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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
1311 lines
40 KiB
C
1311 lines
40 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2007, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Dialing API
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*
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* \author Joshua Colp <jcolp@digium.com>
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <sys/time.h>
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#include <signal.h>
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#include "asterisk/channel.h"
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#include "asterisk/utils.h"
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#include "asterisk/lock.h"
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#include "asterisk/linkedlists.h"
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#include "asterisk/dial.h"
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#include "asterisk/pbx.h"
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#include "asterisk/musiconhold.h"
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#include "asterisk/app.h"
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#include "asterisk/causes.h"
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#include "asterisk/stasis_channels.h"
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/*! \brief Main dialing structure. Contains global options, channels being dialed, and more! */
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struct ast_dial {
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int num; /*!< Current number to give to next dialed channel */
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int timeout; /*!< Maximum time allowed for dial attempts */
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int actual_timeout; /*!< Actual timeout based on all factors (ie: channels) */
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enum ast_dial_result state; /*!< Status of dial */
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void *options[AST_DIAL_OPTION_MAX]; /*!< Global options */
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ast_dial_state_callback state_callback; /*!< Status callback */
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void *user_data; /*!< Attached user data */
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AST_LIST_HEAD(, ast_dial_channel) channels; /*!< Channels being dialed */
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pthread_t thread; /*!< Thread (if running in async) */
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struct ast_callid *callid; /*!< callid pointer (if running in async) */
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ast_mutex_t lock; /*! Lock to protect the thread information above */
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};
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/*! \brief Dialing channel structure. Contains per-channel dialing options, asterisk channel, and more! */
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struct ast_dial_channel {
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int num; /*!< Unique number for dialed channel */
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int timeout; /*!< Maximum time allowed for attempt */
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char *tech; /*!< Technology being dialed */
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char *device; /*!< Device being dialed */
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void *options[AST_DIAL_OPTION_MAX]; /*!< Channel specific options */
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int cause; /*!< Cause code in case of failure */
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unsigned int is_running_app:1; /*!< Is this running an application? */
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char *assignedid1; /*!< UniqueID to assign channel */
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char *assignedid2; /*!< UniqueID to assign 2nd channel */
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struct ast_channel *owner; /*!< Asterisk channel */
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AST_LIST_ENTRY(ast_dial_channel) list; /*!< Linked list information */
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};
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/*! \brief Typedef for dial option enable */
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typedef void *(*ast_dial_option_cb_enable)(void *data);
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/*! \brief Typedef for dial option disable */
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typedef int (*ast_dial_option_cb_disable)(void *data);
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/*! \brief Structure for 'ANSWER_EXEC' option */
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struct answer_exec_struct {
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char app[AST_MAX_APP]; /*!< Application name */
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char *args; /*!< Application arguments */
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};
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/*! \brief Enable function for 'ANSWER_EXEC' option */
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static void *answer_exec_enable(void *data)
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{
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struct answer_exec_struct *answer_exec = NULL;
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char *app = ast_strdupa((char*)data), *args = NULL;
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/* Not giving any data to this option is bad, mmmk? */
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if (ast_strlen_zero(app))
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return NULL;
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/* Create new data structure */
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if (!(answer_exec = ast_calloc(1, sizeof(*answer_exec))))
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return NULL;
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/* Parse out application and arguments */
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if ((args = strchr(app, ','))) {
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*args++ = '\0';
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answer_exec->args = ast_strdup(args);
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}
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/* Copy application name */
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ast_copy_string(answer_exec->app, app, sizeof(answer_exec->app));
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return answer_exec;
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}
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/*! \brief Disable function for 'ANSWER_EXEC' option */
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static int answer_exec_disable(void *data)
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{
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struct answer_exec_struct *answer_exec = data;
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/* Make sure we have a value */
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if (!answer_exec)
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return -1;
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/* If arguments are present, free them too */
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if (answer_exec->args)
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ast_free(answer_exec->args);
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/* This is simple - just free the structure */
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ast_free(answer_exec);
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return 0;
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}
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static void *music_enable(void *data)
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{
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return ast_strdup(data);
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}
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static int music_disable(void *data)
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{
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if (!data)
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return -1;
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ast_free(data);
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return 0;
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}
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static void *predial_enable(void *data)
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{
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return ast_strdup(data);
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}
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static int predial_disable(void *data)
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{
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if (!data) {
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return -1;
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}
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ast_free(data);
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return 0;
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}
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/*! \brief Application execution function for 'ANSWER_EXEC' option */
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static void answer_exec_run(struct ast_dial *dial, struct ast_dial_channel *dial_channel, char *app, char *args)
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{
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struct ast_channel *chan = dial_channel->owner;
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struct ast_app *ast_app = pbx_findapp(app);
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/* If the application was not found, return immediately */
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if (!ast_app)
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return;
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/* All is well... execute the application */
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pbx_exec(chan, ast_app, args);
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/* If another thread is not taking over hang up the channel */
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ast_mutex_lock(&dial->lock);
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if (dial->thread != AST_PTHREADT_STOP) {
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ast_hangup(chan);
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dial_channel->owner = NULL;
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}
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ast_mutex_unlock(&dial->lock);
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return;
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}
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struct ast_option_types {
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enum ast_dial_option option;
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ast_dial_option_cb_enable enable;
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ast_dial_option_cb_disable disable;
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};
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/*!
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* \brief Map options to respective handlers (enable/disable).
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*
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* \note This list MUST be perfectly kept in order with enum
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* ast_dial_option, or else madness will happen.
