: /* 22 */
2456 return 410;
2457 case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
2458 return 480;
2460 return 484;
2462 return 486;
2463 case AST_CAUSE_FAILURE:
2464 return 500;
2465 case AST_CAUSE_FACILITY_REJECTED: /* 29 */
2466 return 501;
2468 return 503;
2470 return 502;
2471 case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
2472 return 488;
2473 case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
2474 return 500;
2476 default:
2477 ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
2478 return 0;
2479 }
2480
2481 /* Never reached */
2482 return 0;
2483}
#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 2497 of file chan_pjsip.c.

2498{
2499 struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
2500
2501 if (!h_data) {
2502 return NULL;
2503 }
2504
2505 h_data->cause = cause;
2506 h_data->chan = ast_channel_ref(chan);
2507
2508 return h_data;
2509}
#define ao2_alloc(data_size, destructor_fn)
Definition: astobj2.h:409
static void hangup_data_destroy(void *obj)
Definition: chan_pjsip.c:2490

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 2490 of file chan_pjsip.c.

2491{
2492 struct hangup_data *h_data = obj;
2493
2494 h_data->chan = ast_channel_unref(h_data->chan);
2495}

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 2241 of file chan_pjsip.c.

2242{
2243 struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
2244 if (!dtmf_data) {
2245 return NULL;
2246 }
2247 ao2_ref(session, +1);
2248 dtmf_data->session = session;
2249 dtmf_data->digit = digit;
2250 dtmf_data->duration = duration;
2251 return dtmf_data;
2252}
static void info_dtmf_data_destroy(void *obj)
Definition: chan_pjsip.c:2235
struct ast_sip_session * session
Definition: chan_pjsip.c:2230

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 2235 of file chan_pjsip.c.

2236{
2237 struct info_dtmf_data *dtmf_data = obj;
2238 ao2_ref(dtmf_data->session, -1);
2239}

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:791
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 3289 of file chan_pjsip.c.

3290{
3291 struct ao2_container *endpoints;
3292
3295 }
3296
3298
3300
3302 ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
3303 goto end;
3304 }
3305
3307 ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
3308 goto end;
3309 }
3310
3312 ast_log(LOG_ERROR, "Unable to register PJSIP_PARSE_URI dialplan function\n");
3313 goto end;
3314 }
3315
3317 ast_log(LOG_ERROR, "Unable to register PJSIP_PARSE_URI_FROM dialplan function\n");
3318 goto end;
3319 }
3320
3322 ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
3323 goto end;
3324 }
3325
3327 ast_log(LOG_WARNING, "Unable to register PJSIP_DTMF_MODE dialplan function\n");
3328 goto end;
3329 }
3330
3332 ast_log(LOG_WARNING, "Unable to register PJSIP_MOH_PASSTHROUGH dialplan function\n");
3333 goto end;
3334 }
3335
3337 ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n");
3338 goto end;
3339 }
3340
3342 ast_log(LOG_WARNING, "Unable to register PJSIPHangup dialplan application\n");
3343 goto end;
3344 }
3346
3347
3349
3352
3356 ast_log(LOG_ERROR, "Unable to create held channels container\n");
3357 goto end;
3358 }
3359
3364
3366 ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
3367 goto end;
3368 }
3369
3370 /* since endpoints are loaded before the channel driver their device
3371 states get set to 'invalid', so they need to be updated */
3372 if ((endpoints = ast_sip_get_endpoints())) {
3374 ao2_ref(endpoints, -1);
3375 }
3376
3377 return 0;
3378
3379end:
3398
3401
3403}
@ 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:3277
static int update_devstate(void *obj, void *arg, int flags)
Definition: chan_pjsip.c:3232
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:3239
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:3113
static struct ast_custom_function chan_pjsip_parse_uri_from_function
Definition: chan_pjsip.c:3249
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:1951
static const char channel_type[]
Definition: chan_pjsip.c:78
static struct ast_custom_function moh_passthrough_function
Definition: chan_pjsip.c:3266
static struct ast_custom_function media_offer_function
Definition: chan_pjsip.c:3254
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:3244
static struct ast_custom_function session_refresh_function
Definition: chan_pjsip.c:3272
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:3073
static struct ast_custom_function dtmf_mode_function
Definition: chan_pjsip.c:3260
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:7608
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 3079 of file chan_pjsip.c.

3080{
3081 int res;
3083
3084 /* Check for a to-tag to determine if this is a reinvite */
3085 if (rdata->msg_info.to->tag.slen) {
3086 /* We don't care about reinvites */
3087 SCOPE_EXIT_RTN_VALUE(0, "Reinvite\n");
3088 }
3089
3090 res = ast_pbx_start(session->channel);
3091
3092 switch (res) {
3093 case AST_PBX_FAILED:
3094 ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
3096 ast_hangup(session->channel);
3097 break;
3098 case AST_PBX_CALL_LIMIT:
3099 ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
3101 ast_hangup(session->channel);
3102 break;
3103 case AST_PBX_SUCCESS:
3104 default:
3105 break;
3106 }
3107
3108 ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
3109
3110 SCOPE_EXIT_RTN_VALUE((res == AST_PBX_SUCCESS) ? 0 : -1, "RC: %d\n", res);
3111}
@ 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:622
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 2604 of file chan_pjsip.c.

