Files
asterisk/channels/pjsip/dialplan_functions.c
George Joseph f309ffad3d chan_pjsip: Add PJSIPHangup dialplan app and manager action
See UserNote below.

Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.

Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code.  I.E.  ast_sip_str2rc("DECLINE") returns
603.  This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).

Also extracted the XML documentation to its own file since it was
almost as large as the code itself.

UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
2023-11-07 16:32:22 +00:00

1349 lines
42 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*!
* \file
*
* \author \verbatim Joshua Colp <jcolp@digium.com> \endverbatim
* \author \verbatim Matt Jordan <mjordan@digium.com> \endverbatim
*
* \ingroup functions
*
* \brief PJSIP channel dialplan functions
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <pjsip.h>
#include <pjlib.h>
#include <pjsip_ua.h>
#include "asterisk/astobj2.h"
#include "asterisk/module.h"
#include "asterisk/acl.h"
#include "asterisk/app.h"
#include "asterisk/conversions.h"
#include "asterisk/channel.h"
#include "asterisk/stream.h"
#include "asterisk/format.h"
#include "asterisk/dsp.h"
#include "asterisk/pbx.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "include/chan_pjsip.h"
#include "include/dialplan_functions.h"
/*!
* \brief String representations of the T.38 state enum
*/
static const char *t38state_to_string[T38_MAX_ENUM] = {
[T38_DISABLED] = "DISABLED",
[T38_LOCAL_REINVITE] = "LOCAL_REINVITE",
[T38_PEER_REINVITE] = "REMOTE_REINVITE",
[T38_ENABLED] = "ENABLED",
[T38_REJECTED] = "REJECTED",
};
/*!
* \internal \brief Handle reading RTP information
*/
static int channel_read_rtp(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct ast_sip_session *session;
struct ast_sip_session_media *media;
struct ast_sockaddr addr;
if (!channel) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
return -1;
}
session = channel->session;
if (!session) {
ast_log(AST_LOG_WARNING, "Channel %s has no session!\n", ast_channel_name(chan));
return -1;
}
if (ast_strlen_zero(type)) {
ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtp' information\n");
return -1;
}
if (ast_strlen_zero(field) || !strcmp(field, "audio")) {
media = session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
} else if (!strcmp(field, "video")) {
media = session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO];
} else {
ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtp' information\n", field);
return -1;
}
if (!media || !media->rtp) {
ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
ast_channel_name(chan), S_OR(field, "audio"));
return -1;
}
if (!strcmp(type, "src")) {
ast_rtp_instance_get_local_address(media->rtp, &addr);
ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
} else if (!strcmp(type, "dest")) {
ast_rtp_instance_get_remote_address(media->rtp, &addr);
ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
} else if (!strcmp(type, "direct")) {
ast_copy_string(buf, ast_sockaddr_stringify(&media->direct_media_addr), buflen);
} else if (!strcmp(type, "secure")) {
if (media->srtp) {
struct ast_sdp_srtp *srtp = media->srtp;
int flag = ast_test_flag(srtp, AST_SRTP_CRYPTO_OFFER_OK);
snprintf(buf, buflen, "%d", flag ? 1 : 0);
} else {
snprintf(buf, buflen, "%d", 0);
}
} else if (!strcmp(type, "hold")) {
snprintf(buf, buflen, "%d", media->remotely_held ? 1 : 0);
} else {
ast_log(AST_LOG_WARNING, "Unknown type field '%s' specified for 'rtp' information\n", type);
return -1;
}
return 0;
}
/*!
