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asterisk/res/ari/resource_channels.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2012 - 2013, Digium, Inc.
*
* David M. Lee, II <dlee@digium.com>
*
* 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
*
* \brief Implementation for ARI stubs.
*
* \author David M. Lee, II <dlee@digium.com>
*/
/*** MODULEINFO
<depend type="module">res_stasis_app_playback</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/pbx.h"
#include "asterisk/bridge.h"
#include "asterisk/callerid.h"
#include "asterisk/stasis_app.h"
#include "asterisk/stasis_app_playback.h"
#include "asterisk/stasis_app_recording.h"
#include "asterisk/stasis_app_snoop.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/causes.h"
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
#include "asterisk/format_cache.h"
ARI/res_stasis: Subscribe to both Local channel halves when originating to app This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close ........ Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07 02:15:00 +00:00
#include "asterisk/core_local.h"
#include "asterisk/dial.h"
#include "resource_channels.h"
#include <limits.h>
/*!
* \brief Finds the control object for a channel, filling the response with an
* error, if appropriate.
* \param[out] response Response to fill with an error if control is not found.
* \param channel_id ID of the channel to lookup.
* \return Channel control object.
* \return \c NULL if control object does not exist.
*/
static struct stasis_app_control *find_control(
struct ast_ari_response *response,
const char *channel_id)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
ast_assert(response != NULL);
control = stasis_app_control_find_by_channel_id(channel_id);
if (control == NULL) {
/* Distinguish between 404 and 409 errors */
RAII_VAR(struct ast_channel *, chan, NULL, ao2_cleanup);
chan = ast_channel_get_by_name(channel_id);
if (chan == NULL) {
ast_ari_response_error(response, 404, "Not Found",
"Channel not found");
return NULL;
}
ast_ari_response_error(response, 409, "Conflict",
"Channel not in Stasis application");
return NULL;
}
ao2_ref(control, +1);
return control;
}
void ast_ari_channels_continue_in_dialplan(
struct ast_variable *headers,
struct ast_ari_channels_continue_in_dialplan_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, snapshot, NULL, ao2_cleanup);
int ipri;
const char *context;
const char *exten;
ast_assert(response != NULL);
control = find_control(response, args->channel_id);
if (control == NULL) {
return;
}
snapshot = stasis_app_control_get_snapshot(control);
if (!snapshot) {
return;
}
if (ast_strlen_zero(args->context)) {
context = snapshot->context;
exten = S_OR(args->extension, snapshot->exten);
} else {
context = args->context;
exten = S_OR(args->extension, "s");
}
if (!ast_strlen_zero(args->label)) {
/* A label was provided in the request, use that */
if (sscanf(args->label, "%30d", &ipri) != 1) {
ipri = ast_findlabel_extension(NULL, context, exten, args->label, NULL);
if (ipri == -1) {
ast_log(AST_LOG_ERROR, "Requested label: %s can not be found in context: %s\n", args->label, context);
ast_ari_response_error(response, 404, "Not Found", "Requested label can not be found");
return;
}
} else {
ast_debug(3, "Numeric value provided for label, jumping to that priority\n");
}
if (ipri == 0) {
ast_log(AST_LOG_ERROR, "Invalid priority label '%s' specified for extension %s in context: %s\n",
args->label, exten, context);
ast_ari_response_error(response, 400, "Bad Request", "Requested priority is illegal");
return;
}
} else if (args->priority) {
/* No label provided, use provided priority */
ipri = args->priority;
} else if (ast_strlen_zero(args->context) && ast_strlen_zero(args->extension)) {
/* Special case. No exten, context, or priority provided, then move on to the next priority */
ipri = snapshot->priority + 1;
} else {
ipri = 1;
}
if (stasis_app_control_continue(control, context, exten, ipri)) {
ast_ari_response_alloc_failed(response);
return;
}
ast_ari_response_no_content(response);
}
ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan ........ Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-12 20:34:37 +00:00
void ast_ari_channels_redirect(struct ast_variable *headers,
struct ast_ari_channels_redirect_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, chan_snapshot, NULL, ao2_cleanup);
char *tech;
char *resource;
int tech_len;
control = find_control(response, args->channel_id);
if (!control) {
return;
}
if (ast_strlen_zero(args->endpoint)) {
ast_ari_response_error(response, 400, "Not Found",
"Required parameter 'endpoint' not provided.");
return;
}
tech = ast_strdupa(args->endpoint);
if (!(resource = strchr(tech, '/')) || !(tech_len = resource - tech)) {
ast_ari_response_error(response, 422, "Unprocessable Entity",
"Endpoint parameter '%s' does not contain tech/resource", args->endpoint);
return;
}
*resource++ = '\0';
if (ast_strlen_zero(resource)) {
