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_module *	AST_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_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. More...
static int	chan_pjsip_devicestate (const char *data)
	Function called to get the device state of an endpoint. More...
static int	chan_pjsip_digit_begin (struct ast_channel *chan, char digit)
	Function called by core to start a DTMF digit. More...
static int	chan_pjsip_digit_end (struct ast_channel *ast, char digit, unsigned int duration)
	Function called by core to stop a DTMF digit. More...
static int	chan_pjsip_fixup (struct ast_channel *oldchan, struct ast_channel *newchan)
	Function called by core to change the underlying owner channel. More...
static void	chan_pjsip_get_codec (struct ast_channel *chan, struct ast_format_cap *result)
	Function called by RTP engine to get peer capabilities. More...
static int	chan_pjsip_get_hold (const char *chan_uid)
	Determine whether a channel ID is in the list of PJSIP channels on hold. More...
static enum ast_rtp_glue_result	chan_pjsip_get_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
	Function called by RTP engine to get local audio RTP peer. More...
static const char *	chan_pjsip_get_uniqueid (struct ast_channel *ast)
static enum ast_rtp_glue_result	chan_pjsip_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
	Function called by RTP engine to get local video RTP peer. More...
static int	chan_pjsip_hangup (struct ast_channel *ast)
	Function called by core to hang up a PJSIP session. More...
static int	chan_pjsip_incoming_ack (struct ast_sip_session *session, struct pjsip_rx_data *rdata)
static int	chan_pjsip_incoming_request (struct ast_sip_session *session, struct pjsip_rx_data *rdata)
	Function called when a request is received on the session. More...
static void	chan_pjsip_incoming_response (struct ast_sip_session *session, struct pjsip_rx_data *rdata)
	Function called when a response is received on the session. More...
static void	chan_pjsip_incoming_response_update_cause (struct ast_sip_session *session, struct pjsip_rx_data *rdata)
	Function called when a response is received on the session. More...
static int	chan_pjsip_indicate (struct ast_channel *ast, int condition, const void *data, size_t datalen)
	Function called by core to ask the channel to indicate some sort of condition. More...
static struct ast_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. 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_frame *	chan_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_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 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. 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_data *	hangup_data_alloc (int cause, struct ast_channel *chan)
static void	hangup_data_destroy (void *obj)
static int	hangup_sip2cause (int cause)
	Convert SIP hangup causes to Asterisk hangup causes. More...
static int	indicate (void *data)
static struct indicate_data *	indicate_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_data *	info_dtmf_data_alloc (struct ast_sip_session *session, char digit, unsigned int duration)
static void	info_dtmf_data_destroy (void *obj)
static int	is_colp_update_allowed (struct ast_sip_session *session)
static int	is_compatible_format (struct ast_sip_session *session, struct ast_frame *f)
	Determine if the given frame is in a format we've negotiated. More...
static int	load_module (void)
	Load the module. More...
static void	local_hold_set_state (struct ast_sip_session_media *session_media, unsigned int held)
	Callback which changes the value of locally held on the media stream. More...
static int	on_topology_change_response (struct ast_sip_session *session, pjsip_rx_data *rdata)
static int	pbx_start_incoming_request (struct ast_sip_session *session, pjsip_rx_data *rdata)
static int	remote_send_hold (void *data)
	Update local hold state to be held. More...
static int	remote_send_hold_refresh (struct ast_sip_session *session, unsigned int held)
	Update local hold state and send a re-INVITE with the new SDP. More...
static int	remote_send_unhold (void *data)
	Update local hold state to be unheld. More...
static int	request (void *obj)
static struct rtp_direct_media_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 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_data *	sendtext_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_data *	topology_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_data *	transfer_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 = "30ef0c93b36035ec78c9cfd712d36d9b" , .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_pjsip,res_pjsip_session,res_pjsip_pubsub", }
static const struct ast_module_info *	ast_module_info = &__mod_info
static struct ast_sip_session_supplement	call_pickup_supplement
static unsigned int	chan_idx
static struct ast_sip_session_supplement	chan_pjsip_ack_supplement
static struct ast_custom_function	chan_pjsip_dial_contacts_function
static struct ast_custom_function	chan_pjsip_parse_uri_function
static struct ast_rtp_glue	chan_pjsip_rtp_glue
	Local glue for interacting with the RTP engine core. More...
static struct ast_sip_session_supplement	chan_pjsip_supplement
	SIP session supplement structure. More...
static struct ast_sip_session_supplement	chan_pjsip_supplement_response
	SIP session supplement structure just for responses. More...
struct ast_channel_tech	chan_pjsip_tech
	PBX interface structure for channel registration. More...
static const char	channel_type [] = "PJSIP"
static struct ast_datastore_info	direct_media_mitigation_info = { }
static struct ast_custom_function	dtmf_mode_function
static struct ast_custom_function	media_offer_function
static struct ast_custom_function	moh_passthrough_function
static struct ast_sip_session_supplement	pbx_start_supplement
static struct ao2_container *	pjsip_uids_onhold
static pjsip_module	refer_callback_module
	REFER Callback module, used to attach session data structure to subscription. More...
static struct ast_custom_function	session_refresh_function
static struct ast_datastore_info	transport_info
	Datastore used to store local/remote addresses for the INVITE request that created the PJSIP channel. More...
static struct ast_threadstorage	uniqueid_threadbuf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_uniqueid_threadbuf , .custom_init = NULL , }

Detailed Description

PSJIP SIP Channel Driver.

Author

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

Definition in file chan_pjsip.c.

Macro Definition Documentation

UNIQUEID_BUFSIZE

#define UNIQUEID_BUFSIZE   256

Definition at line 76 of file chan_pjsip.c.

Referenced by chan_pjsip_get_uniqueid().

Function Documentation

__init_uniqueid_threadbuf()

static void __init_uniqueid_threadbuf ( void  )

static

Definition at line 75 of file chan_pjsip.c.

{

__reg_module()

static void __reg_module ( void  )

static

Definition at line 3438 of file chan_pjsip.c.

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 3438 of file chan_pjsip.c.

answer()

static int answer ( void *  data )

static

Definition at line 682 of file chan_pjsip.c.

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

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

{
 struct answer_data *ans_data = data;
 pj_status_t status = PJ_SUCCESS;
 pjsip_tx_data *packet = NULL;
 struct ast_sip_session *session = ans_data->session;
 SCOPE_ENTER_TASK(1, ans_data->indent, "%s\n", ast_sip_session_get_name(session));

 if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
 ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
 session->inv_session->cause,
 pjsip_get_status_text(session->inv_session->cause)->ptr);
 SCOPE_EXIT_RTN_VALUE(0, "Disconnected\n");
 }

 pjsip_dlg_inc_lock(session->inv_session->dlg);
 if (session->inv_session->invite_tsx) {
 status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
 } else {
 ast_log(LOG_ERROR,"Cannot answer '%s' because there is no associated SIP transaction\n",
 ast_channel_name(session->channel));
 }
 pjsip_dlg_dec_lock(session->inv_session->dlg);

 if (status == PJ_SUCCESS && packet) {
 ast_sip_session_send_response(session, packet);
 }

 if (status != PJ_SUCCESS) {
 char err[PJ_ERR_MSG_SIZE];

 pj_strerror(status, err, sizeof(err));
 ast_log(LOG_WARNING,"Cannot answer '%s': %s\n",
 ast_channel_name(session->channel), err);
 /*
 * Return this value so we can distinguish between this
 * failure and the threadpool synchronous push failing.
 */
 SCOPE_EXIT_RTN_VALUE(-2, "pjproject failure\n");
 }
 SCOPE_EXIT_RTN_VALUE(0);
}

AST_MODULE_SELF_SYM()

struct ast_module* AST_MODULE_SELF_SYM

(

void

)

Definition at line 3438 of file chan_pjsip.c.

call()

static int call ( void *  data )

static

Definition at line 2358 of file chan_pjsip.c.

