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asterisk/channels/chan_h323.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005
*
* OpenH323 Channel Driver for ASTERISK PBX.
* By Jeremy McNamara
* For The NuFone Network
*
* chan_h323 has been derived from code created by
* Michael Manousos and Mark Spencer
*
* 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 This file is part of the chan_h323 driver for Asterisk
*
* \author Jeremy McNamara
*
* \par See also
* \arg Config_h323
* \extref OpenH323 http://www.voxgratia.org/
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<depend>openh323</depend>
<defaultenabled>yes</defaultenabled>
***/
#ifdef __cplusplus
extern "C" {
#endif
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#ifdef __cplusplus
}
#endif
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/signal.h>
#include <sys/param.h>
#include <arpa/inet.h>
#include <net/if.h>
#include <netinet/in.h>
#include <netinet/in_systm.h>
#include <netinet/ip.h>
#include <netdb.h>
#include <fcntl.h>
#ifdef __cplusplus
extern "C" {
#endif
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/musiconhold.h"
#include "asterisk/pbx.h"
#include "asterisk/utils.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/dsp.h"
#include "asterisk/causes.h"
#include "asterisk/stringfields.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/astobj.h"
#include "asterisk/format.h"
#include "asterisk/format_cap.h"
#ifdef __cplusplus
}
#endif
#undef open
#undef close
#include "h323/chan_h323.h"
receive_digit_cb on_receive_digit;
on_rtp_cb on_external_rtp_create;
start_rtp_cb on_start_rtp_channel;
setup_incoming_cb on_incoming_call;
setup_outbound_cb on_outgoing_call;
chan_ringing_cb on_chan_ringing;
con_established_cb on_connection_established;
clear_con_cb on_connection_cleared;
answer_call_cb on_answer_call;
progress_cb on_progress;
rfc2833_cb on_set_rfc2833_payload;
hangup_cb on_hangup;
setcapabilities_cb on_setcapabilities;
setpeercapabilities_cb on_setpeercapabilities;
onhold_cb on_hold;
int h323debug; /*!< global debug flag */
/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
.flags = 0,
.max_size = -1,
.resync_threshold = -1,
.impl = "",
.target_extra = -1,
};
static struct ast_jb_conf global_jbconf;
/** Variables required by Asterisk */
static const char tdesc[] = "The NuFone Network's Open H.323 Channel Driver";
static const char config[] = "h323.conf";
static char default_context[AST_MAX_CONTEXT] = "default";
static struct sockaddr_in bindaddr;
#define GLOBAL_CAPABILITY (ast_format_id_to_old_bitfield(AST_FORMAT_G723_1) | \
ast_format_id_to_old_bitfield(AST_FORMAT_GSM) | \
ast_format_id_to_old_bitfield(AST_FORMAT_ULAW) | \
ast_format_id_to_old_bitfield(AST_FORMAT_ALAW) | \
ast_format_id_to_old_bitfield(AST_FORMAT_G729A) | \
ast_format_id_to_old_bitfield(AST_FORMAT_G726_AAL2) | \
ast_format_id_to_old_bitfield(AST_FORMAT_H261)) \
/** H.323 configuration values */
static int h323_signalling_port = 1720;
static char gatekeeper[100];
static int gatekeeper_disable = 1;
static int gatekeeper_discover = 0;
static int gkroute = 0;
/* Find user by alias (h.323 id) is default, alternative is the incoming call's source IP address*/
static int userbyalias = 1;
static int acceptAnonymous = 1;
static unsigned int tos = 0;
static unsigned int cos = 0;
static char secret[50];
static unsigned int unique = 0;
static call_options_t global_options;
/*! \brief Private structure of a OpenH323 channel */
static struct oh323_pvt {
ast_mutex_t lock; /*!< Channel private lock */
call_options_t options; /*!<!< Options to be used during call setup */
int alreadygone; /*!< Whether or not we've already been destroyed by our peer */
int needdestroy; /*!< if we need to be destroyed */
call_details_t cd; /*!< Call details */
struct ast_channel *owner; /*!< Who owns us */
struct sockaddr_in sa; /*!< Our peer */
struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
int nonCodecCapability; /*!< non-audio capability */