2605{
2606 struct request_data *req_data = obj;
2607 struct ast_sip_session *session = NULL;
2608 char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
2609 struct ast_sip_endpoint *endpoint;
2610
2612 AST_APP_ARG(endpoint);
2613 AST_APP_ARG(aor);
2614 );
2615 SCOPE_ENTER(1, "%s\n",tmp);
2616
2617 if (ast_strlen_zero(tmp)) {
2618 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
2620 SCOPE_EXIT_RTN_VALUE(-1, "Empty destination\n");
2621 }
2622
2624
2626 /* If a request user has been specified extract it from the endpoint name portion */
2627 if ((endpoint_name = strchr(args.endpoint, '@'))) {
2628 request_user = args.endpoint;
2629 *endpoint_name++ = '\0';
2630 } else {
2631 endpoint_name = args.endpoint;
2632 }
2633
2634 if (ast_strlen_zero(endpoint_name)) {
2635 if (request_user) {
2636 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@<endpoint-name>\n",
2637 request_user);
2638 } else {
2639 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
2640 }
2642 SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
2643 }
2644 endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
2645 endpoint_name);
2646 if (!endpoint) {
2647 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
2649 SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
2650 }
2651 } else {
2652 /* First try to find an exact endpoint match, for single (user) or multi-domain (user@domain) */
2653 endpoint_name = args.endpoint;
2654 if (ast_strlen_zero(endpoint_name)) {
2655 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
2657 SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
2658 }
2659 endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
2660 endpoint_name);
2661 if (!endpoint) {
2662 /* It seems it's not a multi-domain endpoint or single endpoint exact match,
2663 * it's possible that it's a SIP trunk with a specified user (user@trunkname),
2664 * so extract the user before @ sign.
2665 */
2666 endpoint_name = strchr(args.endpoint, '@');
2667 if (!endpoint_name) {
2668 /*
2669 * Couldn't find an '@' so it had to be an endpoint
2670 * name that doesn't exist.
2671 */
2672 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n",
2673 args.endpoint);
2675 SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
2676 }
2677 request_user = args.endpoint;
2678 *endpoint_name++ = '\0';
2679
2680 if (ast_strlen_zero(endpoint_name)) {
2681 ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@<endpoint-name>\n",
2682 request_user);
2684 SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
2685 }
2686
2687 endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
2688 endpoint_name);
2689 if (!endpoint) {
2690 ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
2692 SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
2693 }
2694 }
2695 }
2696
2697 session = ast_sip_session_create_outgoing(endpoint, NULL, args.aor, request_user,
2698 req_data->topology);
2699 ao2_ref(endpoint, -1);
2700 if (!session) {
2701 ast_log(LOG_ERROR, "Failed to create outgoing session to endpoint '%s'\n", endpoint_name);
2703 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create session\n");
2704 }
2705
2706 req_data->session = session;
2707
2709}
#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:2598
const char * dest
Definition: chan_pjsip.c:2600
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:620
@ AST_SIP_SESSION_REFRESH_METHOD_UPDATE
Definition: res_pjsip.h:624
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 2789 of file chan_pjsip.c.

2790{
2791 struct sendtext_data *data = obj;
2792 pjsip_tx_data *tdata;
2793 const char *body_text = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_BODY);
2794 const char *content_type = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_CONTENT_TYPE);
2795 char *sep;
2796 struct ast_sip_body body = {
2797 .type = "text",
2798 .subtype = "plain",
2799 .body_text = body_text,
2800 };
2801
2802 if (!ast_strlen_zero(content_type)) {
2803 sep = strchr(content_type, '/');
2804 if (sep) {
2805 *sep = '\0';
2806 body.type = content_type;
2807 body.subtype = ++sep;
2808 }
2809 }
2810
2811 if (data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
2812 ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
2813 data->session->inv_session->cause,
2814 pjsip_get_status_text(data->session->inv_session->cause)->ptr);
2815 } else {
2816 pjsip_from_hdr *hdr;
2817 pjsip_name_addr *name_addr;
2818 const char *from = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_FROM);
2819 const char *to = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_TO);
2820 int invalidate_tdata = 0;
2821
2822 ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, NULL, &tdata);
2823 ast_sip_add_body(tdata, &body);
2824
2825 /*
2826 * If we have a 'from' in the msg, set the display name in the From
2827 * header to it.
2828 */
2829 if (!ast_strlen_zero(from)) {
2830 hdr = PJSIP_MSG_FROM_HDR(tdata->msg);
2831 name_addr = (pjsip_name_addr *) hdr->uri;
2832 pj_strdup2(tdata->pool, &name_addr->display, from);
2833 invalidate_tdata = 1;
2834 }
2835
2836 /*
2837 * If we have a 'to' in the msg, set the display name in the To
2838 * header to it.
2839 */
2840 if (!ast_strlen_zero(to)) {
2841 hdr = PJSIP_MSG_TO_HDR(tdata->msg);
2842 name_addr = (pjsip_name_addr *) hdr->uri;
2843 pj_strdup2(tdata->pool, &name_addr->display, to);
2844 invalidate_tdata = 1;
2845 }
2846
2847 if (invalidate_tdata) {
2848 pjsip_tx_data_invalidate_msg(tdata);
2849 }
2850
2851 ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint, NULL, NULL);
2852 }
2853
2854 ao2_cleanup(data);
2855
2856 return 0;
2857}
@ AST_MSG_DATA_ATTR_CONTENT_TYPE
Definition: message.h:457
int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint, void *token, void(*callback)(void *token, pjsip_event *e))
General purpose method for sending a SIP request.
Definition: res_pjsip.c:1979
int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body)
Add a body to an outbound SIP message.
Definition: res_pjsip.c:2052
int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint, const char *uri, struct ast_sip_contact *contact, pjsip_tx_data **tdata)
General purpose method for creating a SIP request.
Definition: res_pjsip.c:1435
SIP body description.
Definition: res_pjsip.h:2324
const char * type
Definition: res_pjsip.h:2326
const char * body_text
Definition: res_pjsip.h:2330
const char * subtype
Definition: res_pjsip.h:2328
struct ast_sip_session * session
Definition: chan_pjsip.c:2757

References ao2_cleanup, ast_log, AST_MSG_DATA_ATTR_BODY, AST_MSG_DATA_ATTR_CONTENT_TYPE, AST_MSG_DATA_ATTR_FROM, AST_MSG_DATA_ATTR_TO, ast_msg_data_get_attribute(), ast_sip_add_body(), ast_sip_create_request(), ast_sip_send_request(), ast_strlen_zero(), ast_sip_body::body_text, ast_sip_session::endpoint, ast_sip_session::inv_session, LOG_ERROR, sendtext_data::msg, NULL, sendtext_data::session, ast_sip_body::subtype, and ast_sip_body::type.