* \internal \brief Handle reading RTCP information
*/
static int channel_read_rtcp(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct ast_sip_session *session;
struct ast_sip_session_media *media;
if (!channel) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
return -1;
}
session = channel->session;
if (!session) {
ast_log(AST_LOG_WARNING, "Channel %s has no session!\n", ast_channel_name(chan));
return -1;
}
if (ast_strlen_zero(type)) {
ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtcp' information\n");
return -1;
}
if (ast_strlen_zero(field) || !strcmp(field, "audio")) {
media = session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
} else if (!strcmp(field, "video")) {
media = session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO];
} else {
ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtcp' information\n", field);
return -1;
}
if (!media || !media->rtp) {
ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
ast_channel_name(chan), S_OR(field, "audio"));
return -1;
}
if (!strncasecmp(type, "all", 3)) {
enum ast_rtp_instance_stat_field stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY;
if (!strcasecmp(type, "all_jitter")) {
stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER;
} else if (!strcasecmp(type, "all_rtt")) {
stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT;
} else if (!strcasecmp(type, "all_loss")) {
stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS;
} else if (!strcasecmp(type, "all_mes")) {
stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_MES;
}
if (!ast_rtp_instance_get_quality(media->rtp, stat_field, buf, buflen)) {
ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
return -1;
}
} else {
struct ast_rtp_instance_stats stats;
int i;
struct {
const char *name;
enum { INT, DBL } type;
union {
unsigned int *i4;
double *d8;
};
} lookup[] = {
{ "txcount", INT, { .i4 = &stats.txcount, }, },
{ "rxcount", INT, { .i4 = &stats.rxcount, }, },
{ "txjitter", DBL, { .d8 = &stats.txjitter, }, },
{ "rxjitter", DBL, { .d8 = &stats.rxjitter, }, },
{ "remote_maxjitter", DBL, { .d8 = &stats.remote_maxjitter, }, },
{ "remote_minjitter", DBL, { .d8 = &stats.remote_minjitter, }, },
{ "remote_normdevjitter", DBL, { .d8 = &stats.remote_normdevjitter, }, },
{ "remote_stdevjitter", DBL, { .d8 = &stats.remote_stdevjitter, }, },
{ "local_maxjitter", DBL, { .d8 = &stats.local_maxjitter, }, },
{ "local_minjitter", DBL, { .d8 = &stats.local_minjitter, }, },
{ "local_normdevjitter", DBL, { .d8 = &stats.local_normdevjitter, }, },
{ "local_stdevjitter", DBL, { .d8 = &stats.local_stdevjitter, }, },
{ "txploss", INT, { .i4 = &stats.txploss, }, },
{ "rxploss", INT, { .i4 = &stats.rxploss, }, },
{ "remote_maxrxploss", DBL, { .d8 = &stats.remote_maxrxploss, }, },
{ "remote_minrxploss", DBL, { .d8 = &stats.remote_minrxploss, }, },
{ "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, },
{ "remote_stdevrxploss", DBL, { .d8 = &stats.remote_stdevrxploss, }, },
{ "local_maxrxploss", DBL, { .d8 = &stats.local_maxrxploss, }, },
{ "local_minrxploss", DBL, { .d8 = &stats.local_minrxploss, }, },
{ "local_normdevrxploss", DBL, { .d8 = &stats.local_normdevrxploss, }, },
{ "local_stdevrxploss", DBL, { .d8 = &stats.local_stdevrxploss, }, },
{ "rtt", DBL, { .d8 = &stats.rtt, }, },
{ "maxrtt", DBL, { .d8 = &stats.maxrtt, }, },
{ "minrtt", DBL, { .d8 = &stats.minrtt, }, },
{ "normdevrtt", DBL, { .d8 = &stats.normdevrtt, }, },
{ "stdevrtt", DBL, { .d8 = &stats.stdevrtt, }, },
{ "local_ssrc", INT, { .i4 = &stats.local_ssrc, }, },
{ "remote_ssrc", INT, { .i4 = &stats.remote_ssrc, }, },
{ "txmes", DBL, { .d8 = &stats.txmes, }, },
{ "rxmes", DBL, { .d8 = &stats.rxmes, }, },
{ "remote_maxmes", DBL, { .d8 = &stats.remote_maxmes, }, },
{ "remote_minmes", DBL, { .d8 = &stats.remote_minmes, }, },
{ "remote_normdevmes", DBL, { .d8 = &stats.remote_normdevmes, }, },
{ "remote_stdevmes", DBL, { .d8 = &stats.remote_stdevmes, }, },
{ "local_maxmes", DBL, { .d8 = &stats.local_maxmes, }, },
{ "local_minmes", DBL, { .d8 = &stats.local_minmes, }, },
{ "local_normdevmes", DBL, { .d8 = &stats.local_normdevmes, }, },
{ "local_stdevmes", DBL, { .d8 = &stats.local_stdevmes, }, },
{ NULL, },
};
if (ast_rtp_instance_get_stats(media->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
return -1;
}
for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
if (!strcasecmp(type, lookup[i].name)) {
if (lookup[i].type == INT) {
snprintf(buf, buflen, "%u", *lookup[i].i4);
} else {
snprintf(buf, buflen, "%f", *lookup[i].d8);
}
return 0;
}
}
ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'rtcp' information\n", type);
return -1;
}
return 0;
}
static int print_escaped_uri(struct ast_channel *chan, const char *type,
pjsip_uri_context_e context, const void *uri, char *buf, size_t size)
{
int res;
char *buf_copy;
res = pjsip_uri_print(context, uri, buf, size);
if (res < 0) {
ast_log(LOG_ERROR, "Channel %s: Unescaped %s too long for %d byte buffer\n",
ast_channel_name(chan), type, (int) size);
/* Empty buffer that likely is not terminated. */
buf[0] = '\0';
return -1;
}
buf_copy = ast_strdupa(buf);
ast_escape_quoted(buf_copy, buf, size);
return 0;
}
/*!