ast_ari_response_error(response, 422, "Unprocessable Entity",
"No resource provided in endpoint parameter '%s'", args->endpoint);
return;
}
chan_snapshot = ast_channel_snapshot_get_latest(args->channel_id);
if (!chan_snapshot) {
ast_ari_response_error(response, 500, "Internal Server Error",
"Unable to find channel snapshot for '%s'", args->channel_id);
return;
}
if (strncasecmp(chan_snapshot->type, tech, tech_len)) {
ast_ari_response_error(response, 422, "Unprocessable Entity",
"Endpoint technology '%s' does not match channel technology '%s'",
tech, chan_snapshot->type);
return;
}
if (stasis_app_control_redirect(control, resource)) {
ast_ari_response_error(response, 500, "Internal Server Error",
"Failed to redirect channel");
return;
}
ast_ari_response_no_content(response);
}
void ast_ari_channels_answer(struct ast_variable *headers,
struct ast_ari_channels_answer_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
return;
}
if (stasis_app_control_answer(control) != 0) {
ast_ari_response_error(
response, 500, "Internal Server Error",
"Failed to answer channel");
return;
}
ast_ari_response_no_content(response);
}
void ast_ari_channels_ring(struct ast_variable *headers,
struct ast_ari_channels_ring_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
return;
}
stasis_app_control_ring(control);
ast_ari_response_no_content(response);
}
void ast_ari_channels_ring_stop(struct ast_variable *headers,
struct ast_ari_channels_ring_stop_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
return;
}
stasis_app_control_ring_stop(control);
ast_ari_response_no_content(response);
}
void ast_ari_channels_mute(struct ast_variable *headers,
struct ast_ari_channels_mute_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
unsigned int direction = 0;
enum ast_frame_type frametype = AST_FRAME_VOICE;
control = find_control(response, args->channel_id);
if (control == NULL) {
return;
}
if (ast_strlen_zero(args->direction)) {
ast_ari_response_error(
response, 400, "Bad Request",
"Direction is required");
return;
}
if (!strcmp(args->direction, "in")) {
direction = AST_MUTE_DIRECTION_READ;
} else if (!strcmp(args->direction, "out")) {
direction = AST_MUTE_DIRECTION_WRITE;
} else if (!strcmp(args->direction, "both")) {
direction = AST_MUTE_DIRECTION_READ | AST_MUTE_DIRECTION_WRITE;
} else {
ast_ari_response_error(
response, 400, "Bad Request",
"Invalid direction specified");
return;
}
stasis_app_control_mute(control, direction, frametype);
ast_ari_response_no_content(response);
}
void ast_ari_channels_unmute(struct ast_variable *headers,
struct ast_ari_channels_unmute_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
unsigned int direction = 0;
enum ast_frame_type frametype = AST_FRAME_VOICE;
control = find_control(response, args->channel_id);
if (control == NULL) {
return;
}
if (ast_strlen_zero(args->direction)) {
ast_ari_response_error(
response, 400, "Bad Request",
"Direction is required");
return;
}
if (!strcmp(args->direction, "in")) {
direction = AST_MUTE_DIRECTION_READ;
} else if (!strcmp(args->direction, "out")) {
direction = AST_MUTE_DIRECTION_WRITE;
} else if (!strcmp(args->direction, "both")) {
direction = AST_MUTE_DIRECTION_READ | AST_MUTE_DIRECTION_WRITE;
} else {
ast_ari_response_error(
response, 400, "Bad Request",
"Invalid direction specified");
return;
}
stasis_app_control_unmute(control, direction, frametype);
ast_ari_response_no_content(response);
}
void ast_ari_channels_send_dtmf(struct ast_variable *headers,
struct ast_ari_channels_send_dtmf_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
return;
}
if (ast_strlen_zero(args->dtmf)) {
ast_ari_response_error(
response, 400, "Bad Request",
"DTMF is required");
return;
}
stasis_app_control_dtmf(control, args->dtmf, args->before, args->between, args->duration, args->after);
ast_ari_response_no_content(response);
}
void ast_ari_channels_hold(struct ast_variable *headers,
struct ast_ari_channels_hold_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
/* Response filled in by find_control */
return;
}
stasis_app_control_hold(control);
ast_ari_response_no_content(response);
}
void ast_ari_channels_unhold(struct ast_variable *headers,
struct ast_ari_channels_unhold_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
/* Response filled in by find_control */
return;
}
stasis_app_control_unhold(control);
ast_ari_response_no_content(response);
}
void ast_ari_channels_start_moh(struct ast_variable *headers,
struct ast_ari_channels_start_moh_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
/* Response filled in by find_control */
return;
}
stasis_app_control_moh_start(control, args->moh_class);
ast_ari_response_no_content(response);
}
void ast_ari_channels_stop_moh(struct ast_variable *headers,
struct ast_ari_channels_stop_moh_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
/* Response filled in by find_control */
return;
}
stasis_app_control_moh_stop(control);
ast_ari_response_no_content(response);
}
void ast_ari_channels_start_silence(struct ast_variable *headers,