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*/
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static const struct ast_option_types option_types[] = {
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{ AST_DIAL_OPTION_RINGING, NULL, NULL }, /*!< Always indicate ringing to caller */
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{ AST_DIAL_OPTION_ANSWER_EXEC, answer_exec_enable, answer_exec_disable }, /*!< Execute application upon answer in async mode */
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{ AST_DIAL_OPTION_MUSIC, music_enable, music_disable }, /*!< Play music to the caller instead of ringing */
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{ AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL, NULL }, /*!< Disable call forwarding on channels */
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{ AST_DIAL_OPTION_PREDIAL, predial_enable, predial_disable }, /*!< Execute a subroutine on the outbound channels prior to dialing */
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{ AST_DIAL_OPTION_MAX, NULL, NULL }, /*!< Terminator of list */
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};
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/*! \brief Maximum number of channels we can watch at a time */
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#define AST_MAX_WATCHERS 256
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/*! \brief Macro for finding the option structure to use on a dialed channel */
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#define FIND_RELATIVE_OPTION(dial, dial_channel, ast_dial_option) (dial_channel->options[ast_dial_option] ? dial_channel->options[ast_dial_option] : dial->options[ast_dial_option])
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/*! \brief Macro that determines whether a channel is the caller or not */
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#define IS_CALLER(chan, owner) (chan == owner ? 1 : 0)
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/*! \brief New dialing structure
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* \note Create a dialing structure
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* \return Returns a calloc'd ast_dial structure, NULL on failure
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*/
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struct ast_dial *ast_dial_create(void)
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{
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struct ast_dial *dial = NULL;
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/* Allocate new memory for structure */
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if (!(dial = ast_calloc(1, sizeof(*dial))))
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return NULL;
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/* Initialize list of channels */
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AST_LIST_HEAD_INIT(&dial->channels);
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/* Initialize thread to NULL */
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dial->thread = AST_PTHREADT_NULL;
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/* No timeout exists... yet */
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dial->timeout = -1;
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dial->actual_timeout = -1;
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/* Can't forget about the lock */
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ast_mutex_init(&dial->lock);
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return dial;
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}
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/*! \brief Append a channel
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* \note Appends a channel to a dialing structure
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* \return Returns channel reference number on success, -1 on failure
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*/
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int ast_dial_append(struct ast_dial *dial, const char *tech, const char *device, const struct ast_assigned_ids *assignedids)
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{
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struct ast_dial_channel *channel = NULL;
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/* Make sure we have required arguments */
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if (!dial || !tech || !device)
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return -1;
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/* Allocate new memory for dialed channel structure */
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if (!(channel = ast_calloc(1, sizeof(*channel))))
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return -1;
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/* Record technology and device for when we actually dial */
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channel->tech = ast_strdup(tech);
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channel->device = ast_strdup(device);
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/* Store the assigned id */
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if (assignedids && !ast_strlen_zero(assignedids->uniqueid)) {
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channel->assignedid1 = ast_strdup(assignedids->uniqueid);
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if (!ast_strlen_zero(assignedids->uniqueid2)) {
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channel->assignedid2 = ast_strdup(assignedids->uniqueid2);
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}
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}
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/* Grab reference number from dial structure */
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channel->num = ast_atomic_fetchadd_int(&dial->num, +1);
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/* No timeout exists... yet */
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channel->timeout = -1;
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/* Insert into channels list */
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AST_LIST_INSERT_TAIL(&dial->channels, channel, list);
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return channel->num;
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}
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/*! \brief Helper function that requests all channels */
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static int begin_dial_prerun(struct ast_dial_channel *channel, struct ast_channel *chan, struct ast_format_cap *cap, const char *predial_string)
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{
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char numsubst[AST_MAX_EXTENSION];
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struct ast_format_cap *cap_all_audio = NULL;
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struct ast_format_cap *cap_request;
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struct ast_assigned_ids assignedids = {
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.uniqueid = channel->assignedid1,