* \internal \brief Handle reading signalling information
*/
static int channel_read_pjsip(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
char *buf_copy;
pjsip_dialog *dlg;
int res = 0;
if (!channel) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
return -1;
}
dlg = channel->session->inv_session->dlg;
if (ast_strlen_zero(type)) {
ast_log(LOG_WARNING, "You must supply a type field for 'pjsip' information\n");
return -1;
} else if (!strcmp(type, "call-id")) {
snprintf(buf, buflen, "%.*s", (int) pj_strlen(&dlg->call_id->id), pj_strbuf(&dlg->call_id->id));
} else if (!strcmp(type, "secure")) {
#ifdef HAVE_PJSIP_GET_DEST_INFO
pjsip_host_info dest;
pj_pool_t *pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "secure-check", 128, 128);
pjsip_get_dest_info(dlg->target, NULL, pool, &dest);
snprintf(buf, buflen, "%d", dest.flag & PJSIP_TRANSPORT_SECURE ? 1 : 0);
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
#else
ast_log(LOG_WARNING, "Asterisk has been built against a version of pjproject which does not have the required functionality to support the 'secure' argument. Please upgrade to version 2.3 or later.\n");
return -1;
#endif
} else if (!strcmp(type, "target_uri")) {
res = print_escaped_uri(chan, type, PJSIP_URI_IN_REQ_URI, dlg->target, buf,
buflen);
} else if (!strcmp(type, "local_uri")) {
res = print_escaped_uri(chan, type, PJSIP_URI_IN_FROMTO_HDR, dlg->local.info->uri,
buf, buflen);
} else if (!strcmp(type, "local_tag")) {
ast_copy_pj_str(buf, &dlg->local.info->tag, buflen);
buf_copy = ast_strdupa(buf);
ast_escape_quoted(buf_copy, buf, buflen);
} else if (!strcmp(type, "remote_uri")) {
res = print_escaped_uri(chan, type, PJSIP_URI_IN_FROMTO_HDR,
dlg->remote.info->uri, buf, buflen);
} else if (!strcmp(type, "remote_tag")) {
ast_copy_pj_str(buf, &dlg->remote.info->tag, buflen);
buf_copy = ast_strdupa(buf);
ast_escape_quoted(buf_copy, buf, buflen);
} else if (!strcmp(type, "request_uri")) {
if (channel->session->request_uri) {
res = print_escaped_uri(chan, type, PJSIP_URI_IN_REQ_URI,
channel->session->request_uri, buf, buflen);
}
} else if (!strcmp(type, "t38state")) {
ast_copy_string(buf, t38state_to_string[channel->session->t38state], buflen);
} else if (!strcmp(type, "local_addr")) {
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
struct transport_info_data *transport_data;
datastore = ast_sip_session_get_datastore(channel->session, "transport_info");
if (!datastore) {
ast_log(AST_LOG_WARNING, "No transport information for channel %s\n", ast_channel_name(chan));
return -1;
}
transport_data = datastore->data;
if (pj_sockaddr_has_addr(&transport_data->local_addr)) {
pj_sockaddr_print(&transport_data->local_addr, buf, buflen, 3);
}
} else if (!strcmp(type, "remote_addr")) {
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
struct transport_info_data *transport_data;
datastore = ast_sip_session_get_datastore(channel->session, "transport_info");
if (!datastore) {
ast_log(AST_LOG_WARNING, "No transport information for channel %s\n", ast_channel_name(chan));
return -1;
}
transport_data = datastore->data;
if (pj_sockaddr_has_addr(&transport_data->remote_addr)) {
pj_sockaddr_print(&transport_data->remote_addr, buf, buflen, 3);
}
} else {
ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'pjsip' information\n", type);
return -1;
}
return res;
}
/*! \brief Struct used to push function arguments to task processor */
struct pjsip_func_args {
struct ast_sip_session *session;
const char *param;
const char *type;
const char *field;
char *buf;
size_t len;
int ret;
};
/*! \internal \brief Taskprocessor callback that handles the read on a PJSIP thread */
static int read_pjsip(void *data)
{
struct pjsip_func_args *func_args = data;
if (!strcmp(func_args->param, "rtp")) {
if (!func_args->session->channel) {
func_args->ret = -1;
return 0;
}
func_args->ret = channel_read_rtp(func_args->session->channel, func_args->type,
func_args->field, func_args->buf,
func_args->len);
} else if (!strcmp(func_args->param, "rtcp")) {
if (!func_args->session->channel) {
func_args->ret = -1;
return 0;
}
func_args->ret = channel_read_rtcp(func_args->session->channel, func_args->type,
func_args->field, func_args->buf,
func_args->len);
} else if (!strcmp(func_args->param, "endpoint")) {
if (!func_args->session->endpoint) {
ast_log(AST_LOG_WARNING, "Channel %s has no endpoint!\n", func_args->session->channel ?