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

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

{
 struct ast_sip_channel_pvt *channel = data;
 struct ast_sip_session *session = channel->session;
 pjsip_tx_data *tdata;
 int res = 0;
 SCOPE_ENTER(1, "%s Topology: %s\n",
 ast_sip_session_get_name(session),
 ast_str_tmp(256, ast_stream_topology_to_str(channel->session->pending_media_state->topology, &STR_TMP))
 );


 res = ast_sip_session_create_invite(session, &tdata);

 if (res) {
 ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
 ast_queue_hangup(session->channel);
 } else {
 set_channel_on_rtp_instance(session, ast_channel_uniqueid(session->channel));
 update_initial_connected_line(session);
 ast_sip_session_send_request(session, tdata);
 }
 ao2_ref(channel, -1);
 SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
}

call_pickup_incoming_request()

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

static

Definition at line 3065 of file chan_pjsip.c.

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

{
 struct ast_features_pickup_config *pickup_cfg;
 struct ast_channel *chan;

 /* Check for a to-tag to determine if this is a reinvite */
 if (rdata->msg_info.to->tag.slen) {
 /* We don't care about reinvites */
 return 0;
 }

 pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
 if (!pickup_cfg) {
 ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n");
 return 0;
 }

 if (strcmp(session->exten, pickup_cfg->pickupexten)) {
 ao2_ref(pickup_cfg, -1);
 return 0;
 }
 ao2_ref(pickup_cfg, -1);

 /* We can't directly use session->channel because the pickup operation will cause a masquerade to occur,
 * changing the channel pointer in session to a different channel. To ensure we work on the right channel
 * we store a pointer locally before we begin and keep a reference so it remains valid no matter what.
 */
 chan = ast_channel_ref(session->channel);
 if (ast_pickup_call(chan)) {
 ast_channel_hangupcause_set(chan, AST_CAUSE_CALL_REJECTED);
 } else {
 ast_channel_hangupcause_set(chan, AST_CAUSE_NORMAL_CLEARING);
 }
 /* A hangup always occurs because the pickup operation will have either failed resulting in the call
 * needing to be hung up OR the pickup operation was a success and the channel we now have is actually
 * the channel that was replaced, which should be hung up since it is literally in limbo not connected
 * to anything at all.
 */
 ast_hangup(chan);
 ast_channel_unref(chan);

 return 1;
}

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

0	Channel has been added to or was already in the hold list
-1	Failed to add channel to the hold list

Definition at line 1117 of file chan_pjsip.c.

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

Referenced by chan_pjsip_indicate().

{
 RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);

 hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
 if (hold_uid) {
 /* Device is already on hold. Nothing to do. */
 return 0;
 }

 /* Device wasn't in hold list already. Create a new one. */
 hold_uid = ao2_alloc_options(strlen(chan_uid) + 1, NULL,
 AO2_ALLOC_OPT_LOCK_NOLOCK);
 if (!hold_uid) {
 return -1;
 }

 ast_copy_string(hold_uid, chan_uid, strlen(chan_uid) + 1);

 if (ao2_link(pjsip_uids_onhold, hold_uid) == 0) {
 return -1;
 }

 return 0;
}

chan_pjsip_answer()

static int chan_pjsip_answer ( struct ast_channel *  ast )

static

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

Definition at line 726 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 struct ast_sip_session *session;
 struct answer_data ans_data = { 0, };
 int res;
 SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));

 if (ast_channel_state(ast) == AST_STATE_UP) {
 SCOPE_EXIT_RTN_VALUE(0, "Already up\n");
 return 0;
 }

 ast_setstate(ast, AST_STATE_UP);
 session = ao2_bump(channel->session);

 /* the answer task needs to be pushed synchronously otherwise a race condition
 can occur between this thread and bridging (specifically when native bridging
 attempts to do direct media) */
 ast_channel_unlock(ast);
 ans_data.session = session;
 ans_data.indent = ast_trace_get_indent();
 res = ast_sip_push_task_wait_serializer(session->serializer, answer, &ans_data);
 if (res) {
 if (res == -1) {
 ast_log(LOG_ERROR,"Cannot answer '%s': Unable to push answer task to the threadpool.\n",
 ast_channel_name(session->channel));
 }
 ao2_ref(session, -1);
 ast_channel_lock(ast);
 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
 }
 ao2_ref(session, -1);
 ast_channel_lock(ast);

 SCOPE_EXIT_RTN_VALUE(0);
}

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 2385 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 SCOPE_ENTER(1, "%s Topology: %s\n", ast_sip_session_get_name(channel->session),
 ast_str_tmp(256, ast_stream_topology_to_str(channel->session->pending_media_state->topology, &STR_TMP)));

 ao2_ref(channel, +1);
 if (ast_sip_push_task(channel->session->serializer, call, channel)) {
 ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest);
 ao2_cleanup(channel);
 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
 }

 SCOPE_EXIT_RTN_VALUE(0, "'call' task pushed\n");
}

chan_pjsip_cng_tone_detected()

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

static

Internal helper function called when CNG tone is detected.

Definition at line 765 of file chan_pjsip.c.

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

Referenced by chan_pjsip_read_stream().

{
 const char *target_context;
 int exists;
 int dsp_features;

 dsp_features = ast_dsp_get_features(session->dsp);
 dsp_features &= ~DSP_FEATURE_FAX_DETECT;
 if (dsp_features) {
 ast_dsp_set_features(session->dsp, dsp_features);
 } else {
 ast_dsp_free(session->dsp);
 session->dsp = NULL;
 }

 /* If already executing in the fax extension don't do anything */
 if (!strcmp(ast_channel_exten(ast), "fax")) {
 return f;
 }

 target_context = S_OR(ast_channel_macrocontext(ast), ast_channel_context(ast));

 /*
 * We need to unlock the channel here because ast_exists_extension has the
 * potential to start and stop an autoservice on the channel. Such action
 * is prone to deadlock if the channel is locked.
 *
 * ast_async_goto() has its own restriction on not holding the channel lock.
 */
 ast_channel_unlock(ast);
 ast_frfree(f);
 f = &ast_null_frame;
 exists = ast_exists_extension(ast, target_context, "fax", 1,
 S_COR(ast_channel_caller(ast)->id.number.valid,
 ast_channel_caller(ast)->id.number.str, NULL));
 if (exists) {
 ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
 ast_channel_name(ast));
 pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
 if (ast_async_goto(ast, target_context, "fax", 1)) {
 ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
 ast_channel_name(ast), target_context);
 }
 } else {
 ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
 ast_channel_name(ast), target_context);
 }

 /* It's possible for a masquerade to have occurred when doing the ast_async_goto resulting in
 * the channel on the session having changed. Since we need to return with the original channel
 * locked we lock the channel that was passed in and not session->channel.
 */
 ast_channel_lock(ast);

 return f;
}

chan_pjsip_devicestate()

static int chan_pjsip_devicestate ( const char *  data )

static

Function called to get the device state of an endpoint.