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.uniqueid2 = channel->assignedid2,
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};
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/* Copy device string over */
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ast_copy_string(numsubst, channel->device, sizeof(numsubst));
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if (ast_format_cap_count(cap)) {
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cap_request = cap;
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} else if (chan) {
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cap_request = ast_channel_nativeformats(chan);
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} else {
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cap_all_audio = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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ast_format_cap_append_by_type(cap_all_audio, AST_MEDIA_TYPE_AUDIO);
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cap_request = cap_all_audio;
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}
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/* If we fail to create our owner channel bail out */
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if (!(channel->owner = ast_request(channel->tech, cap_request, &assignedids, chan, numsubst, &channel->cause))) {
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ao2_cleanup(cap_all_audio);
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return -1;
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}
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cap_request = NULL;
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ao2_cleanup(cap_all_audio);
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ast_channel_lock(channel->owner);
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ast_channel_stage_snapshot(channel->owner);
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|
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ast_channel_appl_set(channel->owner, "AppDial2");
|
|
ast_channel_data_set(channel->owner, "(Outgoing Line)");
|
|
|
|
memset(ast_channel_whentohangup(channel->owner), 0, sizeof(*ast_channel_whentohangup(channel->owner)));
|
|
|
|
/* Inherit everything from he who spawned this dial */
|
|
if (chan) {
|
|
ast_channel_inherit_variables(chan, channel->owner);
|
|
ast_channel_datastore_inherit(chan, channel->owner);
|
|
|
|
/* Copy over callerid information */
|
|
ast_party_redirecting_copy(ast_channel_redirecting(channel->owner), ast_channel_redirecting(chan));
|
|
|
|
ast_channel_dialed(channel->owner)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
|
|
|
|
ast_connected_line_copy_from_caller(ast_channel_connected(channel->owner), ast_channel_caller(chan));
|
|
|
|
ast_channel_language_set(channel->owner, ast_channel_language(chan));
|
|
ast_channel_accountcode_set(channel->owner, ast_channel_accountcode(chan));
|
|
if (ast_strlen_zero(ast_channel_musicclass(channel->owner)))
|
|
ast_channel_musicclass_set(channel->owner, ast_channel_musicclass(chan));
|
|
|
|
ast_channel_adsicpe_set(channel->owner, ast_channel_adsicpe(chan));
|
|
ast_channel_transfercapability_set(channel->owner, ast_channel_transfercapability(chan));
|
|
}
|
|
|
|
ast_channel_stage_snapshot_done(channel->owner);
|
|
ast_channel_unlock(channel->owner);
|
|
|
|
if (!ast_strlen_zero(predial_string)) {
|
|
const char *predial_callee = ast_app_expand_sub_args(chan, predial_string);
|
|
if (!predial_callee) {
|
|
ast_log(LOG_ERROR, "Could not expand subroutine arguments in predial request '%s'\n", predial_string);
|
|
}
|
|
ast_autoservice_start(chan);
|
|
ast_pre_call(channel->owner, predial_callee);
|
|
ast_autoservice_stop(chan);
|
|
ast_free((char *) predial_callee);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_dial_prerun(struct ast_dial *dial, struct ast_channel *chan, struct ast_format_cap *cap)
|
|
{
|
|
struct ast_dial_channel *channel;
|
|
int res = -1;
|
|
char *predial_string = dial->options[AST_DIAL_OPTION_PREDIAL];
|
|
|
|
if (!ast_strlen_zero(predial_string)) {
|
|
ast_replace_subargument_delimiter(predial_string);
|
|
}
|
|
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
if ((res = begin_dial_prerun(channel, chan, cap, predial_string))) {
|
|
break;
|
|
}
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
return res;
|
|
}
|
|
|
|
/*! \brief Helper function that does the beginning dialing per-appended channel */
|
|
static int begin_dial_channel(struct ast_dial_channel *channel, struct ast_channel *chan, int async, const char *predial_string)
|
|
{
|
|
char numsubst[AST_MAX_EXTENSION];
|
|
int res = 1;
|
|
|
|
/* If no owner channel exists yet execute pre-run */
|
|
if (!channel->owner && begin_dial_prerun(channel, chan, NULL, predial_string)) {
|
|
return 0;
|
|
}
|
|
|
|
/* Copy device string over */
|
|
ast_copy_string(numsubst, channel->device, sizeof(numsubst));
|
|
|
|
/* Attempt to actually call this device */
|
|
if ((res = ast_call(channel->owner, numsubst, 0))) {
|
|
res = 0;
|
|
ast_hangup(channel->owner);
|
|
channel->owner = NULL;
|
|
} else {
|
|
if (chan) {
|
|
ast_poll_channel_add(chan, channel->owner);
|
|
}
|
|
ast_channel_publish_dial(async ? NULL : chan, channel->owner, channel->device, NULL);
|
|
res = 1;
|
|
ast_verb(3, "Called %s\n", numsubst);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/*! \brief Helper function that does the beginning dialing per dial structure */
|
|
static int begin_dial(struct ast_dial *dial, struct ast_channel *chan, int async)
|
|
{
|
|
struct ast_dial_channel *channel = NULL;
|
|
int success = 0;
|
|
char *predial_string = dial->options[AST_DIAL_OPTION_PREDIAL];
|
|
|
|
if (!ast_strlen_zero(predial_string)) {
|
|
ast_replace_subargument_delimiter(predial_string);
|
|
}
|
|
|
|
/* Iterate through channel list, requesting and calling each one */
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
success += begin_dial_channel(channel, chan, async, predial_string);
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
/* If number of failures matches the number of channels, then this truly failed */
|
|
return success;
|
|
}
|
|
|
|
/*! \brief Helper function to handle channels that have been call forwarded */
|
|
static int handle_call_forward(struct ast_dial *dial, struct ast_dial_channel *channel, struct ast_channel *chan)
|
|
{
|
|
struct ast_channel *original = channel->owner;
|
|
char *tmp = ast_strdupa(ast_channel_call_forward(channel->owner));
|
|
char *tech = "Local", *device = tmp, *stuff;
|
|
char *predial_string = dial->options[AST_DIAL_OPTION_PREDIAL];
|
|
|
|
if (!ast_strlen_zero(predial_string)) {
|
|
ast_replace_subargument_delimiter(predial_string);
|
|
}
|
|
|
|
/* If call forwarding is disabled just drop the original channel and don't attempt to dial the new one */
|
|
if (FIND_RELATIVE_OPTION(dial, channel, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING)) {
|
|
ast_hangup(original);
|
|
channel->owner = NULL;
|
|
return 0;
|
|
}
|
|
|
|
/* Figure out the new destination */
|
|
if ((stuff = strchr(tmp, '/'))) {
|
|
*stuff++ = '\0';
|
|
tech = tmp;
|
|
device = stuff;
|
|
} else {
|
|
const char *forward_context;
|
|
char destination[AST_MAX_CONTEXT + AST_MAX_EXTENSION + 1];
|
|
|
|
ast_channel_lock(original);
|
|