Referenced by chan_pjsip_sendtext_data().

◆ sendtext_data_create()

static struct sendtext_data * sendtext_data_create ( struct ast_channel chan,
struct ast_msg_data msg 
)
static

Definition at line 2768 of file chan_pjsip.c.

2770{
2771 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
2772 struct sendtext_data *data = ao2_alloc(sizeof(*data), sendtext_data_destroy);
2773
2774 if (!data) {
2775 return NULL;
2776 }
2777
2778 data->msg = ast_msg_data_dup(msg);
2779 if (!data->msg) {
2780 ao2_cleanup(data);
2781 return NULL;
2782 }
2783 data->session = channel->session;
2784 ao2_ref(data->session, +1);
2785
2786 return data;
2787}
static void sendtext_data_destroy(void *obj)
Definition: chan_pjsip.c:2761
struct ast_msg_data * ast_msg_data_dup(struct ast_msg_data *msg)
Clone an ast_msg_data structure.

References ao2_alloc, ao2_cleanup, ao2_ref, ast_channel_tech_pvt(), ast_msg_data_dup(), sendtext_data::msg, NULL, sendtext_data_destroy(), sendtext_data::session, and ast_sip_channel_pvt::session.

Referenced by chan_pjsip_sendtext_data().

◆ sendtext_data_destroy()

static void sendtext_data_destroy ( void *  obj)
static

Definition at line 2761 of file chan_pjsip.c.

2762{
2763 struct sendtext_data *data = obj;
2764 ao2_cleanup(data->session);
2765 ast_free(data->msg);
2766}

References ao2_cleanup, ast_free, sendtext_data::msg, and sendtext_data::session.

Referenced by sendtext_data_create().

◆ set_channel_on_rtp_instance()

static void set_channel_on_rtp_instance ( const struct ast_sip_session session,
const char *  channel_id 
)
static

Definition at line 494 of file chan_pjsip.c.

496{
497 int i;
498
499 for (i = 0; i < AST_VECTOR_SIZE(&session->active_media_state->sessions); ++i) {
500 struct ast_sip_session_media *session_media;
501
502 session_media = AST_VECTOR_GET(&session->active_media_state->sessions, i);
503 if (!session_media || !session_media->rtp) {
504 continue;
505 }
506
507 ast_rtp_instance_set_channel_id(session_media->rtp, channel_id);
508 }
509}
void ast_rtp_instance_set_channel_id(struct ast_rtp_instance *instance, const char *uniqueid)
Set the channel that owns this RTP instance.
Definition: rtp_engine.c:575

References ast_rtp_instance_set_channel_id(), AST_VECTOR_GET, AST_VECTOR_SIZE, ast_sip_session_media::rtp, and session.

Referenced by call(), chan_pjsip_fixup(), chan_pjsip_new(), and clear_session_and_channel().

◆ set_sipdomain_variable()

static void set_sipdomain_variable ( struct ast_sip_session session)
static

Definition at line 2958 of file chan_pjsip.c.

2959{
2960 const pj_str_t *host = ast_sip_pjsip_uri_get_hostname(session->request_uri);
2961 size_t size = pj_strlen(host) + 1;
2962 char *domain = ast_alloca(size);
2963
2964 ast_copy_pj_str(domain, host, size);
2965
2966 pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
2967 return;
2968}
const pj_str_t * ast_sip_pjsip_uri_get_hostname(pjsip_uri *uri)
Get the host portion of the pjsip_uri.
Definition: res_pjsip.c:3496

References ast_alloca, ast_copy_pj_str(), ast_sip_pjsip_uri_get_hostname(), pbx_builtin_setvar_helper(), and session.

Referenced by chan_pjsip_incoming_request().

◆ topology_change_refresh_data_alloc()

static struct topology_change_refresh_data * topology_change_refresh_data_alloc ( struct ast_sip_session session,
const struct ast_stream_topology topology 
)
static

Definition at line 1524 of file chan_pjsip.c.

1526{
1528
1529 refresh_data = ast_calloc(1, sizeof(*refresh_data));
1530 if (!refresh_data) {
1531 return NULL;
1532 }
1533
1536 if (!refresh_data->media_state) {
1538 return NULL;
1539 }
1540 refresh_data->media_state->topology = ast_stream_topology_clone(topology);
1541 if (!refresh_data->media_state->topology) {
1543 return NULL;
1544 }
1545
1546 return refresh_data;
1547}
struct ast_sip_session_media_state * ast_sip_session_media_state_alloc(void)
Allocate a session media state structure.

References ao2_bump, ast_calloc, ast_sip_session_media_state_alloc(), ast_stream_topology_clone(), NULL, refresh_data::session, session, and topology_change_refresh_data_free().

Referenced by handle_topology_request_change().

◆ topology_change_refresh_data_free()

static void topology_change_refresh_data_free ( struct topology_change_refresh_data refresh_data)
static

Definition at line 1516 of file chan_pjsip.c.

1517{
1519
1522}
void ast_sip_session_media_state_free(struct ast_sip_session_media_state *media_state)
Free a session media state structure.

References ao2_cleanup, ast_free, ast_sip_session_media_state_free(), and refresh_data::session.

Referenced by handle_topology_request_change(), send_topology_change_refresh(), and topology_change_refresh_data_alloc().

◆ transfer()

static int transfer ( void *  data)
static

Definition at line 2132 of file chan_pjsip.c.