Definition at line 1174 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
 enum ast_device_state state = AST_DEVICE_UNKNOWN;
 RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
 struct ast_devstate_aggregate aggregate;
 int num, inuse = 0;

 if (!endpoint) {
 return AST_DEVICE_INVALID;
 }

 endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
 ast_endpoint_get_resource(endpoint->persistent));

 if (!endpoint_snapshot) {
 return AST_DEVICE_INVALID;
 }

 if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
 state = AST_DEVICE_UNAVAILABLE;
 } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
 state = AST_DEVICE_NOT_INUSE;
 }

 if (!endpoint_snapshot->num_channels) {
 return state;
 }

 ast_devstate_aggregate_init(&aggregate);

 for (num = 0; num < endpoint_snapshot->num_channels; num++) {
 struct ast_channel_snapshot *snapshot;

 snapshot = ast_channel_snapshot_get_latest(endpoint_snapshot->channel_ids[num]);
 if (!snapshot) {
 continue;
 }

 if (chan_pjsip_get_hold(snapshot->base->uniqueid)) {
 ast_devstate_aggregate_add(&aggregate, AST_DEVICE_ONHOLD);
 } else {
 ast_devstate_aggregate_add(&aggregate, ast_state_chan2dev(snapshot->state));
 }

 if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
 (snapshot->state == AST_STATE_BUSY)) {
 inuse++;
 }

 ao2_ref(snapshot, -1);
 }

 if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
 state = AST_DEVICE_BUSY;
 } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
 state = ast_devstate_aggregate_result(&aggregate);
 }

 return state;
}

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 2150 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 struct ast_sip_session_media *media;

 media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];

 switch (channel->session->dtmf) {
 case AST_SIP_DTMF_RFC_4733:
 if (!media || !media->rtp) {
 return 0;
 }

 ast_rtp_instance_dtmf_begin(media->rtp, digit);
 break;
 case AST_SIP_DTMF_AUTO:
 if (!media || !media->rtp) {
 return 0;
 }

 if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
 return -1;
 }

 ast_rtp_instance_dtmf_begin(media->rtp, digit);
 break;
 case AST_SIP_DTMF_AUTO_INFO:
 if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) {
 return 0;
 }
 ast_rtp_instance_dtmf_begin(media->rtp, digit);
 break;
 case AST_SIP_DTMF_NONE:
 break;
 case AST_SIP_DTMF_INBAND:
 return -1;
 default:
 break;
 }

 return 0;
}

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 2262 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 struct ast_sip_session_media *media;

 if (!channel || !channel->session) {
 /* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */
 ast_debug(3, "Channel %s disappeared while calling digit_end\n", ast_channel_name(ast));
 return -1;
 }

 media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];

 switch (channel->session->dtmf) {
 case AST_SIP_DTMF_AUTO_INFO:
 {
 if (!media || !media->rtp) {
 return 0;
 }

 if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_NONE) {
 ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
 ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
 break;
 }
 /* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */
 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));
 }

 case AST_SIP_DTMF_INFO:
 {
 struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);

 if (!dtmf_data) {
 return -1;
 }

 if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
 ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
 ao2_cleanup(dtmf_data);
 return -1;
 }
 break;
 }
 case AST_SIP_DTMF_RFC_4733:
 if (!media || !media->rtp) {
 return 0;
 }

 ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
 break;
 case AST_SIP_DTMF_AUTO:
 if (!media || !media->rtp) {
 return 0;
 }

 if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
 return -1;
 }

 ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
 break;
 case AST_SIP_DTMF_NONE:
 break;
 case AST_SIP_DTMF_INBAND:
 return -1;
 }

 return 0;
}

chan_pjsip_fixup()

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

static

Function called by core to change the underlying owner channel.

Definition at line 1045 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);

 if (channel->session->channel != oldchan) {
 return -1;
 }

 /*
 * The masquerade has suspended the channel's session
 * serializer so we can safely change it outside of
 * the serializer thread.
 */
 channel->session->channel = newchan;

 set_channel_on_rtp_instance(channel->session, ast_channel_uniqueid(newchan));

 return 0;
}

chan_pjsip_get_codec()

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

static

Function called by RTP engine to get peer capabilities.

Definition at line 244 of file chan_pjsip.c.

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

{
 SCOPE_ENTER(1, "%s Native formats %s\n", ast_channel_name(chan),
 ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(ast_channel_nativeformats(chan), &STR_TMP)));
 ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
 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

0	The channel is not in the hold list
1	The channel is in the hold list

Definition at line 1161 of file chan_pjsip.c.

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

Referenced by chan_pjsip_devicestate().

{
 RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);

 hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
 if (!hold_uid) {
 return 0;
 }

 return 1;
}

chan_pjsip_get_rtp_peer()

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

static

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

Definition at line 171 of file chan_pjsip.c.

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

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 struct ast_sip_endpoint *endpoint;
 struct ast_datastore *datastore;
 struct ast_sip_session_media *media;

 if (!channel || !channel->session) {
 return AST_RTP_GLUE_RESULT_FORBID;
 }

 /* XXX Getting the first RTP instance for direct media related stuff seems just
 * absolutely wrong. But the native RTP bridge knows no other method than single-stream
 * for direct media. So this is the best we can do.
 */
 media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
 if (!media || !media->rtp) {
 return AST_RTP_GLUE_RESULT_FORBID;
 }

 datastore = ast_sip_session_get_datastore(channel->session, "t38");
 if (datastore) {
 ao2_ref(datastore, -1);
 return AST_RTP_GLUE_RESULT_FORBID;
 }

 endpoint = channel->session->endpoint;

 *instance = media->rtp;
 ao2_ref(*instance, +1);

 ast_assert(endpoint != NULL);
 if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
 return AST_RTP_GLUE_RESULT_FORBID;
 }

 if (endpoint->media.direct_media.enabled) {
 return AST_RTP_GLUE_RESULT_REMOTE;
 }

 return AST_RTP_GLUE_RESULT_LOCAL;
}

chan_pjsip_get_uniqueid()

static const char * chan_pjsip_get_uniqueid ( struct ast_channel *  ast )

static

Definition at line 1278 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 char *uniqueid = ast_threadstorage_get(&uniqueid_threadbuf, UNIQUEID_BUFSIZE);

 if (!uniqueid) {
 return "";
 }

 ast_copy_pj_str(uniqueid, &channel->session->inv_session->dlg->call_id->id, UNIQUEID_BUFSIZE);

 return uniqueid;
}

chan_pjsip_get_vrtp_peer()

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

static

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

Definition at line 215 of file chan_pjsip.c.

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

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 struct ast_sip_endpoint *endpoint;
 struct ast_sip_session_media *media;

 if (!channel || !channel->session) {
 return AST_RTP_GLUE_RESULT_FORBID;
 }

 media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO];
 if (!media || !media->rtp) {
 return AST_RTP_GLUE_RESULT_FORBID;
 }

 endpoint = channel->session->endpoint;

 *instance = media->rtp;
 ao2_ref(*instance, +1);

 ast_assert(endpoint != NULL);
 if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
 return AST_RTP_GLUE_RESULT_FORBID;
 }

 return AST_RTP_GLUE_RESULT_LOCAL;
}

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 2516 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 int cause;
 struct hangup_data *h_data;
 SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));

 if (!channel || !channel->session) {
 SCOPE_EXIT_RTN_VALUE(-1, "No channel or session\n");
 }

 cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
 h_data = hangup_data_alloc(cause, ast);

 if (!h_data) {
 goto failure;
 }

 if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
 ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
 goto failure;
 }

 SCOPE_EXIT_RTN_VALUE(0, "Cause: %d\n", cause);

failure:
 /* Go ahead and do our cleanup of the session and channel even if we're not going
 * to be able to send our SIP request/response
 */
 clear_session_and_channel(channel->session, ast);
 ao2_cleanup(channel);
 ao2_cleanup(h_data);

 SCOPE_EXIT_RTN_VALUE(-1, "Cause: %d\n", cause);
}

chan_pjsip_incoming_ack()

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

static

Definition at line 3241 of file chan_pjsip.c.

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

{
 SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));

 if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
 if (session->endpoint->media.direct_media.enabled && session->channel) {
 ast_trace(-1, "%s: Queueing SRCCHANGE\n", ast_sip_session_get_name(session));
 ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
 }
 }
 SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(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 3007 of file chan_pjsip.c.