ast_channel_name(func_args->session->channel) : "<unknown>");
func_args->ret = -1;
return 0;
}
snprintf(func_args->buf, func_args->len, "%s", ast_sorcery_object_get_id(func_args->session->endpoint));
} else if (!strcmp(func_args->param, "contact")) {
if (!func_args->session->contact) {
return 0;
}
snprintf(func_args->buf, func_args->len, "%s", ast_sorcery_object_get_id(func_args->session->contact));
} else if (!strcmp(func_args->param, "aor")) {
if (!func_args->session->aor) {
return 0;
}
snprintf(func_args->buf, func_args->len, "%s", ast_sorcery_object_get_id(func_args->session->aor));
} else if (!strcmp(func_args->param, "pjsip")) {
if (!func_args->session->channel) {
func_args->ret = -1;
return 0;
}
func_args->ret = channel_read_pjsip(func_args->session->channel, func_args->type,
func_args->field, func_args->buf,
func_args->len);
} else {
func_args->ret = -1;
}
return 0;
}
int pjsip_acf_channel_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct pjsip_func_args func_args = { 0, };
struct ast_sip_channel_pvt *channel;
char *parse = ast_strdupa(data);
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(param);
AST_APP_ARG(type);
AST_APP_ARG(field);
);
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
/* Check for zero arguments */
if (ast_strlen_zero(parse)) {
ast_log(LOG_ERROR, "Cannot call %s without arguments\n", cmd);
return -1;
}
AST_STANDARD_APP_ARGS(args, parse);
ast_channel_lock(chan);
/* Sanity check */
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
ast_channel_unlock(chan);
return 0;
}
channel = ast_channel_tech_pvt(chan);
if (!channel) {
ast_log(LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
ast_channel_unlock(chan);
return -1;
}
if (!channel->session) {
ast_log(LOG_WARNING, "Channel %s has no session\n", ast_channel_name(chan));
ast_channel_unlock(chan);
return -1;
}
func_args.session = ao2_bump(channel->session);
ast_channel_unlock(chan);
memset(buf, 0, len);
func_args.param = args.param;
func_args.type = args.type;
func_args.field = args.field;
func_args.buf = buf;
func_args.len = len;
if (ast_sip_push_task_wait_serializer(func_args.session->serializer, read_pjsip, &func_args)) {
ast_log(LOG_WARNING, "Unable to read properties of channel %s: failed to push task\n", ast_channel_name(chan));
ao2_ref(func_args.session, -1);
return -1;
}
ao2_ref(func_args.session, -1);
return func_args.ret;
}
int pjsip_acf_dial_contacts_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
const char *aor_name;
char *rest;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(endpoint_name);
AST_APP_ARG(aor_name);
AST_APP_ARG(request_user);
);
AST_STANDARD_APP_ARGS(args, data);
if (ast_strlen_zero(args.endpoint_name)) {
ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
return -1;
} else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
return -1;
}
aor_name = S_OR(args.aor_name, endpoint->aors);
if (ast_strlen_zero(aor_name)) {
ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
return -1;
} else if (!(dial = ast_str_create(len))) {
ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
return -1;
} else if (!(rest = ast_strdupa(aor_name))) {
ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
return -1;
}
while ((aor_name = ast_strip(strsep(&rest, ",")))) {
RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
struct ao2_iterator it_contacts;
struct ast_sip_contact *contact;
if (!aor) {
/* If the AOR provided is not found skip it, there may be more */
continue;
} else if (!(contacts = ast_sip_location_retrieve_aor_contacts_filtered(aor, AST_SIP_CONTACT_FILTER_REACHABLE))) {
/* No contacts are available, skip it as well */
continue;
} else if (!ao2_container_count(contacts)) {
/* We were given a container but no contacts are in it... */
continue;
}
it_contacts = ao2_iterator_init(contacts, 0);
for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
ast_str_append(&dial, -1, "PJSIP/");
if (!ast_strlen_zero(args.request_user)) {
ast_str_append(&dial, -1, "%s@", args.request_user);
}
ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
}
ao2_iterator_destroy(&it_contacts);
}
/* Trim the '&' at the end off */
ast_str_truncate(dial, ast_str_strlen(dial) - 1);
ast_copy_string(buf, ast_str_buffer(dial), len);
return 0;
}
/*! \brief Session refresh state information */
struct session_refresh_state {
/*! \brief Created proposed media state */
struct ast_sip_session_media_state *media_state;
};
/*! \brief Destructor for session refresh information */
static void session_refresh_state_destroy(void *obj)
{
struct session_refresh_state *state = obj;
ast_sip_session_media_state_free(state->media_state);
ast_free(obj);
}
/*! \brief Datastore for attaching session refresh state information */
static const struct ast_datastore_info session_refresh_datastore = {
.type = "pjsip_session_refresh",