forward_context = pbx_builtin_getvar_helper(original, "FORWARD_CONTEXT");
|
|
snprintf(destination, sizeof(destination), "%s@%s", tmp, S_OR(forward_context, ast_channel_context(original)));
|
|
ast_channel_unlock(original);
|
|
device = ast_strdupa(destination);
|
|
}
|
|
|
|
/* Drop old destination information */
|
|
ast_free(channel->tech);
|
|
ast_free(channel->device);
|
|
ast_free(channel->assignedid1);
|
|
channel->assignedid1 = NULL;
|
|
ast_free(channel->assignedid2);
|
|
channel->assignedid2 = NULL;
|
|
|
|
/* Update the dial channel with the new destination information */
|
|
channel->tech = ast_strdup(tech);
|
|
channel->device = ast_strdup(device);
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
/* Drop the original channel */
|
|
channel->owner = NULL;
|
|
|
|
/* Finally give it a go... send it out into the world */
|
|
begin_dial_channel(channel, chan, chan ? 0 : 1, predial_string);
|
|
|
|
ast_channel_publish_dial_forward(chan, original, channel->owner, NULL, "CANCEL",
|
|
ast_channel_call_forward(original));
|
|
|
|
ast_hangup(original);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Helper function that finds the dialed channel based on owner */
|
|
static struct ast_dial_channel *find_relative_dial_channel(struct ast_dial *dial, struct ast_channel *owner)
|
|
{
|
|
struct ast_dial_channel *channel = NULL;
|
|
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
if (channel->owner == owner)
|
|
break;
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
return channel;
|
|
}
|
|
|
|
static void set_state(struct ast_dial *dial, enum ast_dial_result state)
|
|
{
|
|
dial->state = state;
|
|
|
|
if (dial->state_callback)
|
|
dial->state_callback(dial);
|
|
}
|
|
|
|
/*! \brief Helper function that handles control frames WITH owner */
|
|
static void handle_frame(struct ast_dial *dial, struct ast_dial_channel *channel, struct ast_frame *fr, struct ast_channel *chan)
|
|
{
|
|
if (fr->frametype == AST_FRAME_CONTROL) {
|
|
switch (fr->subclass.integer) {
|
|
case AST_CONTROL_ANSWER:
|
|
ast_verb(3, "%s answered %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_REMOVE(&dial->channels, channel, list);
|
|
AST_LIST_INSERT_HEAD(&dial->channels, channel, list);
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
ast_channel_publish_dial(chan, channel->owner, channel->device, "ANSWER");
|
|
set_state(dial, AST_DIAL_RESULT_ANSWERED);
|
|
break;
|
|
case AST_CONTROL_BUSY:
|
|
ast_verb(3, "%s is busy\n", ast_channel_name(channel->owner));
|
|
ast_channel_publish_dial(chan, channel->owner, channel->device, "BUSY");
|
|
ast_hangup(channel->owner);
|
|
channel->cause = AST_CAUSE_USER_BUSY;
|
|
channel->owner = NULL;
|
|
break;
|
|
case AST_CONTROL_CONGESTION:
|
|
ast_verb(3, "%s is circuit-busy\n", ast_channel_name(channel->owner));
|
|
ast_channel_publish_dial(chan, channel->owner, channel->device, "CONGESTION");
|
|
ast_hangup(channel->owner);
|
|
channel->cause = AST_CAUSE_NORMAL_CIRCUIT_CONGESTION;
|
|
channel->owner = NULL;
|
|
break;
|
|
case AST_CONTROL_INCOMPLETE:
|
|
ast_verb(3, "%s dialed Incomplete extension %s\n", ast_channel_name(channel->owner), ast_channel_exten(channel->owner));
|
|
ast_indicate(chan, AST_CONTROL_INCOMPLETE);
|
|
break;
|
|
case AST_CONTROL_RINGING:
|
|
ast_verb(3, "%s is ringing\n", ast_channel_name(channel->owner));
|
|
if (!dial->options[AST_DIAL_OPTION_MUSIC])
|
|
ast_indicate(chan, AST_CONTROL_RINGING);
|
|
set_state(dial, AST_DIAL_RESULT_RINGING);
|
|
break;
|
|
case AST_CONTROL_PROGRESS:
|
|
ast_verb(3, "%s is making progress, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
|
|
ast_indicate(chan, AST_CONTROL_PROGRESS);
|
|
set_state(dial, AST_DIAL_RESULT_PROGRESS);
|
|
break;
|
|
case AST_CONTROL_VIDUPDATE:
|
|
ast_verb(3, "%s requested a video update, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
|
|
ast_indicate(chan, AST_CONTROL_VIDUPDATE);
|
|
break;
|
|
case AST_CONTROL_SRCUPDATE:
|
|
ast_verb(3, "%s requested a source update, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
|
|
ast_indicate(chan, AST_CONTROL_SRCUPDATE);
|
|
break;
|
|
case AST_CONTROL_CONNECTED_LINE:
|
|
ast_verb(3, "%s connected line has changed, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
|
|
if (ast_channel_connected_line_sub(channel->owner, chan, fr, 1) &&
|
|
ast_channel_connected_line_macro(channel->owner, chan, fr, 1, 1)) {
|
|
ast_indicate_data(chan, AST_CONTROL_CONNECTED_LINE, fr->data.ptr, fr->datalen);
|
|
}
|
|
break;
|
|
case AST_CONTROL_REDIRECTING:
|
|
ast_verb(3, "%s redirecting info has changed, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
|
|
if (ast_channel_redirecting_sub(channel->owner, chan, fr, 1) &&
|
|
ast_channel_redirecting_macro(channel->owner, chan, fr, 1, 1)) {
|
|
ast_indicate_data(chan, AST_CONTROL_REDIRECTING, fr->data.ptr, fr->datalen);
|
|
}
|
|
break;
|
|
case AST_CONTROL_PROCEEDING:
|
|
ast_verb(3, "%s is proceeding, passing it to %s\n", ast_channel_name(channel->owner), ast_channel_name(chan));
|
|
ast_indicate(chan, AST_CONTROL_PROCEEDING);
|
|
set_state(dial, AST_DIAL_RESULT_PROCEEDING);
|
|
break;
|
|
case AST_CONTROL_HOLD:
|
|
ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(chan));
|
|
ast_indicate_data(chan, AST_CONTROL_HOLD, fr->data.ptr, fr->datalen);
|
|
break;
|
|
case AST_CONTROL_UNHOLD:
|
|
ast_verb(3, "Call on %s left from hold\n", ast_channel_name(chan));
|
|
ast_indicate(chan, AST_CONTROL_UNHOLD);
|
|
break;
|
|
case AST_CONTROL_OFFHOOK:
|
|
case AST_CONTROL_FLASH:
|
|
break;
|
|
case AST_CONTROL_PVT_CAUSE_CODE:
|
|
ast_indicate_data(chan, AST_CONTROL_PVT_CAUSE_CODE, fr->data.ptr, fr->datalen);
|
|
break;
|
|
case -1:
|
|
/* Prod the channel */
|
|
ast_indicate(chan, -1);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*! \brief Helper function that handles control frames WITHOUT owner */
|
|
static void handle_frame_ownerless(struct ast_dial *dial, struct ast_dial_channel *channel, struct ast_frame *fr)
|
|
{
|
|
/* If we have no owner we can only update the state of the dial structure, so only look at control frames */
|
|
if (fr->frametype != AST_FRAME_CONTROL)
|
|
return;
|
|
|
|
switch (fr->subclass.integer) {
|
|
case AST_CONTROL_ANSWER:
|
|
ast_verb(3, "%s answered\n", ast_channel_name(channel->owner));
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_REMOVE(&dial->channels, channel, list);
|
|
AST_LIST_INSERT_HEAD(&dial->channels, channel, list);
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
ast_channel_publish_dial(NULL, channel->owner, channel->device, "ANSWER");
|
|
set_state(dial, AST_DIAL_RESULT_ANSWERED);
|
|
break;
|
|
case AST_CONTROL_BUSY:
|
|
ast_verb(3, "%s is busy\n", ast_channel_name(channel->owner));
|
|
ast_channel_publish_dial(NULL, channel->owner, channel->device, "BUSY");
|
|
ast_hangup(channel->owner);
|
|
channel->cause = AST_CAUSE_USER_BUSY;
|
|
channel->owner = NULL;
|
|
break;
|
|
case AST_CONTROL_CONGESTION:
|
|
ast_verb(3, "%s is circuit-busy\n", ast_channel_name(channel->owner));
|
|
ast_channel_publish_dial(NULL, channel->owner, channel->device, "CONGESTION");
|
|
ast_hangup(channel->owner);
|
|
channel->cause = AST_CAUSE_NORMAL_CIRCUIT_CONGESTION;
|
|
channel->owner = NULL;
|
|
break;
|
|
case AST_CONTROL_INCOMPLETE:
|
|
/*
|
|
* Nothing to do but abort the call since we have no
|
|
* controlling channel to ask for more digits.