struct ast_ari_channels_start_silence_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
/* Response filled in by find_control */
return;
}
stasis_app_control_silence_start(control);
ast_ari_response_no_content(response);
}
void ast_ari_channels_stop_silence(struct ast_variable *headers,
struct ast_ari_channels_stop_silence_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
control = find_control(response, args->channel_id);
if (control == NULL) {
/* Response filled in by find_control */
return;
}
stasis_app_control_silence_stop(control);
ast_ari_response_no_content(response);
}
static void ari_channels_handle_play(
const char *args_channel_id,
const char *args_media,
const char *args_lang,
int args_offsetms,
int args_skipms,
const char *args_playback_id,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, snapshot, NULL, ao2_cleanup);
RAII_VAR(struct stasis_app_playback *, playback, NULL, ao2_cleanup);
RAII_VAR(char *, playback_url, NULL, ast_free);
struct ast_json *json;
const char *language;
ast_assert(response != NULL);
control = find_control(response, args_channel_id);
if (control == NULL) {
/* Response filled in by find_control */
return;
}
snapshot = stasis_app_control_get_snapshot(control);
if (!snapshot) {
ast_ari_response_error(
response, 404, "Not Found",
"Channel not found");
return;
}
if (args_skipms < 0) {
ast_ari_response_error(
response, 400, "Bad Request",
"skipms cannot be negative");
return;
}
if (args_offsetms < 0) {
ast_ari_response_error(
response, 400, "Bad Request",
"offsetms cannot be negative");
return;
}
language = S_OR(args_lang, snapshot->language);
playback = stasis_app_control_play_uri(control, args_media, language,
args_channel_id, STASIS_PLAYBACK_TARGET_CHANNEL, args_skipms, args_offsetms, args_playback_id);
if (!playback) {
ast_ari_response_error(
response, 500, "Internal Server Error",
"Failed to queue media for playback");
return;
}
if (ast_asprintf(&playback_url, "/playback/%s",
stasis_app_playback_get_id(playback)) == -1) {
playback_url = NULL;
ast_ari_response_error(
response, 500, "Internal Server Error",
"Out of memory");
return;
}
json = stasis_app_playback_to_json(playback);
if (!json) {
ast_ari_response_error(
response, 500, "Internal Server Error",
"Out of memory");
return;
}
ast_ari_response_created(response, playback_url, json);
}
void ast_ari_channels_play(struct ast_variable *headers,
struct ast_ari_channels_play_args *args,
struct ast_ari_response *response)
{
ari_channels_handle_play(
args->channel_id,
args->media,
args->lang,
args->offsetms,
args->skipms,
args->playback_id,
response);
}
void ast_ari_channels_play_with_id(struct ast_variable *headers,
struct ast_ari_channels_play_with_id_args *args,
struct ast_ari_response *response)
{
ari_channels_handle_play(
args->channel_id,
args->media,
args->lang,
args->offsetms,
args->skipms,
args->playback_id,
response);
}
void ast_ari_channels_record(struct ast_variable *headers,
struct ast_ari_channels_record_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
RAII_VAR(struct stasis_app_recording *, recording, NULL, ao2_cleanup);
RAII_VAR(char *, recording_url, NULL, ast_free);
struct ast_json *json;
RAII_VAR(struct stasis_app_recording_options *, options, NULL,
ao2_cleanup);
RAII_VAR(char *, uri_encoded_name, NULL, ast_free);
size_t uri_name_maxlen;
ast_assert(response != NULL);
if (args->max_duration_seconds < 0) {
ast_ari_response_error(
response, 400, "Bad Request",
"max_duration_seconds cannot be negative");
return;
}
if (args->max_silence_seconds < 0) {
ast_ari_response_error(
response, 400, "Bad Request",
"max_silence_seconds cannot be negative");
return;
}
control = find_control(response, args->channel_id);
if (control == NULL) {
/* Response filled in by find_control */
return;
}
options = stasis_app_recording_options_create(args->name, args->format);
if (options == NULL) {
ast_ari_response_error(
response, 500, "Internal Server Error",
"Out of memory");
}
ast_string_field_build(options, target, "channel:%s", args->channel_id);
options->max_silence_seconds = args->max_silence_seconds;
options->max_duration_seconds = args->max_duration_seconds;
options->terminate_on =
stasis_app_recording_termination_parse(args->terminate_on);
options->if_exists =
stasis_app_recording_if_exists_parse(args->if_exists);
options->beep = args->beep;
if (options->terminate_on == STASIS_APP_RECORDING_TERMINATE_INVALID) {
ast_ari_response_error(
response, 400, "Bad Request",
"terminateOn invalid");
return;
}
if (options->if_exists == -1) {
ast_ari_response_error(
response, 400, "Bad Request",
"ifExists invalid");
return;
}
if (!ast_get_format_for_file_ext(options->format)) {
ast_ari_response_error(
response, 422, "Unprocessable Entity",
"specified format is unknown on this system");
return;
}
recording = stasis_app_control_record(control, options);
if (recording == NULL) {
switch(errno) {
case EINVAL:
/* While the arguments are invalid, we should have
* caught them prior to calling record.