2133{
2134 struct transfer_data *trnf_data = data;
2135 struct ast_sip_endpoint *endpoint = NULL;
2136 struct ast_sip_contact *contact = NULL;
2137 const char *target = trnf_data->target;
2138
2139 if (trnf_data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
2140 ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
2141 trnf_data->session->inv_session->cause,
2142 pjsip_get_status_text(trnf_data->session->inv_session->cause)->ptr);
2143 } else {
2144 /* See if we have an endpoint; if so, use its contact */
2146 if (endpoint) {
2148 if (contact && !ast_strlen_zero(contact->uri)) {
2149 target = contact->uri;
2150 }
2151 }
2152
2153 if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
2154 transfer_redirect(trnf_data->session, target);
2155 } else {
2156 transfer_refer(trnf_data->session, target);
2157 }
2158 }
2159
2160 ao2_ref(trnf_data, -1);
2162 ao2_cleanup(contact);
2163 return 0;
2164}
static void transfer_redirect(struct ast_sip_session *session, const char *target)
Definition: chan_pjsip.c:1913
static void transfer_refer(struct ast_sip_session *session, const char *target)
Definition: chan_pjsip.c:2080
struct ast_sip_contact * ast_sip_location_retrieve_contact_from_aor_list(const char *aor_list)
Retrieve the first bound contact from a list of AORs.
Definition: location.c:304
Contact associated with an address of record.
Definition: res_pjsip.h:389
const ast_string_field uri
Definition: res_pjsip.h:411
struct ast_sip_endpoint * endpoint
Definition: res_pjsip.h:421
const ast_string_field aors
Definition: res_pjsip.h:987
struct ast_sip_session * session
Definition: chan_pjsip.c:1882

References ao2_cleanup, ao2_ref, ast_sip_endpoint::aors, ast_log, ast_sip_get_sorcery(), ast_sip_location_retrieve_contact_from_aor_list(), ast_sorcery_retrieve_by_id(), AST_STATE_RING, ast_strlen_zero(), ast_sip_session::channel, ast_sip_contact::endpoint, ast_sip_session::inv_session, LOG_ERROR, NULL, transfer_data::session, transfer_data::target, transfer_redirect(), transfer_refer(), and ast_sip_contact::uri.

Referenced by __send_command(), ast_transfer_protocol(), chan_pjsip_transfer(), iax2_send(), leave_voicemail(), send_packet(), and transfer_exec().

◆ transfer_data_alloc()

static struct transfer_data * transfer_data_alloc ( struct ast_sip_session session,
const char *  target 
)
static

Definition at line 1894 of file chan_pjsip.c.

1895{
1896 struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy);
1897
1898 if (!trnf_data) {
1899 return NULL;
1900 }
1901
1902 if (!(trnf_data->target = ast_strdup(target))) {
1903 ao2_ref(trnf_data, -1);
1904 return NULL;
1905 }
1906
1907 ao2_ref(session, +1);
1908 trnf_data->session = session;
1909
1910 return trnf_data;
1911}
static void transfer_data_destroy(void *obj)
Definition: chan_pjsip.c:1886

References ao2_alloc, ao2_ref, ast_strdup, NULL, transfer_data::session, session, transfer_data::target, and transfer_data_destroy().

Referenced by chan_pjsip_transfer().

◆ transfer_data_destroy()

static void transfer_data_destroy ( void *  obj)
static

Definition at line 1886 of file chan_pjsip.c.

1887{
1888 struct transfer_data *trnf_data = obj;
1889
1890 ast_free(trnf_data->target);
1891 ao2_cleanup(trnf_data->session);
1892}

References ao2_cleanup, ast_free, transfer_data::session, and transfer_data::target.

Referenced by transfer_data_alloc().

◆ transfer_redirect()

static void transfer_redirect ( struct ast_sip_session session,
const char *  target 
)
static

Definition at line 1913 of file chan_pjsip.c.

1914{
1915 pjsip_tx_data *packet;
1917 pjsip_contact_hdr *contact;
1918 pj_str_t tmp;
1919
1920 if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS
1921 || !packet) {
1922 ast_log(LOG_WARNING, "Failed to redirect PJSIP session for channel %s\n",
1923 ast_channel_name(session->channel));
1926
1927 return;
1928 }
1929
1930 if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
1931 contact = pjsip_contact_hdr_create(packet->pool);
1932 }
1933
1934 pj_strdup2_with_null(packet->pool, &tmp, target);
1935 if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
1936 ast_log(LOG_WARNING, "Failed to parse destination URI '%s' for channel %s\n",
1937 target, ast_channel_name(session->channel));
1940 pjsip_tx_data_dec_ref(packet);
1941
1942 return;
1943 }
1944 pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
1945
1948}
ast_control_transfer
@ AST_TRANSFER_FAILED
@ AST_TRANSFER_SUCCESS
@ AST_CONTROL_TRANSFER

References ast_channel_name(), AST_CONTROL_TRANSFER, ast_log, ast_queue_control_data(), ast_sip_session_send_response(), AST_TRANSFER_FAILED, AST_TRANSFER_SUCCESS, LOG_WARNING, NULL, session, transfer_data::target, and tmp().

Referenced by transfer().

◆ transfer_refer()

static void transfer_refer ( struct ast_sip_session session,
const char *  target 
)
static

Definition at line 2080 of file chan_pjsip.c.