|
|
*/
|
|
ast_verb(3, "%s dialed Incomplete extension %s\n",
|
|
ast_channel_name(channel->owner), ast_channel_exten(channel->owner));
|
|
ast_hangup(channel->owner);
|
|
channel->cause = AST_CAUSE_UNALLOCATED;
|
|
channel->owner = NULL;
|
|
break;
|
|
case AST_CONTROL_RINGING:
|
|
ast_verb(3, "%s is ringing\n", ast_channel_name(channel->owner));
|
|
set_state(dial, AST_DIAL_RESULT_RINGING);
|
|
break;
|
|
case AST_CONTROL_PROGRESS:
|
|
ast_verb(3, "%s is making progress\n", ast_channel_name(channel->owner));
|
|
set_state(dial, AST_DIAL_RESULT_PROGRESS);
|
|
break;
|
|
case AST_CONTROL_PROCEEDING:
|
|
ast_verb(3, "%s is proceeding\n", ast_channel_name(channel->owner));
|
|
set_state(dial, AST_DIAL_RESULT_PROCEEDING);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
/*! \brief Helper function to handle when a timeout occurs on dialing attempt */
|
|
static int handle_timeout_trip(struct ast_dial *dial, struct timeval start)
|
|
{
|
|
struct ast_dial_channel *channel = NULL;
|
|
int diff = ast_tvdiff_ms(ast_tvnow(), start), lowest_timeout = -1, new_timeout = -1;
|
|
|
|
/* If there is no difference yet return the dial timeout so we can go again, we were likely interrupted */
|
|
if (!diff) {
|
|
return dial->timeout;
|
|
}
|
|
|
|
/* If the global dial timeout tripped switch the state to timeout so our channel loop will drop every channel */
|
|
if (diff >= dial->timeout) {
|
|
set_state(dial, AST_DIAL_RESULT_TIMEOUT);
|
|
new_timeout = 0;
|
|
}
|
|
|
|
/* Go through dropping out channels that have met their timeout */
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
if (dial->state == AST_DIAL_RESULT_TIMEOUT || diff >= channel->timeout) {
|
|
ast_hangup(channel->owner);
|
|
channel->cause = AST_CAUSE_NO_ANSWER;
|
|
channel->owner = NULL;
|
|
} else if ((lowest_timeout == -1) || (lowest_timeout > channel->timeout)) {
|
|
lowest_timeout = channel->timeout;
|
|
}
|
|
}
|
|
|
|
/* Calculate the new timeout using the lowest timeout found */
|
|
if (lowest_timeout >= 0)
|
|
new_timeout = lowest_timeout - diff;
|
|
|
|
return new_timeout;
|
|
}
|
|
|
|
const char *ast_hangup_cause_to_dial_status(int hangup_cause)
|
|
{
|
|
switch(hangup_cause) {
|
|
case AST_CAUSE_BUSY:
|
|
return "BUSY";
|
|
case AST_CAUSE_CONGESTION:
|
|
return "CONGESTION";
|
|
case AST_CAUSE_NO_ROUTE_DESTINATION:
|
|
case AST_CAUSE_UNREGISTERED:
|
|
return "CHANUNAVAIL";
|
|
case AST_CAUSE_NO_ANSWER:
|
|
default:
|
|
return "NOANSWER";
|
|
}
|
|
}
|
|
|
|
/*! \brief Helper function that basically keeps tabs on dialing attempts */
|
|
static enum ast_dial_result monitor_dial(struct ast_dial *dial, struct ast_channel *chan)
|
|
{
|
|
int timeout = -1;
|
|
struct ast_channel *cs[AST_MAX_WATCHERS], *who = NULL;
|
|
struct ast_dial_channel *channel = NULL;
|
|
struct answer_exec_struct *answer_exec = NULL;
|
|
struct timeval start;
|
|
|
|
set_state(dial, AST_DIAL_RESULT_TRYING);
|
|
|
|
/* If the "always indicate ringing" option is set, change state to ringing and indicate to the owner if present */
|
|
if (dial->options[AST_DIAL_OPTION_RINGING]) {
|
|
set_state(dial, AST_DIAL_RESULT_RINGING);
|
|
if (chan)
|
|
ast_indicate(chan, AST_CONTROL_RINGING);
|
|
} else if (chan && dial->options[AST_DIAL_OPTION_MUSIC] &&
|
|
!ast_strlen_zero(dial->options[AST_DIAL_OPTION_MUSIC])) {
|
|
char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
|
|
ast_indicate(chan, -1);
|
|
ast_channel_musicclass_set(chan, dial->options[AST_DIAL_OPTION_MUSIC]);
|
|
ast_moh_start(chan, dial->options[AST_DIAL_OPTION_MUSIC], NULL);
|
|
ast_channel_musicclass_set(chan, original_moh);
|
|
}
|
|
|
|
/* Record start time for timeout purposes */
|
|
start = ast_tvnow();
|
|
|
|
/* We actually figured out the maximum timeout we can do as they were added, so we can directly access the info */
|
|
timeout = dial->actual_timeout;
|
|
|
|
/* Go into an infinite loop while we are trying */
|
|
while ((dial->state != AST_DIAL_RESULT_UNANSWERED) && (dial->state != AST_DIAL_RESULT_ANSWERED) && (dial->state != AST_DIAL_RESULT_HANGUP) && (dial->state != AST_DIAL_RESULT_TIMEOUT)) {
|
|
int pos = 0, count = 0;
|
|
struct ast_frame *fr = NULL;
|
|
|
|
/* Set up channel structure array */
|
|
pos = count = 0;