*/
ast_ari_response_error(
response, 500, "Internal Server Error",
"Error parsing request");
break;
case EEXIST:
ast_ari_response_error(response, 409, "Conflict",
"Recording '%s' already exists and can not be overwritten",
args->name);
break;
case ENOMEM:
ast_ari_response_error(
response, 500, "Internal Server Error",
"Out of memory");
break;
case EPERM:
ast_ari_response_error(
response, 400, "Bad Request",
"Recording name invalid");
break;
default:
ast_log(LOG_WARNING,
"Unrecognized recording error: %s\n",
strerror(errno));
ast_ari_response_error(
response, 500, "Internal Server Error",
"Internal Server Error");
break;
}
return;
}
uri_name_maxlen = strlen(args->name) * 3;
uri_encoded_name = ast_malloc(uri_name_maxlen);
if (!uri_encoded_name) {
ast_ari_response_error(
response, 500, "Internal Server Error",
"Out of memory");
return;
}
ast_uri_encode(args->name, uri_encoded_name, uri_name_maxlen,
ast_uri_http);
if (ast_asprintf(&recording_url, "/recordings/live/%s",
uri_encoded_name) == -1) {
recording_url = NULL;
ast_ari_response_error(
response, 500, "Internal Server Error",
"Out of memory");
return;
}
json = stasis_app_recording_to_json(recording);
if (!json) {
ast_ari_response_error(
response, 500, "Internal Server Error",
"Out of memory");
return;
}
ast_ari_response_created(response, recording_url, json);
}
void ast_ari_channels_get(struct ast_variable *headers,
struct ast_ari_channels_get_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
struct stasis_cache *cache;
struct ast_channel_snapshot *snapshot;
cache = ast_channel_cache();
if (!cache) {
ast_ari_response_error(
response, 500, "Internal Server Error",
"Message bus not initialized");
return;
}
msg = stasis_cache_get(cache, ast_channel_snapshot_type(),
args->channel_id);
if (!msg) {
ast_ari_response_error(
response, 404, "Not Found",
"Channel not found");
return;
}
snapshot = stasis_message_data(msg);
ast_assert(snapshot != NULL);
ast_ari_response_ok(response,
ast_channel_snapshot_to_json(snapshot, NULL));
}
void ast_ari_channels_hangup(struct ast_variable *headers,
struct ast_ari_channels_hangup_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct ast_channel *, chan, NULL, ao2_cleanup);
int cause;
chan = ast_channel_get_by_name(args->channel_id);
if (chan == NULL) {
ast_ari_response_error(
response, 404, "Not Found",
"Channel not found");
return;
}
if (ast_strlen_zero(args->reason) || !strcmp(args->reason, "normal")) {
cause = AST_CAUSE_NORMAL;
} else if (!strcmp(args->reason, "busy")) {
cause = AST_CAUSE_BUSY;
} else if (!strcmp(args->reason, "congestion")) {
cause = AST_CAUSE_CONGESTION;
} else if (!strcmp(args->reason, "no_answer")) {
cause = AST_CAUSE_NOANSWER;
} else {
ast_ari_response_error(
response, 400, "Invalid Reason",
"Invalid reason for hangup provided");
return;
}
ast_channel_hangupcause_set(chan, cause);
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
ast_ari_response_no_content(response);
}
void ast_ari_channels_list(struct ast_variable *headers,
struct ast_ari_channels_list_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_cache *, cache, NULL, ao2_cleanup);
RAII_VAR(struct ao2_container *, snapshots, NULL, ao2_cleanup);
RAII_VAR(struct ast_json *, json, NULL, ast_json_unref);
struct ao2_iterator i;
void *obj;
struct stasis_message_sanitizer *sanitize = stasis_app_get_sanitizer();
cache = ast_channel_cache();
if (!cache) {
ast_ari_response_error(
response, 500, "Internal Server Error",
"Message bus not initialized");
return;
}
ao2_ref(cache, +1);
snapshots = stasis_cache_dump(cache, ast_channel_snapshot_type());
if (!snapshots) {
ast_ari_response_alloc_failed(response);
return;
}
json = ast_json_array_create();
if (!json) {
ast_ari_response_alloc_failed(response);
return;
}
i = ao2_iterator_init(snapshots, 0);
while ((obj = ao2_iterator_next(&i))) {
RAII_VAR(struct stasis_message *, msg, obj, ao2_cleanup);
struct ast_channel_snapshot *snapshot = stasis_message_data(msg);
int r;
if (sanitize && sanitize->channel_snapshot
&& sanitize->channel_snapshot(snapshot)) {
continue;
}
r = ast_json_array_append(
json, ast_channel_snapshot_to_json(snapshot, NULL));
if (r != 0) {
ast_ari_response_alloc_failed(response);
ao2_iterator_destroy(&i);
return;
}
}
ao2_iterator_destroy(&i);
ast_ari_response_ok(response, ast_json_ref(json));
}
/*! \brief Structure used for origination */
struct ari_origination {
/*! \brief Dialplan context */
char context[AST_MAX_CONTEXT];
/*! \brief Dialplan extension */
char exten[AST_MAX_EXTENSION];
/*! \brief Dialplan priority */
int priority;
/*! \brief Application data to pass to Stasis application */
char appdata[0];
};
/*! \brief Thread which dials and executes upon answer */
static void *ari_originate_dial(void *data)
{
struct ast_dial *dial = data;
struct ari_origination *origination = ast_dial_get_user_data(dial);
enum ast_dial_result res;
res = ast_dial_run(dial, NULL, 0);
if (res != AST_DIAL_RESULT_ANSWERED) {
goto end;
}
if (!ast_strlen_zero(origination->appdata)) {
struct ast_app *app = pbx_findapp("Stasis");
if (app) {