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

{
 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
 struct transport_info_data *transport_data;
 pjsip_tx_data *packet = NULL;
 SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));

 if (session->channel) {
 SCOPE_EXIT_RTN_VALUE(0, "%s: No channel\n", ast_sip_session_get_name(session));
 }

 /* Check for a to-tag to determine if this is a reinvite */
 if (rdata->msg_info.to->tag.slen) {
 /* Weird case. We've received a reinvite but we don't have a channel. The most
 * typical case for this happening is that a blind transfer fails, and so the
 * transferer attempts to reinvite himself back into the call. We already got
 * rid of that channel, and the other side of the call is unrecoverable.
 *
 * We treat this as a failure, so our best bet is to just hang this call
 * up and not create a new channel. Clearing defer_terminate here ensures that
 * calling ast_sip_session_terminate() can result in a BYE being sent ASAP.
 */
 session->defer_terminate = 0;
 ast_sip_session_terminate(session, 400);
 SCOPE_EXIT_RTN_VALUE(-1, "%s: We have a To tag but no channel. Terminating session\n", ast_sip_session_get_name(session));
 }

 datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
 if (!datastore) {
 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info datastore\n", ast_sip_session_get_name(session));
 }

 transport_data = ast_calloc(1, sizeof(*transport_data));
 if (!transport_data) {
 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info\n", ast_sip_session_get_name(session));
 }
 pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
 pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
 datastore->data = transport_data;
 ast_sip_session_add_datastore(session, datastore);

 if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL, NULL))) {
 if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS
 && packet) {
 ast_sip_session_send_response(session, packet);
 }

 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n",
 ast_sip_session_get_name(session));
 }

 set_sipdomain_variable(session);

 /* channel gets created on incoming request, but we wait to call start
 so other supplements have a chance to run */
 SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
}

chan_pjsip_incoming_response()

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

static

Function called when a response is received on the session.

Definition at line 3186 of file chan_pjsip.c.

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

{
 struct pjsip_status_line status = rdata->msg_info.msg->line.status;
 SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);

 if (!session->channel) {
 SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
 }

 switch (status.code) {
 case 180: {
 pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
 if (sdp && sdp->body.ptr) {
 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
 ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
 } else {
 ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
 ast_queue_control(session->channel, AST_CONTROL_RINGING);
 }

 ast_channel_lock(session->channel);
 if (ast_channel_state(session->channel) != AST_STATE_UP) {
 ast_setstate(session->channel, AST_STATE_RINGING);
 }
 ast_channel_unlock(session->channel);
 break;
 }
 case 183:
 if (session->endpoint->ignore_183_without_sdp) {
 pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
 if (sdp && sdp->body.ptr) {
 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
 ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS with SDP\n", ast_sip_session_get_name(session),
 (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
 ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
 }
 } else {
 ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
 ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS without SDP\n", ast_sip_session_get_name(session),
 (int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
 ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
 }
 break;
 case 200:
 ast_trace(-1, "%s: Queueing ANSWER\n", ast_sip_session_get_name(session));
 ast_queue_control(session->channel, AST_CONTROL_ANSWER);
 break;
 default:
 ast_trace(-1, "%s: Not queueing anything\n", ast_sip_session_get_name(session));
 break;
 }

 SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
}

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 3156 of file chan_pjsip.c.

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

{
 struct pjsip_status_line status = rdata->msg_info.msg->line.status;
 struct ast_control_pvt_cause_code *cause_code;
 int data_size = sizeof(*cause_code);
 SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);

 if (!session->channel) {
 SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
 }

 /* Build and send the tech-specific cause information */
 /* size of the string making up the cause code is "SIP " number + " " + reason length */
 data_size += 4 + 4 + pj_strlen(&status.reason);
 cause_code = ast_alloca(data_size);
 memset(cause_code, 0, data_size);

 ast_copy_string(cause_code->chan_name, ast_channel_name(session->channel), AST_CHANNEL_NAME);

 snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %d %.*s", status.code,
 (int) pj_strlen(&status.reason), pj_strbuf(&status.reason));

 cause_code->ast_cause = hangup_sip2cause(status.code);
 ast_queue_control_data(session->channel, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
 ast_channel_hangupcause_hash_set(session->channel, cause_code, data_size);

 SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
}

chan_pjsip_indicate()

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

static

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

Definition at line 1612 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 struct ast_sip_session_media *media;
 int response_code = 0;
 int res = 0;
 char *device_buf;
 size_t device_buf_size;
 int i;
 const struct ast_stream_topology *topology;
 struct ast_frame f = {
 .frametype = AST_FRAME_CONTROL,
 .subclass = {
 .integer = condition
 },
 .datalen = datalen,
 .data.ptr = (void *)data,
 };
 char condition_name[256];
 SCOPE_ENTER(3, "%s: Indicated %s\n", ast_channel_name(ast),
 ast_frame_subclass2str(&f, condition_name, sizeof(condition_name), NULL, 0));

 switch (condition) {
 case AST_CONTROL_RINGING:
 if (ast_channel_state(ast) == AST_STATE_RING) {
 if (channel->session->endpoint->inband_progress ||
 (channel->session->inv_session && channel->session->inv_session->neg &&
 pjmedia_sdp_neg_get_state(channel->session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE)) {
 response_code = 183;
 res = -1;
 } else {
 response_code = 180;
 }
 } else {
 res = -1;
 }
 ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
 break;
 case AST_CONTROL_BUSY:
 if (ast_channel_state(ast) != AST_STATE_UP) {
 response_code = 486;
 } else {
 res = -1;
 }
 break;
 case AST_CONTROL_CONGESTION:
 if (ast_channel_state(ast) != AST_STATE_UP) {
 response_code = 503;
 } else {
 res = -1;
 }
 break;
 case AST_CONTROL_INCOMPLETE:
 if (ast_channel_state(ast) != AST_STATE_UP) {
 response_code = 484;
 } else {
 res = -1;
 }
 break;
 case AST_CONTROL_PROCEEDING:
 if (ast_channel_state(ast) != AST_STATE_UP) {
 response_code = 100;
 } else {
 res = -1;
 }
 break;
 case AST_CONTROL_PROGRESS:
 if (ast_channel_state(ast) != AST_STATE_UP) {
 response_code = 183;
 } else {
 res = -1;
 }
 ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
 break;
 case AST_CONTROL_VIDUPDATE:
 for (i = 0; i < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions); ++i) {
 media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, i);
 if (!media || media->type != AST_MEDIA_TYPE_VIDEO) {
 continue;
 }
 if (media->rtp) {
 /* FIXME: Only use this for VP8. Additional work would have to be done to
 * fully support other video codecs */

 if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL ||
 ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp9) != AST_FORMAT_CMP_NOT_EQUAL ||
 ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h265) != AST_FORMAT_CMP_NOT_EQUAL ||
 (channel->session->endpoint->media.webrtc &&
 ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h264) != AST_FORMAT_CMP_NOT_EQUAL)) {
 /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
 * RTP engine would provide a way to externally write/schedule RTCP
 * packets */
 struct ast_frame fr;
 fr.frametype = AST_FRAME_CONTROL;
 fr.subclass.integer = AST_CONTROL_VIDUPDATE;
 res = ast_rtp_instance_write(media->rtp, &fr);
 } else {
 ao2_ref(channel->session, +1);
 if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
 ao2_cleanup(channel->session);
 }
 }
 ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Success");
 } else {
 ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Failure");
 res = -1;
 }
 }
 /* XXX If there were no video streams, then this should set
 * res to -1
 */
 break;
 case AST_CONTROL_CONNECTED_LINE:
 ao2_ref(channel->session, +1);
 if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
 ao2_cleanup(channel->session);
 }
 break;
 case AST_CONTROL_UPDATE_RTP_PEER:
 break;
 case AST_CONTROL_PVT_CAUSE_CODE:
 res = -1;
 break;
 case AST_CONTROL_MASQUERADE_NOTIFY:
 ast_assert(datalen == sizeof(int));
 if (*(int *) data) {
 /*
 * Masquerade is beginning:
 * Wait for session serializer to get suspended.
 */
 ast_channel_unlock(ast);
 ast_sip_session_suspend(channel->session);
 ast_channel_lock(ast);
 } else {
 /*
 * Masquerade is complete:
 * Unsuspend the session serializer.
 */
 ast_sip_session_unsuspend(channel->session);
 }
 break;
 case AST_CONTROL_HOLD:
 chan_pjsip_add_hold(ast_channel_uniqueid(ast));
 device_buf_size = strlen(ast_channel_name(ast)) + 1;
 device_buf = alloca(device_buf_size);
 ast_channel_get_device_name(ast, device_buf, device_buf_size);
 ast_devstate_changed_literal(AST_DEVICE_ONHOLD, 1, device_buf);
 if (!channel->session->moh_passthrough) {
 ast_moh_start(ast, data, NULL);
 } else {
 if (ast_sip_push_task(channel->session->serializer, remote_send_hold, ao2_bump(channel->session))) {
 ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n",
 ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
 ao2_ref(channel->session, -1);
 }
 }
 break;
 case AST_CONTROL_UNHOLD:
 chan_pjsip_remove_hold(ast_channel_uniqueid(ast));
 device_buf_size = strlen(ast_channel_name(ast)) + 1;
 device_buf = alloca(device_buf_size);
 ast_channel_get_device_name(ast, device_buf, device_buf_size);
 ast_devstate_changed_literal(AST_DEVICE_UNKNOWN, 1, device_buf);
 if (!channel->session->moh_passthrough) {
 ast_moh_stop(ast);
 } else {
 if (ast_sip_push_task(channel->session->serializer, remote_send_unhold, ao2_bump(channel->session))) {
 ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n",
 ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
 ao2_ref(channel->session, -1);
 }
 }
 break;
 case AST_CONTROL_SRCUPDATE:
 break;
 case AST_CONTROL_SRCCHANGE:
 break;
 case AST_CONTROL_REDIRECTING:
 if (ast_channel_state(ast) != AST_STATE_UP) {
 response_code = 181;
 } else {
 res = -1;
 }
 break;
 case AST_CONTROL_T38_PARAMETERS:
 res = 0;