2081{
2082 pjsip_evsub *sub;
2084 pj_str_t tmp;
2085 pjsip_tx_data *packet;
2086 const char *ref_by_val;
2087 char local_info[pj_strlen(&session->inv_session->dlg->local.info_str) + 1];
2088 struct pjsip_evsub_user xfer_cb;
2089 struct ast_channel *chan = session->channel;
2090
2091 pj_bzero(&xfer_cb, sizeof(xfer_cb));
2092 xfer_cb.on_evsub_state = &xfer_client_on_evsub_state;
2093
2094 if (pjsip_xfer_create_uac(session->inv_session->dlg, &xfer_cb, &sub) != PJ_SUCCESS) {
2097
2098 return;
2099 }
2100
2101 /* refer_callback_module requires a reference to chan
2102 * which will be released in xfer_client_on_evsub_state()
2103 * when the implicit REFER subscription terminates */
2104 pjsip_evsub_set_mod_data(sub, refer_callback_module.id, chan);
2105 ao2_ref(chan, +1);
2106
2107 if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
2108 goto failure;
2109 }
2110
2111 ref_by_val = pbx_builtin_getvar_helper(chan, "SIPREFERREDBYHDR");
2112 if (!ast_strlen_zero(ref_by_val)) {
2113 ast_sip_add_header(packet, "Referred-By", ref_by_val);
2114 } else {
2115 ast_copy_pj_str(local_info, &session->inv_session->dlg->local.info_str, sizeof(local_info));
2116 ast_sip_add_header(packet, "Referred-By", local_info);
2117 }
2118
2119 if (pjsip_xfer_send_request(sub, packet) == PJ_SUCCESS) {
2120 return;
2121 }
2122
2123failure:
2126 pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
2127 pjsip_evsub_terminate(sub, PJ_FALSE);
2128
2129 ao2_ref(chan, -1);
2130}
static void xfer_client_on_evsub_state(pjsip_evsub *sub, pjsip_event *event)
Callback function to report status of implicit REFER-NOTIFY subscription.
Definition: chan_pjsip.c:1963
const char * pbx_builtin_getvar_helper(struct ast_channel *chan, const char *name)
Return a pointer to the value of the corresponding channel variable.
struct stasis_forward * sub
Definition: res_corosync.c:240
int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value)
Add a header to an outbound SIP message.
Definition: res_pjsip.c:2008

References ao2_ref, AST_CONTROL_TRANSFER, ast_copy_pj_str(), ast_queue_control_data(), ast_sip_add_header(), ast_strlen_zero(), AST_TRANSFER_FAILED, AST_TRANSFER_SUCCESS, NULL, pbx_builtin_getvar_helper(), refer_callback_module, session, sub, tmp(), and xfer_client_on_evsub_state().

Referenced by transfer().

◆ transmit_info_dtmf()

static int transmit_info_dtmf ( void *  data)
static

Definition at line 2254 of file chan_pjsip.c.

2255{
2256 RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
2257
2258 struct ast_sip_session *session = dtmf_data->session;
2259 struct pjsip_tx_data *tdata;
2260
2261 RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
2262
2263 struct ast_sip_body body = {
2264 .type = "application",
2265 .subtype = "dtmf-relay",
2266 };
2267
2268 if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
2269 ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
2270 session->inv_session->cause,
2271 pjsip_get_status_text(session->inv_session->cause)->ptr);
2272 return -1;
2273 }
2274
2275 if (!(body_text = ast_str_create(32))) {
2276 ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
2277 return -1;
2278 }
2279 ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
2280
2282
2283 if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) {
2284 ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
2285 return -1;
2286 }
2287 if (ast_sip_add_body(tdata, &body)) {
2288 ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
2289 pjsip_tx_data_dec_ref(tdata);
2290 return -1;
2291 }
2293
2294 return 0;
2295}
void ast_free_ptr(void *ptr)
free() wrapper
Definition: astmm.c:1739
char * ast_str_buffer(const struct ast_str *buf)
Returns the string buffer within the ast_str buf.
Definition: strings.h:761
#define ast_str_create(init_len)
Create a malloc'ed dynamic length string.
Definition: strings.h:659
int ast_str_set(struct ast_str **buf, ssize_t max_len, const char *fmt,...)
Set a dynamic string using variable arguments.
Definition: strings.h:1113

References ao2_cleanup, ast_free_ptr(), ast_log, ast_sip_add_body(), ast_sip_create_request(), ast_sip_session_send_request(), ast_str_buffer(), ast_str_create, ast_str_set(), ast_sip_body::body_text, LOG_ERROR, NULL, RAII_VAR, session, and ast_sip_body::type.

Referenced by chan_pjsip_digit_end(), and chan_pjsip_indicate().

◆ transmit_info_with_vidupdate()

static int transmit_info_with_vidupdate ( void *  data)
static

Send SIP INFO with video update request.

Definition at line 1356 of file chan_pjsip.c.

1357{
1358 const char * xml =
1359 "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
1360 " <media_control>\r\n"
1361 " <vc_primitive>\r\n"
1362 " <to_encoder>\r\n"
1363 " <picture_fast_update/>\r\n"
1364 " </to_encoder>\r\n"
1365 " </vc_primitive>\r\n"
1366 " </media_control>\r\n";
1367
1368 const struct ast_sip_body body = {
1369 .type = "application",
1370 .subtype = "media_control+xml",
1371 .body_text = xml
1372 };
1373
1374 RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
1375 struct pjsip_tx_data *tdata;
1376
1377 if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
1378 ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
1379 session->inv_session->cause,
1380 pjsip_get_status_text(session->inv_session->cause)->ptr);
1381 return -1;
1382 }
1383
1384 if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) {
1385 ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
1386 return -1;
1387 }
1388 if (ast_sip_add_body(tdata, &body)) {
1389 ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
1390 return -1;
1391 }
1393
1394 return 0;
1395}

References ao2_cleanup, ast_log, ast_sip_add_body(), ast_sip_create_request(), ast_sip_session_send_request(), LOG_ERROR, NULL, RAII_VAR, session, and ast_sip_body::type.

Referenced by chan_pjsip_indicate().

◆ transport_info_destroy()

static void transport_info_destroy ( void *  obj)
static

Destructor function for transport_info_data.

Definition at line 261 of file chan_pjsip.c.

262{
263 struct transport_info_data *data = obj;
264 ast_free(data);
265}

References ast_free.

◆ uid_hold_hash_fn()

static int uid_hold_hash_fn ( const void *  obj,
const int  flags 
)
static

AO2 hash function for on hold UIDs

Definition at line 1071 of file chan_pjsip.c.