ast_verb(4, "Launching Stasis(%s) on %s\n", origination->appdata,
ast_channel_name(ast_dial_answered(dial)));
pbx_exec(ast_dial_answered(dial), app, origination->appdata);
} else {
ast_log(LOG_WARNING, "No such application 'Stasis'\n");
}
} else {
struct ast_channel *answered = ast_dial_answered(dial);
if (!ast_strlen_zero(origination->context)) {
ast_channel_context_set(answered, origination->context);
}
if (!ast_strlen_zero(origination->exten)) {
ast_channel_exten_set(answered, origination->exten);
}
if (origination->priority > 0) {
ast_channel_priority_set(answered, origination->priority);
}
if (ast_pbx_run(answered)) {
ast_log(LOG_ERROR, "Failed to start PBX on %s\n", ast_channel_name(answered));
} else {
/* PBX will have taken care of hanging up, so we steal the answered channel so dial doesn't do it */
ast_dial_answered_steal(dial);
}
}
end:
ast_dial_destroy(dial);
ast_free(origination);
return NULL;
}
static void ari_channels_handle_originate_with_id(const char *args_endpoint,
const char *args_extension,
const char *args_context,
long args_priority,
const char *args_label,
const char *args_app,
const char *args_app_args,
const char *args_caller_id,
int args_timeout,
struct ast_variable *variables,
const char *args_channel_id,
const char *args_other_channel_id,
const char *args_originator,
struct ast_ari_response *response)
{
char *dialtech;
char dialdevice[AST_CHANNEL_NAME];
struct ast_dial *dial;
char *caller_id = NULL;
char *cid_num = NULL;
char *cid_name = NULL;
char *stuff;
struct ast_channel *other = NULL;
struct ast_channel *chan = NULL;
RAII_VAR(struct ast_channel_snapshot *, snapshot, NULL, ao2_cleanup);
struct ast_assigned_ids assignedids = {
.uniqueid = args_channel_id,
.uniqueid2 = args_other_channel_id,
};
struct ari_origination *origination;
pthread_t thread;
if ((assignedids.uniqueid && AST_MAX_PUBLIC_UNIQUEID < strlen(assignedids.uniqueid))
|| (assignedids.uniqueid2 && AST_MAX_PUBLIC_UNIQUEID < strlen(assignedids.uniqueid2))) {
ast_ari_response_error(response, 400, "Bad Request",
"Uniqueid length exceeds maximum of %d", AST_MAX_PUBLIC_UNIQUEID);
return;
}
if (ast_strlen_zero(args_endpoint)) {
ast_ari_response_error(response, 400, "Bad Request",
"Endpoint must be specified");
return;
}
dialtech = ast_strdupa(args_endpoint);
if ((stuff = strchr(dialtech, '/'))) {
*stuff++ = '\0';
ast_copy_string(dialdevice, stuff, sizeof(dialdevice));
}
if (ast_strlen_zero(dialtech) || ast_strlen_zero(dialdevice)) {
ast_ari_response_error(response, 400, "Bad Request",
"Invalid endpoint specified");
return;
}
if (!ast_strlen_zero(args_app)) {
RAII_VAR(struct ast_str *, appdata, ast_str_create(64), ast_free);
if (!appdata) {
ast_ari_response_alloc_failed(response);
return;
}
ast_str_set(&appdata, 0, "%s", args_app);
if (!ast_strlen_zero(args_app_args)) {
ast_str_append(&appdata, 0, ",%s", args_app_args);
}
origination = ast_calloc(1, sizeof(*origination) + ast_str_size(appdata) + 1);
if (!origination) {
ast_ari_response_alloc_failed(response);
return;
}
strcpy(origination->appdata, ast_str_buffer(appdata));
} else if (!ast_strlen_zero(args_extension)) {
origination = ast_calloc(1, sizeof(*origination) + 1);
if (!origination) {
ast_ari_response_alloc_failed(response);
return;
}
ast_copy_string(origination->context, S_OR(args_context, "default"), sizeof(origination->context));
ast_copy_string(origination->exten, args_extension, sizeof(origination->exten));
if (!ast_strlen_zero(args_label)) {
/* A label was provided in the request, use that */
int ipri = 1;
if (sscanf(args_label, "%30d", &ipri) != 1) {
ipri = ast_findlabel_extension(chan, origination->context, origination->exten, args_label, args_caller_id);
if (ipri == -1) {
ast_log(AST_LOG_ERROR, "Requested label: %s can not be found in context: %s\n", args_label, args_context);
ast_ari_response_error(response, 404, "Not Found", "Requested label can not be found");
return;
}
} else {
ast_debug(3, "Numeric value provided for label, jumping to that priority\n");
}
if (ipri == 0) {
ast_log(AST_LOG_ERROR, "Invalid priority label '%s' specified for extension %s in context: %s\n",
args_label, args_extension, args_context);
ast_ari_response_error(response, 400, "Bad Request", "Requested priority is illegal");
return;
}
/* Our priority was provided by a label */
origination->priority = ipri;
} else {
/* No label provided, use provided priority */
origination->priority = args_priority ? args_priority : 1;
}
origination->appdata[0] = '\0';
} else {
ast_ari_response_error(response, 400, "Bad Request",
"Application or extension must be specified");
return;
}
dial = ast_dial_create();
if (!dial) {
ast_ari_response_alloc_failed(response);
ast_free(origination);
return;
}
ast_dial_set_user_data(dial, origination);
if (ast_dial_append(dial, dialtech, dialdevice, &assignedids)) {
ast_ari_response_alloc_failed(response);
ast_dial_destroy(dial);
ast_free(origination);
return;
}
if (args_timeout > 0) {
ast_dial_set_global_timeout(dial, args_timeout * 1000);
} else if (args_timeout == -1) {
ast_dial_set_global_timeout(dial, -1);