 if (channel->session->t38state == T38_PEER_REINVITE) {
 const struct ast_control_t38_parameters *parameters = data;

 if (parameters->request_response == AST_T38_REQUEST_PARMS) {
 res = AST_T38_REQUEST_PARMS;
 }
 }

 break;
 case AST_CONTROL_STREAM_TOPOLOGY_REQUEST_CHANGE:
 topology = data;
 ast_trace(-1, "%s: New topology: %s\n", ast_channel_name(ast),
 ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
 res = handle_topology_request_change(channel->session, topology);
 break;
 case AST_CONTROL_STREAM_TOPOLOGY_CHANGED:
 break;
 case AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED:
 break;
 case -1:
 res = -1;
 break;
 default:
 ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
 res = -1;
 break;
 }

 if (response_code) {
 struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);

 if (!ind_data) {
 SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc indicate data\n", ast_channel_name(ast));
 }

 if (ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
 ast_log(LOG_ERROR, "%s: Cannot send response code %d to endpoint %s. Could not queue task properly\n",
 ast_channel_name(ast), response_code, ast_sorcery_object_get_id(channel->session->endpoint));
 ao2_cleanup(ind_data);
 res = -1;
 }
 }

 SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_channel_name(ast));
}

chan_pjsip_new()

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

static

Function called to create a new PJSIP Asterisk channel.

Definition at line 547 of file chan_pjsip.c.

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

Referenced by chan_pjsip_incoming_request(), and chan_pjsip_request_with_stream_topology().

{
 struct ast_channel *chan;
 struct ast_format_cap *caps;
 RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
 struct ast_sip_channel_pvt *channel;
 struct ast_variable *var;
 struct ast_stream_topology *topology;
 SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));

 if (!(pvt = ao2_alloc_options(sizeof(*pvt), chan_pjsip_pvt_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK))) {
 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt\n");
 }

 chan = ast_channel_alloc_with_endpoint(1, state,
 S_COR(session->id.number.valid, session->id.number.str, ""),
 S_COR(session->id.name.valid, session->id.name.str, ""),
 session->endpoint->accountcode,
 exten, session->endpoint->context,
 assignedids, requestor, 0,
 session->endpoint->persistent, "PJSIP/%s-%08x",
 ast_sorcery_object_get_id(session->endpoint),
 (unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
 if (!chan) {
 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
 }

 ast_channel_tech_set(chan, &chan_pjsip_tech);

 if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
 ast_channel_unlock(chan);
 ast_hangup(chan);
 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt channel\n");
 }

 ast_channel_tech_pvt_set(chan, channel);

 if (!ast_stream_topology_get_count(session->pending_media_state->topology) ||
 !compatible_formats_exist(session->pending_media_state->topology, session->endpoint->media.codecs)) {
 caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
 if (!caps) {
 ast_channel_unlock(chan);
 ast_hangup(chan);
 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create caps\n");
 }
 ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
 topology = ast_stream_topology_clone(session->endpoint->media.topology);
 } else {
 caps = ast_stream_topology_get_formats(session->pending_media_state->topology);
 topology = ast_stream_topology_clone(session->pending_media_state->topology);
 }

 if (!topology || !caps) {
 ao2_cleanup(caps);
 ast_stream_topology_free(topology);
 ast_channel_unlock(chan);
 ast_hangup(chan);
 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't get caps or clone topology\n");
 }

 ast_channel_stage_snapshot(chan);

 ast_channel_nativeformats_set(chan, caps);
 ast_channel_set_stream_topology(chan, topology);

 if (!ast_format_cap_empty(caps)) {
 struct ast_format *fmt;

 fmt = ast_format_cap_get_best_by_type(caps, AST_MEDIA_TYPE_AUDIO);
 if (!fmt) {
 /* Since our capabilities aren't empty, this will succeed */
 fmt = ast_format_cap_get_format(caps, 0);
 }
 ast_channel_set_writeformat(chan, fmt);
 ast_channel_set_rawwriteformat(chan, fmt);
 ast_channel_set_readformat(chan, fmt);
 ast_channel_set_rawreadformat(chan, fmt);
 ao2_ref(fmt, -1);
 }

 ao2_ref(caps, -1);

 if (state == AST_STATE_RING) {
 ast_channel_rings_set(chan, 1);
 }

 ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);

 ast_party_id_copy(&ast_channel_caller(chan)->id, &session->id);
 ast_party_id_copy(&ast_channel_caller(chan)->ani, &session->id);

 if (!ast_strlen_zero(exten)) {
 /* Set provided DNID on the new channel. */
 ast_channel_dialed(chan)->number.str = ast_strdup(exten);
 }

 ast_channel_priority_set(chan, 1);

 ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
 ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup);

 ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups);
 ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups);

 if (!ast_strlen_zero(session->endpoint->language)) {
 ast_channel_language_set(chan, session->endpoint->language);
 }

 if (!ast_strlen_zero(session->endpoint->zone)) {
 struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
 if (!zone) {
 ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
 }
 ast_channel_zone_set(chan, zone);
 }

 for (var = session->endpoint->channel_vars; var; var = var->next) {
 char buf[512];
 pbx_builtin_setvar_helper(chan, var->name, ast_get_encoded_str(
 var->value, buf, sizeof(buf)));
 }

 ast_channel_stage_snapshot_done(chan);
 ast_channel_unlock(chan);

 set_channel_on_rtp_instance(session, ast_channel_uniqueid(chan));

 SCOPE_EXIT_RTN_VALUE(chan);
}

chan_pjsip_pvt_dtor()

static void chan_pjsip_pvt_dtor ( void *  obj )

static

Definition at line 82 of file chan_pjsip.c.

References chan_pjsip_answer(), chan_pjsip_call(), chan_pjsip_devicestate(), chan_pjsip_digit_begin(), chan_pjsip_digit_end(), chan_pjsip_fixup(), chan_pjsip_get_uniqueid(), chan_pjsip_hangup(), chan_pjsip_indicate(), chan_pjsip_queryoption(), chan_pjsip_read_stream(), chan_pjsip_request(), chan_pjsip_request_with_stream_topology(), chan_pjsip_sendtext(), chan_pjsip_sendtext_data(), chan_pjsip_transfer(), chan_pjsip_write(), chan_pjsip_write_stream(), ast_channel::data, ast_frame::data, ast_frame::datalen, digit, ast_frame::stream_num, text, timeout, and type.