1072{
1073 const char *key = obj;
1074
1075 switch (flags & OBJ_SEARCH_MASK) {
1076 case OBJ_SEARCH_KEY:
1077 break;
1078 case OBJ_SEARCH_OBJECT:
1079 break;
1080 default:
1081 /* Hash can only work on something with a full key. */
1082 ast_assert(0);
1083 return 0;
1084 }
1085 return ast_str_hash(key);
1086}
@ OBJ_SEARCH_OBJECT
The arg parameter is an object of the same type.
Definition: astobj2.h:1087
@ OBJ_SEARCH_MASK
Search option field mask.
Definition: astobj2.h:1072
static force_inline int attribute_pure ast_str_hash(const char *str)
Compute a hash value on a string.
Definition: strings.h:1259

References ast_assert, ast_str_hash(), OBJ_SEARCH_KEY, OBJ_SEARCH_MASK, and OBJ_SEARCH_OBJECT.

Referenced by load_module().

◆ uid_hold_sort_fn()

static int uid_hold_sort_fn ( const void *  obj_left,
const void *  obj_right,
const int  flags 
)
static

AO2 sort function for on hold UIDs

Definition at line 1089 of file chan_pjsip.c.

1090{
1091 const char *left = obj_left;
1092 const char *right = obj_right;
1093 int cmp;
1094
1095 switch (flags & OBJ_SEARCH_MASK) {
1096 case OBJ_SEARCH_OBJECT:
1097 case OBJ_SEARCH_KEY:
1098 cmp = strcmp(left, right);
1099 break;
1101 cmp = strncmp(left, right, strlen(right));
1102 break;
1103 default:
1104 /* Sort can only work on something with a full or partial key. */
1105 ast_assert(0);
1106 cmp = 0;
1107 break;
1108 }
1109 return cmp;
1110}
@ OBJ_SEARCH_PARTIAL_KEY
The arg parameter is a partial search key similar to OBJ_SEARCH_KEY.
Definition: astobj2.h:1116

References ast_assert, OBJ_SEARCH_KEY, OBJ_SEARCH_MASK, OBJ_SEARCH_OBJECT, and OBJ_SEARCH_PARTIAL_KEY.

Referenced by load_module().

◆ unload_module()

static int unload_module ( void  )
static

Unload the PJSIP channel from Asterisk.

Definition at line 3406 of file chan_pjsip.c.

References ao2_cleanup, ao2_ref, app_pjsip_hangup, ast_channel_unregister(), ast_custom_function_unregister(), ast_manager_unregister(), ast_rtp_glue_unregister(), 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, dtmf_mode_function, media_offer_function, moh_passthrough_function, NULL, pbx_start_supplement, pjsip_channel_cli_unregister(), pjsip_uids_onhold, refer_callback_module, and session_refresh_function.

◆ update_connected_line_information()

static int update_connected_line_information ( void *  data)
static

Update connected line information.

Definition at line 1434 of file chan_pjsip.c.

1435{
1436 struct ast_sip_session *session = data;
1437
1438 if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
1439 ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
1440 session->inv_session->cause,
1441 pjsip_get_status_text(session->inv_session->cause)->ptr);
1442 ao2_ref(session, -1);
1443 return -1;
1444 }
1445
1446 if (ast_channel_state(session->channel) == AST_STATE_UP
1447 || session->inv_session->role == PJSIP_ROLE_UAC) {
1450 int generate_new_sdp;
1451
1452 method = session->endpoint->id.refresh_method;
1453 if (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE) {
1455 }
1456
1457 /* Only the INVITE method actually needs SDP, UPDATE can do without */
1458 generate_new_sdp = (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE);
1459
1460 ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp, NULL);
1461 }
1462 } else if (session->endpoint->id.rpid_immediate
1463 && session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED
1465 int response_code = 0;
1466
1467 if (ast_channel_state(session->channel) == AST_STATE_RING) {
1468 response_code = !session->endpoint->inband_progress ? 180 : 183;
1469 } else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
1470 response_code = 183;
1471 }
1472
1473 if (response_code) {
1474 struct pjsip_tx_data *packet = NULL;
1475
1476 if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
1478 }
1479 }
1480 }
1481
1482 ao2_ref(session, -1);
1483 return 0;
1484}
static int is_colp_update_allowed(struct ast_sip_session *session)
Definition: chan_pjsip.c:1407

References ao2_ref, ast_log, ast_sip_session_refresh(), AST_SIP_SESSION_REFRESH_METHOD_INVITE, AST_SIP_SESSION_REFRESH_METHOD_UPDATE, ast_sip_session_send_response(), AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, is_colp_update_allowed(), LOG_ERROR, method, NULL, and session.

Referenced by chan_pjsip_indicate().

◆ update_devstate()

static int update_devstate ( void *  obj,
void *  arg,
int  flags 
)
static

◆ update_initial_connected_line()

static void update_initial_connected_line ( struct ast_sip_session session)
static

Definition at line 2369 of file chan_pjsip.c.

2370{
2372
2373 /*
2374 * Use the channel CALLERID() as the initial connected line data.
2375 * The core or a predial handler may have supplied missing values
2376 * from the session->endpoint->id.self about who we are calling.
2377 */
2378 ast_channel_lock(session->channel);
2380 ast_channel_unlock(session->channel);
2381
2382 /* Supply initial connected line information if available. */
2383 if (!session->id.number.valid && !session->id.name.valid) {
2384 return;
2385 }
2386
2388 connected.id = session->id;
2390
2392}
@ AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER
Definition: callerid.h:554
void ast_party_connected_line_init(struct ast_party_connected_line *init)
Initialize the given connected line structure.
Definition: channel.c:2041
void ast_channel_queue_connected_line_update(struct ast_channel *chan, const struct ast_party_connected_line *connected, const struct ast_set_party_connected_line *update)
Queue a connected line update frame on a channel.
Definition: channel.c:9128
char connected
Definition: eagi_proxy.c:82
struct ast_party_id id
Caller party ID.
Definition: channel.h:422
Connected Line/Party information.
Definition: channel.h:458

References ast_channel_caller(), ast_channel_lock, ast_channel_queue_connected_line_update(), ast_channel_unlock, AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER, ast_party_connected_line_init(), ast_party_id_copy(), connected, ast_party_caller::id, NULL, and session.