} else {
ast_dial_set_global_timeout(dial, 30000);
}
if (!ast_strlen_zero(args_caller_id)) {
caller_id = ast_strdupa(args_caller_id);
ast_callerid_parse(caller_id, &cid_name, &cid_num);
if (ast_is_shrinkable_phonenumber(cid_num)) {
ast_shrink_phone_number(cid_num);
}
}
if (!ast_strlen_zero(args_originator)) {
other = ast_channel_get_by_name(args_originator);
if (!other) {
ast_ari_response_error(
response, 400, "Bad Request",
"Provided originator channel was not found");
ast_dial_destroy(dial);
ast_free(origination);
return;
}
}
ARI/PJSIP: Apply requesting channel's format cap to created channels This patch addresses the following problems: * ari/resource_channels: In ARI, we currently create a format capability structure of SLIN and apply it to the new channel being created. This was originally done when the PBX core was used to create the channel, as there was a condition where a newly created channel could be created without any formats. Unfortunately, now that the Dial API is being used, this has two drawbacks: (a) SLIN, while it will ensure audio will flows, can cause a lot of needless transcodings to occur, particularly when a Local channel is created to the dialplan. When no format capabilities are available, the Dial API handles this better by handing all audio formats to the requsted channels. As such, we defer to that API to provide the format capabilities. (b) If a channel (requester) is causing this channel to be created, we currently don't use its format capabilities as we are passing in our own. However, the Dial API will use the requester channel's formats if none are passed into it, and the requester channel exists and has format capabilities. This is the "best" scenario, as it is the most likely to create a media path that minimizes transcoding. Fixing this simply entails removing the providing of the format capabilities structure to the Dial API. * chan_pjsip: Rather than blindly picking the first format in the format capability structure - which actually *can* be a video or text format - we select an audio format, and only pick the first format if that fails. That minimizes the weird scenario where we attempt to transcode between video/audio. * res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure. Since ast_request already limits us down to one format capability once the format capabilities are passed along, there's no reason to squelch it here. * channel: Fixed a comment. The reason we have to minimize our requested format capabilities down to a single format is due to Asterisk's inability to convey the format to be used back "up" a channel chain. Consider the following: PJSIP/A => L;1 <=> L;2 => PJSIP/B g,u,a g,u,a g,u,a u That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local channel has inherited those format capabilities down the line; PJSIP/B supports only ulaw. According to these format capabilities, ulaw is acceptable and should be selected across all the channels, and no transcoding should occur. However, there is no way to convey this: when L;2 and PJSIP/B are put into a bridge, we will select ulaw, but that is not conveyed to PJSIP/A and L;1. Thus, we end up with: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B g g X u u Which causes g722 to be written to PJSIP/B. Even if we can convey the 'ulaw' choice back up the chain (which through some severe hacking in Local channels was accomplished), such that the chain looks like: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B u u u u We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back with only 'ulaw'. This results in all the channel structures being set up correctly, but PJSIP/A *still* sending g722 and causing the chain to fall apart. There's a lot of difficulty just in setting this up, as there are numerous race conditions in the act of bridging, and no clean mechanism to pass the selected format backwards down an established channel chain. As such, the best that can be done at this point in time is clarifying the comment. Review: https://reviewboard.asterisk.org/r/4434/ ASTERISK-24812 #close Reported by: Matt Jordan ........ Merged revisions 432195 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-24 22:00:51 +00:00
if (ast_dial_prerun(dial, other, NULL)) {
ast_ari_response_alloc_failed(response);
ast_dial_destroy(dial);
ast_free(origination);
ast_channel_cleanup(other);
return;
}
ast_channel_cleanup(other);
chan = ast_dial_get_channel(dial, 0);
if (!chan) {
ast_ari_response_alloc_failed(response);
ast_dial_destroy(dial);
ast_free(origination);
return;
}
if (!ast_strlen_zero(cid_num) || !ast_strlen_zero(cid_name)) {
struct ast_party_connected_line connected;
/*
* It seems strange to set the CallerID on an outgoing call leg
* to whom we are calling, but this function's callers are doing
* various Originate methods. This call leg goes to the local
* user. Once the called party answers, the dialplan needs to
* be able to access the CallerID from the CALLERID function as
* if the called party had placed this call.