.destroy = session_refresh_state_destroy,
};
/*! \brief Helper function which retrieves or allocates a session refresh state information datastore */
static struct session_refresh_state *session_refresh_state_get_or_alloc(struct ast_sip_session *session)
{
RAII_VAR(struct ast_datastore *, datastore, ast_sip_session_get_datastore(session, "pjsip_session_refresh"), ao2_cleanup);
struct session_refresh_state *state;
/* While the datastore refcount is decremented this is operating in the serializer so it will remain valid regardless */
if (datastore) {
return datastore->data;
}
if (!(datastore = ast_sip_session_alloc_datastore(&session_refresh_datastore, "pjsip_session_refresh"))
|| !(datastore->data = ast_calloc(1, sizeof(struct session_refresh_state)))
|| ast_sip_session_add_datastore(session, datastore)) {
return NULL;
}
state = datastore->data;
state->media_state = ast_sip_session_media_state_alloc();
if (!state->media_state) {
ast_sip_session_remove_datastore(session, "pjsip_session_refresh");
return NULL;
}
state->media_state->topology = ast_stream_topology_clone(session->endpoint->media.topology);
if (!state->media_state->topology) {
ast_sip_session_remove_datastore(session, "pjsip_session_refresh");
return NULL;
}
datastore->data = state;
return state;
}
/*! \brief Struct used to push PJSIP_PARSE_URI function arguments to task processor */
struct parse_uri_args {
const char *uri;
const char *type;
char *buf;
size_t buflen;
int ret;
};
/*! \internal \brief Taskprocessor callback that handles the PJSIP_PARSE_URI on a PJSIP thread */
static int parse_uri_cb(void *data)
{
struct parse_uri_args *args = data;
pj_pool_t *pool;
pjsip_name_addr *uri;
pjsip_sip_uri *sip_uri;
pj_str_t tmp;
args->ret = 0;
pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "ParseUri", 128, 128);
if (!pool) {
ast_log(LOG_ERROR, "Failed to allocate ParseUri endpoint pool.\n");
args->ret = -1;
return 0;
}
pj_strdup2_with_null(pool, &tmp, args->uri);
uri = (pjsip_name_addr *)pjsip_parse_uri(pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR);
if (!uri || (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
ast_log(LOG_WARNING, "Failed to parse URI '%s'\n", args->uri);
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
args->ret = -1;
return 0;
}
if (!strcmp(args->type, "scheme")) {
ast_copy_pj_str(args->buf, pjsip_uri_get_scheme(uri), args->buflen);
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
return 0;
} else if (!strcmp(args->type, "display")) {
ast_copy_pj_str(args->buf, &uri->display, args->buflen);
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
return 0;
}
sip_uri = pjsip_uri_get_uri(uri);
if (!sip_uri) {
ast_log(LOG_ERROR, "Failed to get an URI object for '%s'\n", args->uri);
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
args->ret = -1;
return 0;
}
if (!strcmp(args->type, "user")) {
ast_copy_pj_str(args->buf, &sip_uri->user, args->buflen);
} else if (!strcmp(args->type, "passwd")) {
ast_copy_pj_str(args->buf, &sip_uri->passwd, args->buflen);
} else if (!strcmp(args->type, "host")) {
ast_copy_pj_str(args->buf, &sip_uri->host, args->buflen);
} else if (!strcmp(args->type, "port")) {
snprintf(args->buf, args->buflen, "%d", sip_uri->port);
} else if (!strcmp(args->type, "user_param")) {
ast_copy_pj_str(args->buf, &sip_uri->user_param, args->buflen);
} else if (!strcmp(args->type, "method_param")) {
ast_copy_pj_str(args->buf, &sip_uri->method_param, args->buflen);
} else if (!strcmp(args->type, "transport_param")) {
ast_copy_pj_str(args->buf, &sip_uri->transport_param, args->buflen);
} else if (!strcmp(args->type, "ttl_param")) {
snprintf(args->buf, args->buflen, "%d", sip_uri->ttl_param);
} else if (!strcmp(args->type, "lr_param")) {
snprintf(args->buf, args->buflen, "%d", sip_uri->lr_param);
} else if (!strcmp(args->type, "maddr_param")) {
ast_copy_pj_str(args->buf, &sip_uri->maddr_param, args->buflen);
} else {
ast_log(AST_LOG_WARNING, "Unknown type part '%s' specified\n", args->type);
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
args->ret = -1;
return 0;
}
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
return 0;
}
int pjsip_acf_parse_uri_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t buflen)
{
struct parse_uri_args func_args = { 0, };
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(uri_str);
AST_APP_ARG(type);
);
AST_STANDARD_APP_ARGS(args, data);
if (ast_strlen_zero(args.uri_str)) {
ast_log(LOG_WARNING, "An URI must be specified when using the '%s' dialplan function\n", cmd);
return -1;
}
if (ast_strlen_zero(args.type)) {
ast_log(LOG_WARNING, "A type part of the URI must be specified when using the '%s' dialplan function\n", cmd);
return -1;
}
memset(buf, 0, buflen);
func_args.uri = args.uri_str;
func_args.type = args.type;
func_args.buf = buf;