Referenced by call().

◆ xfer_client_on_evsub_state()

static void xfer_client_on_evsub_state ( pjsip_evsub *  sub,
pjsip_event *  event 
)
static

Callback function to report status of implicit REFER-NOTIFY subscription.

This function will be called on any state change in the REFER-NOTIFY subscription. Its primary purpose is to report SUCCESS/FAILURE of a transfer initiated via transfer_refer as well as to terminate the subscription, if necessary.

Definition at line 1963 of file chan_pjsip.c.

1964{
1965 struct ast_channel *chan;
1967 int res = 0;
1968
1969 if (!event) {
1970 return;
1971 }
1972
1973 chan = pjsip_evsub_get_mod_data(sub, refer_callback_module.id);
1974 if (!chan) {
1975 return;
1976 }
1977
1978 if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
1979 /* Check if subscription is suppressed and terminate and send completion code, if so. */
1980 pjsip_rx_data *rdata;
1981 pjsip_generic_string_hdr *refer_sub;
1982 const pj_str_t REFER_SUB = { "Refer-Sub", 9 };
1983
1984 ast_debug(3, "Transfer accepted on channel %s\n", ast_channel_name(chan));
1985
1986 /* Check if response message */
1987 if (event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
1988 rdata = event->body.tsx_state.src.rdata;
1989
1990 /* Find Refer-Sub header */
1991 refer_sub = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &REFER_SUB, NULL);
1992
1993 /* Check if subscription is suppressed. If it is, the far end will not terminate it,
1994 * and the subscription will remain active until it times out. Terminating it here
1995 * eliminates the unnecessary timeout.
1996 */
1997 if (refer_sub && !pj_stricmp2(&refer_sub->hvalue, "false")) {
1998 /* Since no subscription is desired, assume that call has been transferred successfully. */
1999 /* Channel reference will be released at end of function */
2000 /* Terminate subscription. */
2001 pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
2002 pjsip_evsub_terminate(sub, PJ_TRUE);
2003 res = -1;
2004 }
2005 }
2006 } else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
2007 pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
2008 /* Check for NOTIFY complete or error. */
2009 pjsip_msg *msg;
2010 pjsip_msg_body *body;
2011 pjsip_status_line status_line = { .code = 0 };
2012 pj_bool_t is_last;
2013 pj_status_t status;
2014
2015 if (event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
2016 pjsip_rx_data *rdata;
2017
2018 rdata = event->body.tsx_state.src.rdata;
2019 msg = rdata->msg_info.msg;
2020
2021 if (msg->type == PJSIP_REQUEST_MSG) {
2022 if (!pjsip_method_cmp(&msg->line.req.method, pjsip_get_notify_method())) {
2023 body = msg->body;
2024 if (body && !pj_stricmp2(&body->content_type.type, "message")
2025 && !pj_stricmp2(&body->content_type.subtype, "sipfrag")) {
2026 pjsip_parse_status_line((char *)body->data, body->len, &status_line);
2027 }
2028 }
2029 } else {
2030 status_line.code = msg->line.status.code;
2031 status_line.reason = msg->line.status.reason;
2032 }
2033 } else {
2034 status_line.code = 500;
2035 status_line.reason = *pjsip_get_status_text(500);
2036 }
2037
2038 is_last = (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED);
2039 /* If the status code is >= 200, the subscription is finished. */
2040 if (status_line.code >= 200 || is_last) {
2041 res = -1;
2042
2043 /* If the subscription has terminated, return AST_TRANSFER_SUCCESS for 2XX.
2044 * Return AST_TRANSFER_FAILED for any code < 200.
2045 * Otherwise, return the status code.
2046 * The subscription should not terminate for any code < 200,
2047 * but if it does, that constitutes a failure. */
2048 if (status_line.code < 200) {
2050 } else if (status_line.code >= 300) {
2051 message = status_line.code;
2052 }
2053
2054 /* If subscription not terminated and subscription is finished (status code >= 200)
2055 * terminate it */
2056 if (!is_last) {
2057 pjsip_tx_data *tdata;
2058
2059 status = pjsip_evsub_initiate(sub, pjsip_get_subscribe_method(), 0, &tdata);
2060 if (status == PJ_SUCCESS) {
2061 pjsip_evsub_send_request(sub, tdata);
2062 }
2063 }
2064 /* Finished. Remove session from subscription */
2065 pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
2066 ast_debug(3, "Transfer channel %s completed: %d %.*s (%s)\n",
2067 ast_channel_name(chan),
2068 status_line.code,
2069 (int)status_line.reason.slen, status_line.reason.ptr,
2070 (message == AST_TRANSFER_SUCCESS) ? "Success" : "Failure");
2071 }
2072 }
2073
2074 if (res) {
2076 ao2_ref(chan, -1);
2077 }
2078}
Definition: astman.c:222

References ao2_ref, ast_channel_name(), AST_CONTROL_TRANSFER, ast_debug, ast_queue_control_data(), AST_TRANSFER_FAILED, AST_TRANSFER_SUCCESS, NULL, refer_callback_module, status, and sub.

Referenced by transfer_refer().

Variable Documentation

◆ __mod_info

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

Definition at line 3445 of file chan_pjsip.c.

◆ app_pjsip_hangup

char* app_pjsip_hangup = "PJSIPHangup"
static

Definition at line 3277 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info
static

Definition at line 3445 of file chan_pjsip.c.