*/
ast_set_callerid(chan, cid_num, cid_name, cid_num);
ast_party_connected_line_set_init(&connected, ast_channel_connected(chan));
if (!ast_strlen_zero(cid_num)) {
connected.id.number.valid = 1;
connected.id.number.str = (char *) cid_num;
connected.id.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
}
if (!ast_strlen_zero(cid_name)) {
connected.id.name.valid = 1;
connected.id.name.str = (char *) cid_name;
connected.id.name.presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
}
ast_channel_set_connected_line(chan, &connected, NULL);
}
ast_channel_lock(chan);
if (variables) {
ast_set_variables(chan, variables);
}
ast_set_flag(ast_channel_flags(chan), AST_FLAG_ORIGINATED);
if (!ast_strlen_zero(args_app)) {
struct ast_channel *local_peer;
ARI/res_stasis: Subscribe to both Local channel halves when originating to app This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close ........ Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07 02:15:00 +00:00
stasis_app_subscribe_channel(args_app, chan);
/* Subscribe to the Local channel peer also. */
local_peer = ast_local_get_peer(chan);
ARI/res_stasis: Subscribe to both Local channel halves when originating to app This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close ........ Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07 02:15:00 +00:00
if (local_peer) {
stasis_app_subscribe_channel(args_app, local_peer);
ast_channel_unref(local_peer);
ARI/res_stasis: Subscribe to both Local channel halves when originating to app This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close ........ Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07 02:15:00 +00:00
}
}
ARI/res_stasis: Subscribe to both Local channel halves when originating to app This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close ........ Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-07 02:15:00 +00:00
snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(chan));
ast_channel_unlock(chan);
/* Before starting the async dial bump the ref in case the dial quickly goes away and takes
* the reference with it
*/
ast_channel_ref(chan);
if (ast_pthread_create_detached(&thread, NULL, ari_originate_dial, dial)) {
ast_ari_response_alloc_failed(response);
ast_dial_destroy(dial);
ast_free(origination);
} else {
ast_ari_response_ok(response, ast_channel_snapshot_to_json(snapshot, NULL));
}
ast_channel_unref(chan);
return;
}
/*!
* \internal
* \brief Convert a \c ast_json list of key/value pair tuples into a \c ast_variable list
* \since 13.3.0
*
* \param[out] response HTTP response if error
* \param json_variables The JSON blob containing the variable
* \param[out] variables An out reference to the variables to populate.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int json_to_ast_variables(struct ast_ari_response *response, struct ast_json *json_variables, struct ast_variable **variables)
{
enum ast_json_to_ast_vars_code res;
res = ast_json_to_ast_variables(json_variables, variables);
switch (res) {
case AST_JSON_TO_AST_VARS_CODE_SUCCESS:
return 0;
case AST_JSON_TO_AST_VARS_CODE_INVALID_TYPE:
ast_ari_response_error(response, 400, "Bad Request",
"Only string values in the 'variables' object allowed");
break;
case AST_JSON_TO_AST_VARS_CODE_OOM:
ast_ari_response_alloc_failed(response);
break;
}
ast_log(AST_LOG_ERROR, "Unable to convert 'variables' in JSON body to channel variables\n");
return -1;
}
void ast_ari_channels_originate_with_id(struct ast_variable *headers,
struct ast_ari_channels_originate_with_id_args *args,
struct ast_ari_response *response)
{
struct ast_variable *variables = NULL;
/* Parse any query parameters out of the body parameter */
if (args->variables) {
struct ast_json *json_variables;
ast_ari_channels_originate_with_id_parse_body(args->variables, args);
json_variables = ast_json_object_get(args->variables, "variables");
if (json_variables
&& json_to_ast_variables(response, json_variables, &variables)) {
return;
}
}
ari_channels_handle_originate_with_id(
args->endpoint,
args->extension,
args->context,
args->priority,
args->label,
args->app,
args->app_args,
args->caller_id,
args->timeout,
variables,
args->channel_id,
args->other_channel_id,
args->originator,
response);
ast_variables_destroy(variables);
}
void ast_ari_channels_originate(struct ast_variable *headers,
struct ast_ari_channels_originate_args *args,
struct ast_ari_response *response)
{
struct ast_variable *variables = NULL;
/* Parse any query parameters out of the body parameter */
if (args->variables) {
struct ast_json *json_variables;
ast_ari_channels_originate_parse_body(args->variables, args);
json_variables = ast_json_object_get(args->variables, "variables");
if (json_variables
&& json_to_ast_variables(response, json_variables, &variables)) {
return;
}
}
ari_channels_handle_originate_with_id(
args->endpoint,
args->extension,
args->context,
args->priority,
args->label,
args->app,
args->app_args,
args->caller_id,
args->timeout,
variables,
args->channel_id,
args->other_channel_id,
args->originator,
response);
ast_variables_destroy(variables);
}
void ast_ari_channels_get_channel_var(struct ast_variable *headers,
struct ast_ari_channels_get_channel_var_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct ast_json *, json, NULL, ast_json_unref);
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
RAII_VAR(struct ast_str *, value, ast_str_create(32), ast_free);
RAII_VAR(struct ast_channel *, channel, NULL, ast_channel_cleanup);
ast_assert(response != NULL);
if (!value) {
ast_ari_response_alloc_failed(response);
return;
}
if (ast_strlen_zero(args->variable)) {
ast_ari_response_error(
response, 400, "Bad Request",
"Variable name is required");
return;
}
if (ast_strlen_zero(args->channel_id)) {
ast_ari_response_error(
response, 400, "Bad Request",
"Channel ID is required");
return;
}
channel = ast_channel_get_by_name(args->channel_id);
if (!channel) {
ast_ari_response_error(
response, 404, "Channel Not Found",
"Provided channel was not found");
return;
}
/* You may be tempted to lock the channel you're about to read from. You
* would be wrong. Some dialplan functions put the channel into
* autoservice, which deadlocks if the channel is already locked.