|
|
if (chan)
|
|
cs[pos++] = chan;
|
|
|
|
/* Add channels we are attempting to dial */
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
if (channel->owner) {
|
|
cs[pos++] = channel->owner;
|
|
count++;
|
|
}
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
/* If we have no outbound channels in progress, switch state to unanswered and stop */
|
|
if (!count) {
|
|
set_state(dial, AST_DIAL_RESULT_UNANSWERED);
|
|
break;
|
|
}
|
|
|
|
/* Just to be safe... */
|
|
if (dial->thread == AST_PTHREADT_STOP)
|
|
break;
|
|
|
|
/* Wait for frames from channels */
|
|
who = ast_waitfor_n(cs, pos, &timeout);
|
|
|
|
/* Check to see if our thread is being canceled */
|
|
if (dial->thread == AST_PTHREADT_STOP)
|
|
break;
|
|
|
|
/* If the timeout no longer exists OR if we got no channel it basically means the timeout was tripped, so handle it */
|
|
if (!timeout || !who) {
|
|
timeout = handle_timeout_trip(dial, start);
|
|
continue;
|
|
}
|
|
|
|
/* Find relative dial channel */
|
|
if (!chan || !IS_CALLER(chan, who))
|
|
channel = find_relative_dial_channel(dial, who);
|
|
|
|
/* See if this channel has been forwarded elsewhere */
|
|
if (!ast_strlen_zero(ast_channel_call_forward(who))) {
|
|
handle_call_forward(dial, channel, chan);
|
|
continue;
|
|
}
|
|
|
|
/* Attempt to read in a frame */
|
|
if (!(fr = ast_read(who))) {
|
|
/* If this is the caller then we switch state to hangup and stop */
|
|
if (chan && IS_CALLER(chan, who)) {
|
|
set_state(dial, AST_DIAL_RESULT_HANGUP);
|
|
break;
|
|
}
|
|
if (chan)
|
|
ast_poll_channel_del(chan, channel->owner);
|
|
ast_channel_publish_dial(chan, who, channel->device, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(who)));
|
|
ast_hangup(who);
|
|
channel->owner = NULL;
|
|
continue;
|
|
}
|
|
|
|
/* Process the frame */
|
|
if (chan)
|
|
handle_frame(dial, channel, fr, chan);
|
|
else
|
|
handle_frame_ownerless(dial, channel, fr);
|
|
|
|
/* Free the received frame and start all over */
|
|
ast_frfree(fr);
|
|
}
|
|
|
|
/* Do post-processing from loop */
|
|
if (dial->state == AST_DIAL_RESULT_ANSWERED) {
|
|
/* Hangup everything except that which answered */
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
if (!channel->owner || channel->owner == who)
|
|
continue;
|
|
if (chan)
|
|
ast_poll_channel_del(chan, channel->owner);
|
|
ast_channel_publish_dial(chan, channel->owner, channel->device, "CANCEL");
|
|
ast_hangup(channel->owner);
|
|
channel->cause = AST_CAUSE_ANSWERED_ELSEWHERE;
|
|
channel->owner = NULL;
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
/* If ANSWER_EXEC is enabled as an option, execute application on answered channel */
|
|
if ((channel = find_relative_dial_channel(dial, who)) && (answer_exec = FIND_RELATIVE_OPTION(dial, channel, AST_DIAL_OPTION_ANSWER_EXEC))) {
|
|
channel->is_running_app = 1;
|
|
answer_exec_run(dial, channel, answer_exec->app, answer_exec->args);
|
|
channel->is_running_app = 0;
|
|
}
|
|
|
|
if (chan && dial->options[AST_DIAL_OPTION_MUSIC] &&
|
|
!ast_strlen_zero(dial->options[AST_DIAL_OPTION_MUSIC])) {
|
|
ast_moh_stop(chan);
|
|
}
|
|
} else if (dial->state == AST_DIAL_RESULT_HANGUP) {
|
|
/* Hangup everything */
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
if (!channel->owner)
|
|
continue;
|
|
if (chan)
|
|
ast_poll_channel_del(chan, channel->owner);
|
|
ast_channel_publish_dial(chan, channel->owner, channel->device, "CANCEL");
|
|
ast_hangup(channel->owner);
|
|
channel->cause = AST_CAUSE_NORMAL_CLEARING;
|
|
channel->owner = NULL;
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
}
|
|
|
|
return dial->state;
|
|
}
|
|
|
|
/*! \brief Dial async thread function */
|
|
static void *async_dial(void *data)
|
|
{
|
|
struct ast_dial *dial = data;
|
|
if (dial->callid) {
|
|
ast_callid_threadassoc_add(dial->callid);
|
|
}
|
|
|
|
/* This is really really simple... we basically pass monitor_dial a NULL owner and it changes it's behavior */
|
|
monitor_dial(dial, NULL);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
/*! \brief Execute dialing synchronously or asynchronously
|
|
* \note Dials channels in a dial structure.
|
|
* \return Returns dial result code. (TRYING/INVALID/FAILED/ANSWERED/TIMEOUT/UNANSWERED).