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 1237 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 int res = -1;
 enum ast_t38_state state = T38_STATE_UNAVAILABLE;

 if (!channel) {
 return -1;
 }

 switch (option) {
 case AST_OPTION_T38_STATE:
 if (channel->session->endpoint->media.t38.enabled) {
 switch (channel->session->t38state) {
 case T38_LOCAL_REINVITE:
 case T38_PEER_REINVITE:
 state = T38_STATE_NEGOTIATING;
 break;
 case T38_ENABLED:
 state = T38_STATE_NEGOTIATED;
 break;
 case T38_REJECTED:
 state = T38_STATE_REJECTED;
 break;
 default:
 state = T38_STATE_UNKNOWN;
 break;
 }
 }

 *((enum ast_t38_state *) data) = state;
 res = 0;

 break;
 default:
 break;
 }

 return res;
}

chan_pjsip_read_stream()

static struct ast_frame * chan_pjsip_read_stream ( struct ast_channel *  ast )

static

Function called by core to read any waiting frames.

Note

The channel is already locked.

Definition at line 838 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 struct ast_sip_session *session = channel->session;
 struct ast_sip_session_media_read_callback_state *callback_state;
 struct ast_frame *f;
 int fdno = ast_channel_fdno(ast) - AST_EXTENDED_FDS;
 struct ast_frame *cur;

 if (fdno >= AST_VECTOR_SIZE(&session->active_media_state->read_callbacks)) {
 return &ast_null_frame;
 }

 callback_state = AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, fdno);
 f = callback_state->read_callback(session, callback_state->session);

 if (!f) {
 return f;
 }

 for (cur = f; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
 if (cur->frametype == AST_FRAME_VOICE) {
 break;
 }
 }

 if (!cur || callback_state->session != session->active_media_state->default_session[callback_state->session->type]) {
 return f;
 }

 session = channel->session;

 /*
 * Asymmetric RTP only has one native format set at a time.
 * Therefore we need to update the native format to the current
 * raw read format BEFORE the native format check
 */
 if (!session->endpoint->asymmetric_rtp_codec &&
 ast_format_cmp(ast_channel_rawwriteformat(ast), cur->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL &&
 is_compatible_format(session, cur)) {
 struct ast_format_cap *caps;

 /* For maximum compatibility we ensure that the formats match that of the received media */
 ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
 ast_format_get_name(cur->subclass.format), ast_channel_name(ast),
 ast_format_get_name(ast_channel_rawwriteformat(ast)));

 caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
 if (caps) {
 ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN);
 ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO);
 ast_format_cap_append(caps, cur->subclass.format, 0);
 ast_channel_nativeformats_set(ast, caps);
 ao2_ref(caps, -1);
 }

 ast_set_write_format_path(ast, ast_channel_writeformat(ast), cur->subclass.format);
 ast_set_read_format_path(ast, ast_channel_readformat(ast), cur->subclass.format);

 if (ast_channel_is_bridged(ast)) {
 ast_channel_set_unbridged_nolock(ast, 1);
 }
 }

 if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), cur->subclass.format)
 == AST_FORMAT_CMP_NOT_EQUAL) {
 ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n",
 ast_format_get_name(cur->subclass.format), ast_channel_name(ast));
 ast_frfree(f);
 return &ast_null_frame;
 }

 if (session->dsp) {
 int dsp_features;

 dsp_features = ast_dsp_get_features(session->dsp);
 if ((dsp_features & DSP_FEATURE_FAX_DETECT)
 && session->endpoint->faxdetect_timeout
 && session->endpoint->faxdetect_timeout <= ast_channel_get_up_time(ast)) {
 dsp_features &= ~DSP_FEATURE_FAX_DETECT;
 if (dsp_features) {
 ast_dsp_set_features(session->dsp, dsp_features);
 } else {
 ast_dsp_free(session->dsp);
 session->dsp = NULL;
 }
 ast_debug(3, "Channel driver fax CNG detection timeout on %s\n",
 ast_channel_name(ast));
 }
 }
 if (session->dsp) {
 f = ast_dsp_process(ast, session->dsp, f);
 if (f && (f->frametype == AST_FRAME_DTMF)) {
 if (f->subclass.integer == 'f') {
 ast_debug(3, "Channel driver fax CNG detected on %s\n",
 ast_channel_name(ast));
 f = chan_pjsip_cng_tone_detected(ast, session, f);
 /* When chan_pjsip_cng_tone_detected returns it is possible for the
 * channel pointed to by ast and by session->channel to differ due to a
 * masquerade. It's best not to touch things after this.
 */
 } else {
 ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
 ast_channel_name(ast));
 }
 }
 }

 return f;
}

chan_pjsip_remove_hold()

static void chan_pjsip_remove_hold ( const char *  chan_uid )

static

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

Parameters

chan_uid

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

Definition at line 1148 of file chan_pjsip.c.

References ao2_find, OBJ_NODATA, OBJ_SEARCH_KEY, and OBJ_UNLINK.

Referenced by chan_pjsip_indicate(), and chan_pjsip_session_end().

{
 ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY | OBJ_UNLINK | OBJ_NODATA);
}

chan_pjsip_request()

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

static

Definition at line 2694 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_stream_topology *topology;
 struct ast_channel *chan;

 topology = ast_stream_topology_create_from_format_cap(cap);
 if (!topology) {
 return NULL;
 }

 chan = chan_pjsip_request_with_stream_topology(type, topology, assignedids, requestor, data, cause);

 ast_stream_topology_free(topology);

 return chan;
}

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 2667 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor(), and chan_pjsip_request().

{
 struct request_data req_data;
 RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
 SCOPE_ENTER(1, "%s Topology: %s\n", data,
 ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));

 req_data.topology = topology;
 req_data.dest = data;
 /* Default failure value in case ast_sip_push_task_wait_servant() itself fails. */
 req_data.cause = AST_CAUSE_FAILURE;

 if (ast_sip_push_task_wait_servant(NULL, request, &req_data)) {
 *cause = req_data.cause;
 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't push task\n");
 }

 session = req_data.session;

 if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) {
 /* Session needs to be terminated prematurely */
 SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
 }

 SCOPE_EXIT_RTN_VALUE(session->channel, "Channel: %s\n", ast_channel_name(session->channel));
}

chan_pjsip_sendtext()

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

static

Definition at line 2837 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_msg_data *msg;
 int rc;
 struct ast_msg_data_attribute attrs[] =
 {
 {
 .type = AST_MSG_DATA_ATTR_BODY,
 .value = (char *)text,
 }
 };

 msg = ast_msg_data_alloc(AST_MSG_DATA_SOURCE_TYPE_UNKNOWN, attrs, ARRAY_LEN(attrs));
 if (!msg) {
 return -1;
 }
 rc = chan_pjsip_sendtext_data(ast, msg);
 ast_free(msg);

 return rc;
}

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 2815 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor(), and chan_pjsip_sendtext().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 struct sendtext_data *data = sendtext_data_create(ast, msg);

 ast_debug(1, "Sending MESSAGE from '%s' to '%s:%s': %s\n",
 ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_FROM),
 ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_TO),
 ast_channel_name(ast),
 ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_BODY));

 if (!data) {
 return -1;
 }

 if (ast_sip_push_task(channel->session->serializer, sendtext, data)) {
 ao2_ref(data, -1);
 return -1;
 }
 return 0;
}

chan_pjsip_session_begin()

static void chan_pjsip_session_begin ( struct ast_sip_session *  session )

static

SIP session interaction functions.

Definition at line 2950 of file chan_pjsip.c.

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

{
 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
 SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));

 if (session->endpoint->media.direct_media.glare_mitigation ==
 AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
 SCOPE_EXIT_RTN("Direct media no glare mitigation\n");
 }

 datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
 "direct_media_glare_mitigation");

 if (!datastore) {
 SCOPE_EXIT_RTN("Couldn't create datastore\n");
 }

 ast_sip_session_add_datastore(session, datastore);
 SCOPE_EXIT_RTN();
}

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 2972 of file chan_pjsip.c.

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

{
 SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));

 if (!session->channel) {
 SCOPE_EXIT_RTN("No channel\n");
 }

 chan_pjsip_remove_hold(ast_channel_uniqueid(session->channel));

 ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
 if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
 int cause = hangup_sip2cause(session->inv_session->cause);

 ast_queue_hangup_with_cause(session->channel, cause);
 } else {
 ast_queue_hangup(session->channel);
 }

 SCOPE_EXIT_RTN();
}

chan_pjsip_set_rtp_peer()

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

static

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

Definition at line 448 of file chan_pjsip.c.