int outgoing; /*!< Outgoing or incoming call? */
char exten[AST_MAX_EXTENSION]; /*!< Requested extension */
char context[AST_MAX_CONTEXT]; /*!< Context where to start */
char accountcode[256]; /*!< Account code */
char rdnis[80]; /*!< Referring DNIS, if available */
int amaflags; /*!< AMA Flags */
struct ast_rtp_instance *rtp; /*!< RTP Session */
struct ast_dsp *vad; /*!< Used for in-band DTMF detection */
int nativeformats; /*!< Codec formats supported by a channel */
int needhangup; /*!< Send hangup when Asterisk is ready */
int hangupcause; /*!< Hangup cause from OpenH323 layer */
int newstate; /*!< Pending state change */
int newcontrol; /*!< Pending control to send */
int newdigit; /*!< Pending DTMF digit to send */
int newduration; /*!< Pending DTMF digit duration to send */
h323_format pref_codec; /*!< Preferred codec */
h323_format peercapability; /*!< Capabilities learned from peer */
h323_format jointcapability; /*!< Common capabilities for local and remote side */
struct ast_codec_pref peer_prefs; /*!< Preferenced list of codecs which remote side supports */
int dtmf_pt[2]; /*!< Payload code used for RFC2833/CISCO messages */
int curDTMF; /*!< DTMF tone being generated to Asterisk side */
int DTMFsched; /*!< Scheduler descriptor for DTMF */
int update_rtp_info; /*!< Configuration of fd's array is pending */
int recvonly; /*!< Peer isn't wish to receive our voice stream */
int txDtmfDigit; /*!< DTMF digit being to send to H.323 side */
int noInbandDtmf; /*!< Inband DTMF processing by DSP isn't available */
int connection_established; /*!< Call got CONNECT message */
int got_progress; /*!< Call got PROGRESS message, pass inband audio */
struct oh323_pvt *next; /*!< Next channel in list */
} *iflist = NULL;
/*! \brief H323 User list */
static struct h323_user_list {
ASTOBJ_CONTAINER_COMPONENTS(struct oh323_user);
} userl;
/*! \brief H323 peer list */
static struct h323_peer_list {
ASTOBJ_CONTAINER_COMPONENTS(struct oh323_peer);
} peerl;
/*! \brief H323 alias list */
static struct h323_alias_list {
ASTOBJ_CONTAINER_COMPONENTS(struct oh323_alias);
} aliasl;
/* Asterisk RTP stuff */
static struct ast_sched_context *sched;
static struct io_context *io;
AST_MUTEX_DEFINE_STATIC(iflock); /*!< Protect the interface list (oh323_pvt) */
/*! \brief Protect the H.323 monitoring thread, so only one process can kill or start it, and not
when it's doing something critical. */
AST_MUTEX_DEFINE_STATIC(monlock);
/*! \brief Protect the H.323 capabilities list, to avoid more than one channel to set the capabilities simultaneaously in the h323 stack. */
AST_MUTEX_DEFINE_STATIC(caplock);
/*! \brief Protect the reload process */
AST_MUTEX_DEFINE_STATIC(h323_reload_lock);
static int h323_reloading = 0;
/*! \brief This is the thread for the monitor which checks for input on the channels
which are not currently in use. */
static pthread_t monitor_thread = AST_PTHREADT_NULL;
static int restart_monitor(void);
static int h323_do_reload(void);
static void delete_users(void);
static void delete_aliases(void);
static void prune_peers(void);
static struct ast_channel *oh323_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause);
static int oh323_digit_begin(struct ast_channel *c, char digit);
2007-01-19 18:06:03 +00:00
static int oh323_digit_end(struct ast_channel *c, char digit, unsigned int duration);
static int oh323_call(struct ast_channel *c, char *dest, int timeout);
static int oh323_hangup(struct ast_channel *c);
static int oh323_answer(struct ast_channel *c);
static struct ast_frame *oh323_read(struct ast_channel *c);
static int oh323_write(struct ast_channel *c, struct ast_frame *frame);
static int oh323_indicate(struct ast_channel *c, int condition, const void *data, size_t datalen);
static int oh323_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static struct ast_channel_tech oh323_tech = {
.type = "H323",
.description = tdesc,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = oh323_request,