func_args.buflen = buflen;
if (ast_sip_push_task_wait_serializer(NULL, parse_uri_cb, &func_args)) {
ast_log(LOG_WARNING, "Unable to parse URI: failed to push task\n");
return -1;
}
return func_args.ret;
}
static int media_offer_read_av(struct ast_sip_session *session, char *buf,
size_t len, enum ast_media_type media_type)
{
struct ast_stream_topology *topology;
int idx;
struct ast_stream *stream = NULL;
const struct ast_format_cap *caps;
size_t accum = 0;
if (session->inv_session->dlg->state == PJSIP_DIALOG_STATE_ESTABLISHED) {
struct session_refresh_state *state;
/* As we've already answered we need to store our media state until we are ready to send it */
state = session_refresh_state_get_or_alloc(session);
if (!state) {
return -1;
}
topology = state->media_state->topology;
} else {
/* The session is not yet up so we are initially answering or offering */
if (!session->pending_media_state->topology) {
session->pending_media_state->topology = ast_stream_topology_clone(session->endpoint->media.topology);
if (!session->pending_media_state->topology) {
return -1;
}
}
topology = session->pending_media_state->topology;
}
/* Find the first suitable stream */
for (idx = 0; idx < ast_stream_topology_get_count(topology); ++idx) {
stream = ast_stream_topology_get_stream(topology, idx);
if (ast_stream_get_type(stream) != media_type ||
ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
stream = NULL;
continue;
}
break;
}
/* If no suitable stream then exit early */
if (!stream) {
buf[0] = '\0';
return 0;
}
caps = ast_stream_get_formats(stream);
/* Note: buf is not terminated while the string is being built. */
for (idx = 0; idx < ast_format_cap_count(caps); ++idx) {
struct ast_format *fmt;
size_t size;
fmt = ast_format_cap_get_format(caps, idx);
/* Add one for a comma or terminator */
size = strlen(ast_format_get_name(fmt)) + 1;
if (len < size) {
ao2_ref(fmt, -1);
break;
}
/* Append the format name */
strcpy(buf + accum, ast_format_get_name(fmt));/* Safe */
ao2_ref(fmt, -1);
accum += size;
len -= size;
/* The last comma on the built string will be set to the terminator. */
buf[accum - 1] = ',';
}
/* Remove the trailing comma or terminate an empty buffer. */
buf[accum ? accum - 1 : 0] = '\0';
return 0;
}
struct media_offer_data {
struct ast_sip_session *session;
enum ast_media_type media_type;
const char *value;
};
static int media_offer_write_av(void *obj)
{
struct media_offer_data *data = obj;
struct ast_stream_topology *topology;
struct ast_stream *stream;
struct ast_format_cap *caps;
if (data->session->inv_session->dlg->state == PJSIP_DIALOG_STATE_ESTABLISHED) {
struct session_refresh_state *state;
/* As we've already answered we need to store our media state until we are ready to send it */
state = session_refresh_state_get_or_alloc(data->session);
if (!state) {
return -1;
}
topology = state->media_state->topology;
} else {
/* The session is not yet up so we are initially answering or offering */
if (!data->session->pending_media_state->topology) {
data->session->pending_media_state->topology = ast_stream_topology_clone(data->session->endpoint->media.topology);
if (!data->session->pending_media_state->topology) {
return -1;
}
}
topology = data->session->pending_media_state->topology;
}
/* XXX This method won't work when it comes time to do multistream support. The proper way to do this
* will either be to
* a) Alter all media streams of a particular type.
* b) Change the dialplan function to be able to specify which stream to alter and alter only that
* one stream
*/
stream = ast_stream_topology_get_first_stream_by_type(topology, data->media_type);
if (!stream) {
return 0;
}
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!caps) {
return -1;
}
ast_format_cap_append_from_cap(caps, ast_stream_get_formats(stream),
AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, data->media_type);
ast_format_cap_update_by_allow_disallow(caps, data->value, 1);
ast_stream_set_formats(stream, caps);
ast_stream_set_metadata(stream, "pjsip_session_refresh", "force");
ao2_ref(caps, -1);
return 0;
}
int pjsip_acf_media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct ast_sip_channel_pvt *channel;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
return -1;
}
channel = ast_channel_tech_pvt(chan);
if (!strcmp(data, "audio")) {
return media_offer_read_av(channel->session, buf, len, AST_MEDIA_TYPE_AUDIO);
} else if (!strcmp(data, "video")) {
return media_offer_read_av(channel->session, buf, len, AST_MEDIA_TYPE_VIDEO);
} else {
/* Ensure that the buffer is empty */
buf[0] = '\0';
}
return 0;
}
int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_sip_channel_pvt *channel;
struct media_offer_data mdata = {
.value = value
};
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
return -1;
}
channel = ast_channel_tech_pvt(chan);