|
|
*/
|
|
enum ast_dial_result ast_dial_run(struct ast_dial *dial, struct ast_channel *chan, int async)
|
|
{
|
|
enum ast_dial_result res = AST_DIAL_RESULT_TRYING;
|
|
|
|
/* Ensure required arguments are passed */
|
|
if (!dial) {
|
|
ast_debug(1, "invalid #1\n");
|
|
return AST_DIAL_RESULT_INVALID;
|
|
}
|
|
|
|
/* If there are no channels to dial we can't very well try to dial them */
|
|
if (AST_LIST_EMPTY(&dial->channels)) {
|
|
ast_debug(1, "invalid #2\n");
|
|
return AST_DIAL_RESULT_INVALID;
|
|
}
|
|
|
|
/* Dial each requested channel */
|
|
if (!begin_dial(dial, chan, async))
|
|
return AST_DIAL_RESULT_FAILED;
|
|
|
|
/* If we are running async spawn a thread and send it away... otherwise block here */
|
|
if (async) {
|
|
/* reference be released at dial destruction if it isn't NULL */
|
|
dial->callid = ast_read_threadstorage_callid();
|
|
dial->state = AST_DIAL_RESULT_TRYING;
|
|
/* Try to create a thread */
|
|
if (ast_pthread_create(&dial->thread, NULL, async_dial, dial)) {
|
|
/* Failed to create the thread - hangup all dialed channels and return failed */
|
|
ast_dial_hangup(dial);
|
|
res = AST_DIAL_RESULT_FAILED;
|
|
}
|
|
} else {
|
|
res = monitor_dial(dial, chan);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/*! \brief Return channel that answered
|
|
* \note Returns the Asterisk channel that answered
|
|
* \param dial Dialing structure
|
|
*/
|
|
struct ast_channel *ast_dial_answered(struct ast_dial *dial)
|
|
{
|
|
if (!dial)
|
|
return NULL;
|
|
|
|
return ((dial->state == AST_DIAL_RESULT_ANSWERED) ? AST_LIST_FIRST(&dial->channels)->owner : NULL);
|
|
}
|
|
|
|
/*! \brief Steal the channel that answered
|
|
* \note Returns the Asterisk channel that answered and removes it from the dialing structure
|
|
* \param dial Dialing structure
|
|
*/
|
|
struct ast_channel *ast_dial_answered_steal(struct ast_dial *dial)
|
|
{
|
|
struct ast_channel *chan = NULL;
|
|
|
|
if (!dial)
|
|
return NULL;
|
|
|
|
if (dial->state == AST_DIAL_RESULT_ANSWERED) {
|
|
chan = AST_LIST_FIRST(&dial->channels)->owner;
|
|
AST_LIST_FIRST(&dial->channels)->owner = NULL;
|
|
}
|
|
|
|
return chan;
|
|
}
|
|
|
|
/*! \brief Return state of dial
|
|
* \note Returns the state of the dial attempt
|
|
* \param dial Dialing structure
|
|
*/
|
|
enum ast_dial_result ast_dial_state(struct ast_dial *dial)
|
|
{
|
|
return dial->state;
|
|
}
|
|
|
|
/*! \brief Cancel async thread
|
|
* \note Cancel a running async thread
|
|
* \param dial Dialing structure
|
|
*/
|
|
enum ast_dial_result ast_dial_join(struct ast_dial *dial)
|
|
{
|
|
pthread_t thread;
|
|
|
|
/* If the dial structure is not running in async, return failed */
|
|
if (dial->thread == AST_PTHREADT_NULL)
|
|
return AST_DIAL_RESULT_FAILED;
|
|
|
|
/* Record thread */
|
|
thread = dial->thread;
|
|
|
|
/* Boom, commence locking */
|
|
ast_mutex_lock(&dial->lock);
|
|
|
|
/* Stop the thread */
|
|
dial->thread = AST_PTHREADT_STOP;
|
|
|
|
/* If the answered channel is running an application we have to soft hangup it, can't just poke the thread */
|
|
AST_LIST_LOCK(&dial->channels);
|
|
if (AST_LIST_FIRST(&dial->channels)->is_running_app) {
|
|
struct ast_channel *chan = AST_LIST_FIRST(&dial->channels)->owner;
|
|
if (chan) {
|
|
ast_channel_lock(chan);
|
|
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
|
|
ast_channel_unlock(chan);
|
|
}
|
|
} else {
|
|
/* Now we signal it with SIGURG so it will break out of it's waitfor */
|
|
pthread_kill(thread, SIGURG);
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
/* Yay done with it */
|
|
ast_mutex_unlock(&dial->lock);
|
|
|
|
/* Finally wait for the thread to exit */
|
|
pthread_join(thread, NULL);
|
|
|
|
/* Yay thread is all gone */
|
|
dial->thread = AST_PTHREADT_NULL;
|
|
|
|
return dial->state;
|
|
}
|
|
|
|
/*! \brief Hangup channels
|
|
* \note Hangup all active channels
|
|
* \param dial Dialing structure
|
|
*/
|
|
void ast_dial_hangup(struct ast_dial *dial)
|
|
{
|
|
struct ast_dial_channel *channel = NULL;
|
|
|
|
if (!dial)
|
|
return;
|
|
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
ast_hangup(channel->owner);
|
|
channel->owner = NULL;
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
return;
|
|
}
|
|
|
|
/*! \brief Destroys a dialing structure
|
|
* \note Destroys (free's) the given ast_dial structure
|
|
* \param dial Dialing structure to free
|
|
* \return Returns 0 on success, -1 on failure
|
|
*/
|
|
int ast_dial_destroy(struct ast_dial *dial)
|
|
{
|
|
int i = 0;
|
|
struct ast_dial_channel *channel = NULL;
|
|
|
|
if (!dial)
|
|
return -1;
|
|
|
|
/* Hangup and deallocate all the dialed channels */
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE_SAFE_BEGIN(&dial->channels, channel, list) {
|
|
/* Disable any enabled options */
|
|
for (i = 0; i < AST_DIAL_OPTION_MAX; i++) {
|
|
if (!channel->options[i])
|
|
continue;
|
|
if (option_types[i].disable)
|
|
option_types[i].disable(channel->options[i]);
|
|
channel->options[i] = NULL;
|
|
}
|
|
|
|
/* Hang up channel if need be */
|
|
ast_hangup(channel->owner);
|
|
channel->owner = NULL;
|
|
|
|
/* Free structure */
|
|
ast_free(channel->tech);
|
|
ast_free(channel->device);
|
|
ast_free(channel->assignedid1);
|
|
ast_free(channel->assignedid2);
|
|
|
|
AST_LIST_REMOVE_CURRENT(list);
|
|
ast_free(channel);
|
|
}
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
/* Disable any enabled options globally */
|
|
for (i = 0; i < AST_DIAL_OPTION_MAX; i++) {
|
|
if (!dial->options[i])
|
|
continue;
|
|
if (option_types[i].disable)
|
|
option_types[i].disable(dial->options[i]);
|
|
dial->options[i] = NULL;
|
|
}
|
|
|
|
/* Lock be gone! */
|
|
ast_mutex_destroy(&dial->lock);
|
|
|
|
/* Get rid of the reference to the ast_callid */
|
|
if (dial->callid) {
|
|
ast_callid_unref(dial->callid);
|
|
}
|
|
|
|
/* Free structure */
|
|
ast_free(dial);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Enables an option globally
|
|
* \param dial Dial structure to enable option on
|
|
* \param option Option to enable
|
|