◆ call_pickup_supplement

struct ast_sip_session_supplement call_pickup_supplement
static
Initial value:
= {
.method = "INVITE",
.incoming_request = call_pickup_incoming_request,
}
static int call_pickup_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
Definition: chan_pjsip.c:3029
@ AST_SIP_SUPPLEMENT_PRIORITY_LAST
Definition: res_pjsip.h:3187

Definition at line 3073 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_idx

unsigned int chan_idx
static

Definition at line 80 of file chan_pjsip.c.

Referenced by chan_pjsip_new().

◆ chan_pjsip_ack_supplement

struct ast_sip_session_supplement chan_pjsip_ack_supplement
static
Initial value:
= {
.method = "ACK",
.incoming_request = chan_pjsip_incoming_ack,
}
static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
Definition: chan_pjsip.c:3207
@ AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL
Definition: res_pjsip.h:3185

Definition at line 164 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_pjsip_dial_contacts_function

struct ast_custom_function chan_pjsip_dial_contacts_function
static
Initial value:
= {
.name = "PJSIP_DIAL_CONTACTS",
}
int pjsip_acf_dial_contacts_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
PJSIP_DIAL_CONTACTS function read callback.

Definition at line 3239 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_pjsip_parse_uri_from_function

struct ast_custom_function chan_pjsip_parse_uri_from_function
static
Initial value:
= {
.name = "PJSIP_PARSE_URI_FROM",
}
int pjsip_acf_parse_uri_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t buflen)
PJSIP_PARSE_URI function read callback.

Definition at line 3249 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_pjsip_parse_uri_function

struct ast_custom_function chan_pjsip_parse_uri_function
static
Initial value:
= {
.name = "PJSIP_PARSE_URI",
}

Definition at line 3244 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_pjsip_prack_supplement

struct ast_sip_session_supplement chan_pjsip_prack_supplement
static
Initial value:
= {
.method = "PRACK",
.incoming_request = chan_pjsip_incoming_prack,
}
static int chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
Definition: chan_pjsip.c:3220

Definition at line 172 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_pjsip_rtp_glue

struct ast_rtp_glue chan_pjsip_rtp_glue
static

Local glue for interacting with the RTP engine core.

Definition at line 486 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_pjsip_supplement

struct ast_sip_session_supplement chan_pjsip_supplement
static

SIP session supplement structure.

Definition at line 143 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_pjsip_supplement_response

struct ast_sip_session_supplement chan_pjsip_supplement_response
static

SIP session supplement structure just for responses.

Definition at line 155 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ chan_pjsip_tech

struct ast_channel_tech chan_pjsip_tech

PBX interface structure for channel registration.

Definition at line 109 of file chan_pjsip.c.

Referenced by chan_pjsip_new(), load_module(), and unload_module().

◆ channel_type

const char channel_type[] = "PJSIP"
static

Definition at line 78 of file chan_pjsip.c.

Referenced by load_module().

◆ direct_media_mitigation_info

struct ast_datastore_info direct_media_mitigation_info = { }
static

Definition at line 274 of file chan_pjsip.c.

Referenced by chan_pjsip_session_begin().

◆ dtmf_mode_function

struct ast_custom_function dtmf_mode_function
static
Initial value:
= {
.name = "PJSIP_DTMF_MODE",
}
int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
PJSIP_DTMF_MODE function write callback.
int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
PJSIP_DTMF_MODE function read callback.

Definition at line 3260 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ media_offer_function

struct ast_custom_function media_offer_function
static
Initial value:
= {
.name = "PJSIP_MEDIA_OFFER",
}
int pjsip_acf_media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
PJSIP_MEDIA_OFFER function read callback.
int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
PJSIP_MEDIA_OFFER function write callback.

Definition at line 3254 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ moh_passthrough_function

struct ast_custom_function moh_passthrough_function
static
Initial value:
= {
.name = "PJSIP_MOH_PASSTHROUGH",
}
int pjsip_acf_moh_passthrough_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
PJSIP_MOH_PASSTHROUGH function read callback.
int pjsip_acf_moh_passthrough_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
PJSIP_MOH_PASSTHROUGH function write callback.

Definition at line 3266 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ pbx_start_supplement

struct ast_sip_session_supplement pbx_start_supplement
static
Initial value:
= {
.method = "INVITE",
.incoming_request = pbx_start_incoming_request,
}
static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
Definition: chan_pjsip.c:3079

Definition at line 3113 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ pjsip_uids_onhold

struct ao2_container* pjsip_uids_onhold
static

◆ refer_callback_module

pjsip_module refer_callback_module
static
Initial value:
= {
.name = { "REFER Callback", 14 },
.id = -1,
}

REFER Callback module, used to attach session data structure to subscription.

Definition at line 1951 of file chan_pjsip.c.

Referenced by load_module(), transfer_refer(), unload_module(), and xfer_client_on_evsub_state().

◆ session_refresh_function

struct ast_custom_function session_refresh_function
static
Initial value:
= {
.name = "PJSIP_SEND_SESSION_REFRESH",
}
int pjsip_acf_session_refresh_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
PJSIP_SEND_SESSION_REFRESH function write callback.

Definition at line 3272 of file chan_pjsip.c.

Referenced by load_module(), and unload_module().

◆ transport_info

struct ast_datastore_info transport_info
static
Initial value:
= {
.type = "chan_pjsip_transport_info",
}
static void transport_info_destroy(void *obj)
Destructor function for transport_info_data.
Definition: chan_pjsip.c:261

Datastore used to store local/remote addresses for the INVITE request that created the PJSIP channel.

Definition at line 269 of file chan_pjsip.c.

Referenced by chan_pjsip_incoming_request().

◆ uniqueid_threadbuf

struct ast_threadstorage uniqueid_threadbuf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_uniqueid_threadbuf , .custom_init = NULL , }
static

Definition at line 75 of file chan_pjsip.c.

Referenced by chan_pjsip_get_uniqueid().