mdata.session = channel->session;
if (!strcmp(data, "audio")) {
mdata.media_type = AST_MEDIA_TYPE_AUDIO;
} else if (!strcmp(data, "video")) {
mdata.media_type = AST_MEDIA_TYPE_VIDEO;
}
return ast_sip_push_task_wait_serializer(channel->session->serializer, media_offer_write_av, &mdata);
}
int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct ast_sip_channel_pvt *channel;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
ast_channel_unlock(chan);
return -1;
}
channel = ast_channel_tech_pvt(chan);
if (ast_sip_dtmf_to_str(channel->session->dtmf, buf, len) < 0) {
ast_log(LOG_WARNING, "Unknown DTMF mode %d on PJSIP channel %s\n", channel->session->dtmf, ast_channel_name(chan));
ast_channel_unlock(chan);
return -1;
}
ast_channel_unlock(chan);
return 0;
}
int pjsip_acf_moh_passthrough_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct ast_sip_channel_pvt *channel;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
if (len < 3) {
ast_log(LOG_WARNING, "%s: buffer too small\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
ast_channel_unlock(chan);
return -1;
}
channel = ast_channel_tech_pvt(chan);
strncpy(buf, AST_YESNO(channel->session->moh_passthrough), len);
ast_channel_unlock(chan);
return 0;
}
struct refresh_data {
struct ast_sip_session *session;
enum ast_sip_session_refresh_method method;
};
static int sip_session_response_cb(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
struct ast_format *fmt;
if (!session->channel) {
/* Egads! */
return 0;
}
fmt = ast_format_cap_get_best_by_type(ast_channel_nativeformats(session->channel), AST_MEDIA_TYPE_AUDIO);
if (!fmt) {
/* No format? That's weird. */
return 0;
}
ast_channel_set_writeformat(session->channel, fmt);
ast_channel_set_rawwriteformat(session->channel, fmt);
ast_channel_set_readformat(session->channel, fmt);
ast_channel_set_rawreadformat(session->channel, fmt);
ao2_ref(fmt, -1);
return 0;
}
static int dtmf_mode_refresh_cb(void *obj)
{
struct refresh_data *data = obj;
if (data->session->inv_session->state == PJSIP_INV_STATE_CONFIRMED) {
ast_debug(3, "Changing DTMF mode on channel %s after OFFER/ANSWER completion. Sending session refresh\n", ast_channel_name(data->session->channel));
ast_sip_session_refresh(data->session, NULL, NULL,
sip_session_response_cb, data->method, 1, NULL);
} else if (data->session->inv_session->state == PJSIP_INV_STATE_INCOMING) {
ast_debug(3, "Changing DTMF mode on channel %s during OFFER/ANSWER exchange. Updating SDP answer\n", ast_channel_name(data->session->channel));
ast_sip_session_regenerate_answer(data->session, NULL);
}
return 0;
}
int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_sip_channel_pvt *channel;
struct ast_sip_session_media *media;
int dsp_features = 0;
int dtmf = -1;
struct refresh_data rdata = {
.method = AST_SIP_SESSION_REFRESH_METHOD_INVITE,
};
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
ast_channel_unlock(chan);
return -1;
}
channel = ast_channel_tech_pvt(chan);
rdata.session = channel->session;
dtmf = ast_sip_str_to_dtmf(value);
if (dtmf == -1) {
ast_log(LOG_WARNING, "Cannot set DTMF mode to '%s' on channel '%s' as value is invalid.\n", value,
ast_channel_name(chan));
ast_channel_unlock(chan);
return -1;
}
if (channel->session->dtmf == dtmf) {
/* DTMF mode unchanged, nothing to do! */
ast_channel_unlock(chan);
return 0;
}
channel->session->dtmf = dtmf;
media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
if (media && media->rtp) {
if (channel->session->dtmf == AST_SIP_DTMF_RFC_4733) {
ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 1);
ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_RFC2833);
} else if (channel->session->dtmf == AST_SIP_DTMF_INFO) {
ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
} else if (channel->session->dtmf == AST_SIP_DTMF_INBAND) {
ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_INBAND);
} else if (channel->session->dtmf == AST_SIP_DTMF_NONE) {
ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
} else if (channel->session->dtmf == AST_SIP_DTMF_AUTO) {
if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_RFC2833) {
/* no RFC4733 negotiated, enable inband */
ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
} else if (channel->session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
/* if inband, switch to INFO */
ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
}
}
}
if (channel->session->dsp) {
dsp_features = ast_dsp_get_features(channel->session->dsp);
}
if (channel->session->dtmf == AST_SIP_DTMF_INBAND ||
channel->session->dtmf == AST_SIP_DTMF_AUTO) {
dsp_features |= DSP_FEATURE_DIGIT_DETECT;
} else {
dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
}
if (dsp_features) {
if (!channel->session->dsp) {
if (!(channel->session->dsp = ast_dsp_new())) {
ast_channel_unlock(chan);
return 0;