* \param data Data to pass to this option (not always needed)
|
|
* \return Returns 0 on success, -1 on failure
|
|
*/
|
|
int ast_dial_option_global_enable(struct ast_dial *dial, enum ast_dial_option option, void *data)
|
|
{
|
|
/* If the option is already enabled, return failure */
|
|
if (dial->options[option])
|
|
return -1;
|
|
|
|
/* Execute enable callback if it exists, if not simply make sure the value is set */
|
|
if (option_types[option].enable)
|
|
dial->options[option] = option_types[option].enable(data);
|
|
else
|
|
dial->options[option] = (void*)1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Helper function for finding a channel in a dial structure based on number
|
|
*/
|
|
static struct ast_dial_channel *find_dial_channel(struct ast_dial *dial, int num)
|
|
{
|
|
struct ast_dial_channel *channel = AST_LIST_LAST(&dial->channels);
|
|
|
|
/* We can try to predict programmer behavior, the last channel they added is probably the one they wanted to modify */
|
|
if (channel->num == num)
|
|
return channel;
|
|
|
|
/* Hrm not at the end... looking through the list it is! */
|
|
AST_LIST_LOCK(&dial->channels);
|
|
AST_LIST_TRAVERSE(&dial->channels, channel, list) {
|
|
if (channel->num == num)
|
|
break;
|
|
}
|
|
AST_LIST_UNLOCK(&dial->channels);
|
|
|
|
return channel;
|
|
}
|
|
|
|
/*! \brief Enables an option per channel
|
|
* \param dial Dial structure
|
|
* \param num Channel number to enable option on
|
|
* \param option Option to enable
|
|
* \param data Data to pass to this option (not always needed)
|
|
* \return Returns 0 on success, -1 on failure
|
|
*/
|
|
int ast_dial_option_enable(struct ast_dial *dial, int num, enum ast_dial_option option, void *data)
|
|
{
|
|
struct ast_dial_channel *channel = NULL;
|
|
|
|
/* Ensure we have required arguments */
|
|
if (!dial || AST_LIST_EMPTY(&dial->channels))
|
|
return -1;
|
|
|
|
if (!(channel = find_dial_channel(dial, num)))
|
|
return -1;
|
|
|
|
/* If the option is already enabled, return failure */
|
|
if (channel->options[option])
|
|
return -1;
|
|
|
|
/* Execute enable callback if it exists, if not simply make sure the value is set */
|
|
if (option_types[option].enable)
|
|
channel->options[option] = option_types[option].enable(data);
|
|
else
|
|
channel->options[option] = (void*)1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Disables an option globally
|
|
* \param dial Dial structure to disable option on
|
|
* \param option Option to disable
|
|
* \return Returns 0 on success, -1 on failure
|
|
*/
|
|
int ast_dial_option_global_disable(struct ast_dial *dial, enum ast_dial_option option)
|
|
{
|
|
/* If the option is not enabled, return failure */
|
|
if (!dial->options[option]) {
|
|
return -1;
|
|
}
|
|
|
|
/* Execute callback of option to disable if it exists */
|
|
if (option_types[option].disable)
|
|
option_types[option].disable(dial->options[option]);
|
|
|
|
/* Finally disable option on the structure */
|
|
dial->options[option] = NULL;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Disables an option per channel
|
|
* \param dial Dial structure
|
|
* \param num Channel number to disable option on
|
|
* \param option Option to disable
|
|
* \return Returns 0 on success, -1 on failure
|
|
*/
|
|
int ast_dial_option_disable(struct ast_dial *dial, int num, enum ast_dial_option option)
|
|
{
|
|
struct ast_dial_channel *channel = NULL;
|
|
|
|
/* Ensure we have required arguments */
|
|
if (!dial || AST_LIST_EMPTY(&dial->channels))
|
|
return -1;
|
|
|
|
if (!(channel = find_dial_channel(dial, num)))
|
|
return -1;
|
|
|
|
/* If the option is not enabled, return failure */
|
|
if (!channel->options[option])
|
|
return -1;
|
|
|
|
/* Execute callback of option to disable it if it exists */
|
|
if (option_types[option].disable)
|
|
option_types[option].disable(channel->options[option]);
|
|
|
|
/* Finally disable the option on the structure */
|
|
channel->options[option] = NULL;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_dial_reason(struct ast_dial *dial, int num)
|
|
{
|
|
struct ast_dial_channel *channel;
|
|
|
|
if (!dial || AST_LIST_EMPTY(&dial->channels) || !(channel = find_dial_channel(dial, num))) {
|
|
return -1;
|
|
}
|
|
|
|
return channel->cause;
|
|
}
|
|
|
|
struct ast_channel *ast_dial_get_channel(struct ast_dial *dial, int num)
|
|
{
|
|
struct ast_dial_channel *channel;
|
|
|
|
if (!dial || AST_LIST_EMPTY(&dial->channels) || !(channel = find_dial_channel(dial, num))) {
|
|
return NULL;
|
|
}
|
|
|
|
return channel->owner;
|
|
}
|
|
|
|
void ast_dial_set_state_callback(struct ast_dial *dial, ast_dial_state_callback callback)
|
|
{
|
|
dial->state_callback = callback;
|
|
}
|
|
|
|
void ast_dial_set_user_data(struct ast_dial *dial, void *user_data)
|
|
{
|
|
dial->user_data = user_data;
|
|
}
|
|
|
|
void *ast_dial_get_user_data(struct ast_dial *dial)
|
|
{
|
|
return dial->user_data;
|
|
}
|
|
|
|
/*! \brief Set the maximum time (globally) allowed for trying to ring phones
|
|
* \param dial The dial structure to apply the time limit to
|
|
* \param timeout Maximum time allowed
|
|
* \return nothing
|
|
*/
|
|
void ast_dial_set_global_timeout(struct ast_dial *dial, int timeout)
|
|
{
|
|
dial->timeout = timeout;
|
|
|
|
if (dial->timeout > 0 && (dial->actual_timeout > dial->timeout || dial->actual_timeout == -1))
|
|
dial->actual_timeout = dial->timeout;
|
|
|
|
return;
|
|
}
|
|
|
|
/*! \brief Set the maximum time (per channel) allowed for trying to ring the phone
|
|
* \param dial The dial structure the channel belongs to
|
|
* \param num Channel number to set timeout on
|
|
* \param timeout Maximum time allowed
|
|
* \return nothing
|
|
*/
|
|
void ast_dial_set_timeout(struct ast_dial *dial, int num, int timeout)
|
|
{
|
|
struct ast_dial_channel *channel = NULL;
|
|
|
|
if (!(channel = find_dial_channel(dial, num)))
|
|
return;
|
|
|
|
channel->timeout = timeout;
|
|
|
|
if (channel->timeout > 0 && (dial->actual_timeout > channel->timeout || dial->actual_timeout == -1))
|
|
dial->actual_timeout = channel->timeout;
|
|
|
|
return;
|
|
}
|