* ast_str_retrieve_variable() does its own locking, and the dialplan
* functions need to as well. We should be fine without the lock.
*/
if (args->variable[strlen(args->variable) - 1] == ')') {
if (ast_func_read2(channel, args->variable, &value, 0)) {
ast_ari_response_error(
response, 500, "Error With Function",
"Unable to read provided function");
return;
}
} else {
if (!ast_str_retrieve_variable(&value, 0, channel, NULL, args->variable)) {
ast_ari_response_error(
response, 404, "Variable Not Found",
"Provided variable was not found");
return;
}
}
if (!(json = ast_json_pack("{s: s}", "value", S_OR(ast_str_buffer(value), "")))) {
ast_ari_response_alloc_failed(response);
return;
}
ast_ari_response_ok(response, ast_json_ref(json));
}
void ast_ari_channels_set_channel_var(struct ast_variable *headers,
struct ast_ari_channels_set_channel_var_args *args,
struct ast_ari_response *response)
{
RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
ast_assert(response != NULL);
if (ast_strlen_zero(args->variable)) {
ast_ari_response_error(
response, 400, "Bad Request",
"Variable name is required");
return;
}
control = find_control(response, args->channel_id);
if (control == NULL) {
/* response filled in by find_control */
return;
}
if (stasis_app_control_set_channel_var(control, args->variable, args->value)) {
ast_ari_response_error(
response, 400, "Bad Request",
"Failed to execute function");
return;
}
ast_ari_response_no_content(response);
}
static void ari_channels_handle_snoop_channel(
const char *args_channel_id,
const char *args_spy,
const char *args_whisper,
const char *args_app,
const char *args_app_args,
const char *args_snoop_id,
struct ast_ari_response *response)
{
enum stasis_app_snoop_direction spy, whisper;
RAII_VAR(struct ast_channel *, chan, NULL, ast_channel_cleanup);
RAII_VAR(struct ast_channel *, snoop, NULL, ast_channel_cleanup);
RAII_VAR(struct ast_channel_snapshot *, snapshot, NULL, ao2_cleanup);
ast_assert(response != NULL);
if (ast_strlen_zero(args_spy) || !strcmp(args_spy, "none")) {
spy = STASIS_SNOOP_DIRECTION_NONE;
} else if (!strcmp(args_spy, "both")) {
spy = STASIS_SNOOP_DIRECTION_BOTH;
} else if (!strcmp(args_spy, "out")) {
spy = STASIS_SNOOP_DIRECTION_OUT;
} else if (!strcmp(args_spy, "in")) {
spy = STASIS_SNOOP_DIRECTION_IN;
} else {
ast_ari_response_error(
response, 400, "Bad Request",
"Invalid direction specified for spy");
return;
}
if (ast_strlen_zero(args_whisper) || !strcmp(args_whisper, "none")) {
whisper = STASIS_SNOOP_DIRECTION_NONE;
} else if (!strcmp(args_whisper, "both")) {
whisper = STASIS_SNOOP_DIRECTION_BOTH;
} else if (!strcmp(args_whisper, "out")) {
whisper = STASIS_SNOOP_DIRECTION_OUT;
} else if (!strcmp(args_whisper, "in")) {
whisper = STASIS_SNOOP_DIRECTION_IN;
} else {
ast_ari_response_error(
response, 400, "Bad Request",
"Invalid direction specified for whisper");
return;
}
if (spy == STASIS_SNOOP_DIRECTION_NONE && whisper == STASIS_SNOOP_DIRECTION_NONE) {
ast_ari_response_error(
response, 400, "Bad Request",
"Direction must be specified for at least spy or whisper");
return;
} else if (ast_strlen_zero(args_app)) {
ast_ari_response_error(
response, 400, "Bad Request",
"Application name is required");
return;
}
chan = ast_channel_get_by_name(args_channel_id);
if (chan == NULL) {
ast_ari_response_error(
response, 404, "Channel Not Found",
"Provided channel was not found");
return;
}
snoop = stasis_app_control_snoop(chan, spy, whisper, args_app, args_app_args,
args_snoop_id);
if (snoop == NULL) {
ast_ari_response_error(
response, 500, "Internal error",
"Snoop channel could not be created");
return;
}
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(snoop));
ast_ari_response_ok(response, ast_channel_snapshot_to_json(snapshot, NULL));
}
void ast_ari_channels_snoop_channel(struct ast_variable *headers,
struct ast_ari_channels_snoop_channel_args *args,
struct ast_ari_response *response)
{
ari_channels_handle_snoop_channel(
args->channel_id,
args->spy,
args->whisper,
args->app,
args->app_args,
args->snoop_id,
response);
}
void ast_ari_channels_snoop_channel_with_id(struct ast_variable *headers,
struct ast_ari_channels_snoop_channel_with_id_args *args,
struct ast_ari_response *response)
{
ari_channels_handle_snoop_channel(
args->channel_id,
args->spy,
args->whisper,
args->app,
args->app_args,
args->snoop_id,
response);
}