}
}
ast_dsp_set_features(channel->session->dsp, dsp_features);
} else if (channel->session->dsp) {
ast_dsp_free(channel->session->dsp);
channel->session->dsp = NULL;
}
ast_channel_unlock(chan);
return ast_sip_push_task_wait_serializer(channel->session->serializer, dtmf_mode_refresh_cb, &rdata);
}
int pjsip_acf_moh_passthrough_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_sip_channel_pvt *channel;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
ast_channel_unlock(chan);
return -1;
}
channel = ast_channel_tech_pvt(chan);
channel->session->moh_passthrough = ast_true(value);
ast_channel_unlock(chan);
return 0;
}
static int refresh_write_cb(void *obj)
{
struct refresh_data *data = obj;
struct session_refresh_state *state;
state = session_refresh_state_get_or_alloc(data->session);
if (!state) {
return -1;
}
ast_sip_session_refresh(data->session, NULL, NULL,
sip_session_response_cb, data->method, 1, state->media_state);
state->media_state = NULL;
ast_sip_session_remove_datastore(data->session, "pjsip_session_refresh");
return 0;
}
int pjsip_acf_session_refresh_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_sip_channel_pvt *channel;
struct refresh_data rdata = {
.method = AST_SIP_SESSION_REFRESH_METHOD_INVITE,
};
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
if (ast_channel_state(chan) != AST_STATE_UP) {
ast_log(LOG_WARNING, "'%s' not allowed on unanswered channel '%s'.\n", cmd, ast_channel_name(chan));
return -1;
}
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
return -1;
}
channel = ast_channel_tech_pvt(chan);
rdata.session = channel->session;
if (!strcmp(value, "invite")) {
rdata.method = AST_SIP_SESSION_REFRESH_METHOD_INVITE;
} else if (!strcmp(value, "update")) {
rdata.method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
}
return ast_sip_push_task_wait_serializer(channel->session->serializer, refresh_write_cb, &rdata);
}
struct hangup_data {
struct ast_sip_session *session;
int response_code;
};
/*!
* \brief Serializer task to hangup channel
*/
static int pjsip_hangup(void *obj)
{
struct hangup_data *hdata = obj;
pjsip_tx_data *packet = NULL;
if ((hdata->session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
(pjsip_inv_answer(hdata->session->inv_session, hdata->response_code, NULL, NULL, &packet) == PJ_SUCCESS)) {
ast_sip_session_send_response(hdata->session, packet);
}
return 0;
}
/*!
* \brief Callback that validates the response code
*/
static int response_code_validator(const char *channel_name,
const char *response) {
int response_code;
int rc = ast_str_to_int(response, &response_code);
if (rc != 0) {
response_code = ast_sip_str2rc(response);
if (response_code < 0) {
ast_log(LOG_WARNING, "%s: Unrecognized response code parameter '%s'."
" Defaulting to 603 DECLINE\n",
channel_name, response);
return PJSIP_SC_DECLINE;
}
}
if (response_code < 400 || response_code > 699) {
ast_log(LOG_WARNING, "%s: Response code %d is out of range 400 -> 699."
" Defaulting to 603 DECLINE\n",
channel_name, response_code);
return PJSIP_SC_DECLINE;
}
return response_code;
}
/*!
* \brief Called by pjsip_app_hangup and pjsip_action_hangup
* to actually perform the hangup
*/
static void pjsip_app_hangup_handler(struct ast_channel *chan, int response_code)
{
struct ast_sip_channel_pvt *channel;
struct hangup_data hdata = { NULL, -1 };
const char *tag = ast_channel_name(chan);
hdata.response_code = response_code;
ast_channel_lock(chan);
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "%s: Not a PJSIP channel\n", tag);
ast_channel_unlock(chan);
return;
}
channel = ast_channel_tech_pvt(chan);
hdata.session = channel->session;
if (hdata.session->inv_session->role != PJSIP_ROLE_UAS || (
hdata.session->inv_session->state != PJSIP_INV_STATE_INCOMING &&
hdata.session->inv_session->state != PJSIP_INV_STATE_EARLY)) {
ast_log(LOG_WARNING, "%s: Not an incoming channel or invalid state '%s'\n",
tag, pjsip_inv_state_name(hdata.session->inv_session->state));
ast_channel_unlock(chan);
return;
}
ast_channel_unlock(chan);
if (ast_sip_push_task_wait_serializer(channel->session->serializer,
pjsip_hangup, &hdata) != 0) {
ast_log(LOG_WARNING, "%s: failed to push hangup task to serializer\n", tag);
}
return;
}
/*!
* \brief PJSIPHangup Dialplan App
*/
int pjsip_app_hangup(struct ast_channel *chan, const char *data)
{
int response_code;
const char *tag = ast_channel_name(chan);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "%s: Missing response code parameter\n", tag);
return -1;
}
response_code = response_code_validator(tag, data);
pjsip_app_hangup_handler(chan, response_code);
return -1;
}
/*!
* \brief PJSIPHangup Manager Action
*/
int pjsip_action_hangup(struct mansession *s, const struct message *m)
{
return ast_manager_hangup_helper(s, m,
pjsip_app_hangup_handler, response_code_validator);
}