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

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 struct ast_sip_session *session = channel->session;
 struct rtp_direct_media_data *cdata;
 SCOPE_ENTER(1, "%s %s\n", ast_channel_name(chan),
 ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(cap, &STR_TMP)));

 /* Don't try to do any direct media shenanigans on early bridges */
 if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) {
 ast_debug(4, "Disregarding setting RTP on %s: channel is not bridged\n", ast_channel_name(chan));
 SCOPE_EXIT_RTN_VALUE(0, "Channel not bridged\n");
 }

 if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
 ast_debug(4, "Disregarding setting RTP on %s: NAT is active\n", ast_channel_name(chan));
 SCOPE_EXIT_RTN_VALUE(0, "NAT is active\n");
 }

 cdata = rtp_direct_media_data_create(chan, rtp, vrtp, cap, session);
 if (!cdata) {
 SCOPE_EXIT_RTN_VALUE(0);
 }

 if (ast_sip_push_task(session->serializer, send_direct_media_request, cdata)) {
 ast_log(LOG_ERROR, "Unable to send direct media request for channel %s\n", ast_channel_name(chan));
 ao2_ref(cdata, -1);
 }

 SCOPE_EXIT_RTN_VALUE(0);
}

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 2131 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
 struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);

 if (!trnf_data) {
 return -1;
 }

 if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
 ast_log(LOG_WARNING, "Error requesting transfer\n");
 ao2_cleanup(trnf_data);
 return -1;
 }

 return 0;
}

chan_pjsip_write()

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

static

Definition at line 1039 of file chan_pjsip.c.

References chan_pjsip_write_stream().

Referenced by chan_pjsip_pvt_dtor().

{
 return chan_pjsip_write_stream(ast, -1, frame);
}

chan_pjsip_write_stream()

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

static

Definition at line 949 of file chan_pjsip.c.

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

Referenced by chan_pjsip_pvt_dtor(), and chan_pjsip_write().

{
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 struct ast_sip_session *session = channel->session;
 struct ast_sip_session_media *media = NULL;
 int res = 0;

 /* The core provides a guarantee that the stream will exist when we are called if stream_num is provided */
 if (stream_num >= 0) {
 /* What is not guaranteed is that a media session will exist */
 if (stream_num < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions)) {
 media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, stream_num);
 }
 }

 switch (frame->frametype) {
 case AST_FRAME_VOICE:
 if (!media) {
 return 0;
 } else if (media->type != AST_MEDIA_TYPE_AUDIO) {
 ast_debug(3, "Channel %s stream %d is of type '%s', not audio!\n",
 ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
 return 0;
 } else if (media == channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO] &&
 ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
 struct ast_str *cap_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
 struct ast_str *write_transpath = ast_str_alloca(256);
 struct ast_str *read_transpath = ast_str_alloca(256);

 ast_log(LOG_WARNING,
 "Channel %s asked to send %s frame when native formats are %s (rd:%s->%s;%s wr:%s->%s;%s)\n",
 ast_channel_name(ast),
 ast_format_get_name(frame->subclass.format),
 ast_format_cap_get_names(ast_channel_nativeformats(ast), &cap_buf),
 ast_format_get_name(ast_channel_rawreadformat(ast)),
 ast_format_get_name(ast_channel_readformat(ast)),
 ast_translate_path_to_str(ast_channel_readtrans(ast), &read_transpath),
 ast_format_get_name(ast_channel_writeformat(ast)),
 ast_format_get_name(ast_channel_rawwriteformat(ast)),
 ast_translate_path_to_str(ast_channel_writetrans(ast), &write_transpath));
 return 0;
 } else if (media->write_callback) {
 res = media->write_callback(session, media, frame);

 }
 break;
 case AST_FRAME_VIDEO:
 if (!media) {
 return 0;
 } else if (media->type != AST_MEDIA_TYPE_VIDEO) {
 ast_debug(3, "Channel %s stream %d is of type '%s', not video!\n",
 ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
 return 0;
 } else if (media->write_callback) {
 res = media->write_callback(session, media, frame);
 }
 break;
 case AST_FRAME_MODEM:
 if (!media) {
 return 0;
 } else if (media->type != AST_MEDIA_TYPE_IMAGE) {
 ast_debug(3, "Channel %s stream %d is of type '%s', not image!\n",
 ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
 return 0;
 } else if (media->write_callback) {
 res = media->write_callback(session, media, frame);
 }
 break;
 case AST_FRAME_CNG:
 break;
 case AST_FRAME_RTCP:
 /* We only support writing out feedback */
 if (frame->subclass.integer != AST_RTP_RTCP_PSFB || !media) {
 return 0;
 } else if (media->type != AST_MEDIA_TYPE_VIDEO) {
 ast_debug(3, "Channel %s stream %d is of type '%s', not video! Unable to write RTCP feedback.\n",
 ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
 return 0;
 } else if (media->write_callback) {
 res = media->write_callback(session, media, frame);
 }
 break;
 default:
 ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP\n", frame->frametype);
 break;
 }

 return res;
}

check_for_rtp_changes()

static int check_for_rtp_changes ( struct ast_channel *  chan, struct ast_rtp_instance *  rtp, struct ast_sip_session_media *  media, struct ast_sip_session *  session  )

static

Precondition

chan is locked

Definition at line 319 of file chan_pjsip.c.

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

Referenced by send_direct_media_request().

{
 int changed = 0, position = -1;

 if (media->rtp) {
 position = rtp_find_rtcp_fd_position(session, media->rtp);
 }

 if (rtp) {
 changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
 if (media->rtp) {
 if (position != -1) {
 ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, -1);
 }
 ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
 }
 } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
 ast_sockaddr_setnull(&media->direct_media_addr);
 changed = 1;
 if (media->rtp) {
 /* Direct media has ended - reset time of last received RTP packet
 * to avoid premature RTP timeout. Synchronisation between the
 * modification of direct_mdedia_addr+last_rx here and reading the
 * values in res_pjsip_sdp_rtp.c:rtp_check_timeout() is provided
 * by the channel's lock (which is held while this function is
 * executed).
 */
 ast_rtp_instance_set_last_rx(media->rtp, time(NULL));
 ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
 if (position != -1) {
 ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, ast_rtp_instance_fd(media->rtp, 1));
 }
 }
 }

 return changed;
}

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 2476 of file chan_pjsip.c.

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

Referenced by chan_pjsip_hangup(), and hangup().

{
 session->channel = NULL;
 set_channel_on_rtp_instance(session, "");
 ast_channel_tech_pvt_set(ast, NULL);
}

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

top	Topology
cap	Format capabilities

Return values

1	The topology has at least one compatible format
0	The topology has no compatible formats or an error occurred.

Definition at line 526 of file chan_pjsip.c.

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

Referenced by chan_pjsip_new().

{
 struct ast_format_cap *cap_from_top;
 int res;
 SCOPE_ENTER(1, "Topology: %s Formats: %s\n",
 ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_stream_topology_to_str(top, &STR_TMP)),
 ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(cap, &STR_TMP)));

 cap_from_top = ast_stream_topology_get_formats(top);

 if (!cap_from_top) {
 SCOPE_EXIT_RTN_VALUE(0, "Topology had no formats\n");
 }

 res = ast_format_cap_iscompatible(cap_from_top, cap);
 ao2_ref(cap_from_top, -1);

 SCOPE_EXIT_RTN_VALUE(res, "Compatible? %s\n", res ? "yes" : "no");
}

direct_media_mitigate_glare()

static int direct_media_mitigate_glare ( struct ast_sip_session *  session )

static

Definition at line 268 of file chan_pjsip.c.

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

Referenced by send_direct_media_request().

{
 RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);

 if (session->endpoint->media.direct_media.glare_mitigation ==
 AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
 return 0;
 }

 datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
 if (!datastore) {
 return 0;
 }

 /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
 ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");

 if ((session->endpoint->media.direct_media.glare_mitigation ==
 AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING &&
 session->inv_session->role == PJSIP_ROLE_UAC) ||
 (session->endpoint->media.direct_media.glare_mitigation ==
 AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING &&
 session->inv_session->role == PJSIP_ROLE_UAS)) {
 return 1;
 }

 return 0;
}

handle_topology_request_change()

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

static

Definition at line 1592 of file chan_pjsip.c.

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

Referenced by chan_pjsip_indicate().

{
 struct topology_change_refresh_data *refresh_data;
 int res;
 SCOPE_ENTER(1);

 refresh_data = topology_change_refresh_data_alloc(session, proposed);
 if (!refresh_data) {
 SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create refresh_data\n");
 }

 res = ast_sip_push_task(session->serializer, send_topology_change_refresh, refresh_data);
 if (res) {
 topology_change_refresh_data_free(refresh_data);
 }
 SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
}

hangup()

static int hangup ( void *  data )

static

Definition at line 2483 of file chan_pjsip.c.

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

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

{
 struct hangup_data *h_data = data;
 struct ast_channel *ast = h_data->chan;
 struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
 SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));

 /*
 * Before cleaning we have to ensure that channel or its session is not NULL
 * we have seen rare case when taskprocessor calls hangup but channel is NULL
 * due to SIP session timeout and answer happening at the same time
 */
 if (channel) {
 struct ast_sip_session *session = channel->session;
 if (session) {
 int cause = h_data->cause;

 /*
 * It's possible that session_terminate might cause the session to be destroyed
 * immediately so we need to keep a reference to it so we can NULL session->channel
 * afterwards.
 */
 ast_sip_session_terminate(ao2_bump(session), cause);
 clear_session_and_channel(session, ast);
 ao2_cleanup(session);
 }
 ao2_cleanup(channel);
 }
 ao2_cleanup(h_data);
 SCOPE_EXIT_RTN_VALUE(0);
}

hangup_cause2sip()

static int hangup_cause2sip ( int  cause )

static

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

Definition at line 2402 of file chan_pjsip.c.

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

Referenced by chan_pjsip_hangup().

{
 switch (cause) {
 case AST_CAUSE_UNALLOCATED: /* 1 */
 case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
 case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
 return 404;
 case AST_CAUSE_CONGESTION: /* 34 */
 case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
 return 503;
 case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
 return 408;
 case AST_CAUSE_NO_ANSWER: /* 19 */
 case AST_CAUSE_UNREGISTERED: /* 20 */
 return 480;
 case AST_CAUSE_CALL_REJECTED: /* 21 */
 return 403;
 case AST_CAUSE_NUMBER_CHANGED: /* 22 */
 return 410;
 case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
 return 480;
 case AST_CAUSE_INVALID_NUMBER_FORMAT:
 return 484;
 case AST_CAUSE_USER_BUSY:
 return 486;
 case AST_CAUSE_FAILURE:
 return 500;
 case AST_CAUSE_FACILITY_REJECTED: /* 29 */
 return 501;
 case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
 return 503;
 case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
 return 502;
 case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
 return 488;
 case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
 return 500;
 case AST_CAUSE_NOTDEFINED:
 default:
 ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
 return 0;
 }

 /* Never reached */
 return 0;
}

hangup_data_alloc()

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

static

Definition at line 2461 of file chan_pjsip.c.

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

Referenced by chan_pjsip_hangup().

{
 struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);

 if (!h_data) {
 return NULL;
 }

 h_data->cause = cause;
 h_data->chan = ast_channel_ref(chan);

 return h_data;
}

hangup_data_destroy()

static void hangup_data_destroy ( void *  obj )

static

Definition at line 2454 of file chan_pjsip.c.

References ast_channel_unref, and hangup_data::chan.

Referenced by hangup_data_alloc().

{
 struct hangup_data *h_data = obj;

 h_data->chan = ast_channel_unref(h_data->chan);
}

hangup_sip2cause()

static int hangup_sip2cause ( int  cause )

static

Convert SIP hangup causes to Asterisk hangup causes.

Definition at line 2860 of file chan_pjsip.c.

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

Referenced by chan_pjsip_incoming_response_update_cause(), and chan_pjsip_session_end().

{
 /* Possible values taken from causes.h */

 switch(cause) {
 case 401: /* Unauthorized */
 return AST_CAUSE_CALL_REJECTED;
 case 403: /* Not found */
 return AST_CAUSE_CALL_REJECTED;
 case 404: /* Not found */
 return AST_CAUSE_UNALLOCATED;
 case 405: /* Method not allowed */
 return AST_CAUSE_INTERWORKING;
 case 407: /* Proxy authentication required */
 return AST_CAUSE_CALL_REJECTED;
 case 408: /* No reaction */
 return AST_CAUSE_NO_USER_RESPONSE;
 case 409: /* Conflict */
 return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
 case 410: /* Gone */
 return AST_CAUSE_NUMBER_CHANGED;
 case 411: /* Length required */
 return AST_CAUSE_INTERWORKING;
 case 413: /* Request entity too large */
 return AST_CAUSE_INTERWORKING;
 case 414: /* Request URI too large */
 return AST_CAUSE_INTERWORKING;
 case 415: /* Unsupported media type */
 return AST_CAUSE_INTERWORKING;
 case 420: /* Bad extension */
 return AST_CAUSE_NO_ROUTE_DESTINATION;
 case 480: /* No answer */
 return AST_CAUSE_NO_ANSWER;
 case 481: /* No answer */
 return AST_CAUSE_INTERWORKING;
 case 482: /* Loop detected */
 return AST_CAUSE_INTERWORKING;
 case 483: /* Too many hops */
 return AST_CAUSE_NO_ANSWER;
 case 484: /* Address incomplete */
 return AST_CAUSE_INVALID_NUMBER_FORMAT;
 case 485: /* Ambiguous */
 return AST_CAUSE_UNALLOCATED;
 case 486: /* Busy everywhere */
 return AST_CAUSE_BUSY;
 case 487: /* Request terminated */
 return AST_CAUSE_INTERWORKING;
 case 488: /* No codecs approved */
 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
 case 491: /* Request pending */
 return AST_CAUSE_INTERWORKING;
 case 493: /* Undecipherable */
 return AST_CAUSE_INTERWORKING;
 case 500: /* Server internal failure */
 return AST_CAUSE_FAILURE;
 case 501: /* Call rejected */
 return AST_CAUSE_FACILITY_REJECTED;
 case 502:
 return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
 case 503: /* Service unavailable */
 return AST_CAUSE_CONGESTION;
 case 504: /* Gateway timeout */
 return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
 case 505: /* SIP version not supported */
 return AST_CAUSE_INTERWORKING;
 case 600: /* Busy everywhere */
 return AST_CAUSE_USER_BUSY;
 case 603: /* Decline */
 return AST_CAUSE_CALL_REJECTED;
 case 604: /* Does not exist anywhere */
 return AST_CAUSE_UNALLOCATED;
 case 606: /* Not acceptable */
 return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
 default:
 if (cause < 500 && cause >= 400) {
 /* 4xx class error that is unknown - someting wrong with our request */
 return AST_CAUSE_INTERWORKING;
 } else if (cause < 600 && cause >= 500) {
 /* 5xx class error - problem in the remote end */
 return AST_CAUSE_CONGESTION;
 } else if (cause < 700 && cause >= 600) {
 /* 6xx - global errors in the 4xx class */
 return AST_CAUSE_INTERWORKING;
 }
 return AST_CAUSE_NORMAL;
 }
 /* Never reached */
 return 0;
}

indicate()

static int indicate ( void *  data )

static

Definition at line 1333 of file chan_pjsip.c.

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

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

{
 pjsip_tx_data *packet = NULL;
 struct indicate_data *ind_data = data;
 struct ast_sip_session *session = ind_data->session;
 int response_code = ind_data->response_code;

 if ((session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
 (pjsip_inv_answer(session->inv_session, response_code, NULL,