.send_digit_begin = oh323_digit_begin,
.send_digit_end = oh323_digit_end,
.call = oh323_call,
.hangup = oh323_hangup,
.answer = oh323_answer,
.read = oh323_read,
.write = oh323_write,
.indicate = oh323_indicate,
.fixup = oh323_fixup,
.bridge = ast_rtp_instance_bridge,
};
static const char* redirectingreason2str(int redirectingreason)
{
switch (redirectingreason) {
case 0:
return "UNKNOWN";
case 1:
return "BUSY";
case 2:
return "NO_REPLY";
case 0xF:
return "UNCONDITIONAL";
default:
return "NOREDIRECT";
}
}
static void oh323_destroy_alias(struct oh323_alias *alias)
{
if (h323debug)
ast_debug(1, "Destroying alias '%s'\n", alias->name);
ast_free(alias);
}
static void oh323_destroy_user(struct oh323_user *user)
{
if (h323debug)
ast_debug(1, "Destroying user '%s'\n", user->name);
ast_free_ha(user->ha);
ast_free(user);
}
static void oh323_destroy_peer(struct oh323_peer *peer)
{
if (h323debug)
ast_debug(1, "Destroying peer '%s'\n", peer->name);
ast_free_ha(peer->ha);
ast_free(peer);
}
static int oh323_simulate_dtmf_end(const void *data)
{
struct oh323_pvt *pvt = (struct oh323_pvt *)data;
if (pvt) {
ast_mutex_lock(&pvt->lock);
/* Don't hold pvt lock while trying to lock the channel */
while (pvt->owner && ast_channel_trylock(pvt->owner)) {
DEADLOCK_AVOIDANCE(&pvt->lock);
}
if (pvt->owner) {
struct ast_frame f = {
.frametype = AST_FRAME_DTMF_END,
.subclass.integer = pvt->curDTMF,
.samples = 0,
.src = "SIMULATE_DTMF_END",
};
ast_queue_frame(pvt->owner, &f);
ast_channel_unlock(pvt->owner);
}
pvt->DTMFsched = -1;
ast_mutex_unlock(&pvt->lock);
}
return 0;
}
/*! \brief Channel and private structures should be already locked */
static void __oh323_update_info(struct ast_channel *c, struct oh323_pvt *pvt)
{
h323_format chan_nativeformats_bits = ast_format_cap_to_old_bitfield(c->nativeformats);
if (chan_nativeformats_bits != pvt->nativeformats) {
if (h323debug)
ast_debug(1, "Preparing %s for new native format\n", c->name);
ast_format_cap_from_old_bitfield(c->nativeformats, pvt->nativeformats);
ast_set_read_format(c, &c->readformat);
ast_set_write_format(c, &c->writeformat);
}
if (pvt->needhangup) {
if (h323debug)
ast_debug(1, "Process pending hangup for %s\n", c->name);
c->_softhangup |= AST_SOFTHANGUP_DEV;
c->hangupcause = pvt->hangupcause;
ast_queue_hangup_with_cause(c, pvt->hangupcause);
pvt->needhangup = 0;
pvt->newstate = pvt->newcontrol = pvt->newdigit = pvt->DTMFsched = -1;
}
if (pvt->newstate >= 0) {
ast_setstate(c, pvt->newstate);
pvt->newstate = -1;
}
if (pvt->newcontrol >= 0) {
ast_queue_control(c, pvt->newcontrol);
pvt->newcontrol = -1;
}
if (pvt->newdigit >= 0) {
struct ast_frame f = {
.frametype = AST_FRAME_DTMF_END,
.subclass.integer = pvt->newdigit,
.samples = pvt->newduration * 8,
.len = pvt->newduration,
.src = "UPDATE_INFO",
};
if (pvt->newdigit == ' ') { /* signalUpdate message */
f.subclass.integer = pvt->curDTMF;
if (pvt->DTMFsched >= 0) {
AST_SCHED_DEL(sched, pvt->DTMFsched);
}
} else { /* Regular input or signal message */
if (pvt->newduration) { /* This is a signal, signalUpdate follows */
f.frametype = AST_FRAME_DTMF_BEGIN;
AST_SCHED_DEL(sched, pvt->DTMFsched);
pvt->DTMFsched = ast_sched_add(sched, pvt->newduration, oh323_simulate_dtmf_end, pvt);
if (h323debug)
ast_log(LOG_DTMF, "Scheduled DTMF END simulation for %d ms, id=%d\n", pvt->newduration, pvt->DTMFsched);
}
pvt->curDTMF = pvt->newdigit;
}
ast_queue_frame(c, &f);
pvt->newdigit = -1;
}
if (pvt->update_rtp_info > 0) {
if (pvt->rtp) {
ast_jb_configure(c, &global_jbconf);
ast_channel_set_fd(c, 0, ast_rtp_instance_fd(pvt->rtp, 0));
ast_channel_set_fd(c, 1, ast_rtp_instance_fd(pvt->rtp, 1));
ast_queue_frame(pvt->owner, &ast_null_frame); /* Tell Asterisk to apply changes */
}
pvt->update_rtp_info = -1;
}
}
/*! \brief Only channel structure should be locked */
static void oh323_update_info(struct ast_channel *c)
{
struct oh323_pvt *pvt = c->tech_pvt;
if (pvt) {
ast_mutex_lock(&pvt->lock);
__oh323_update_info(c, pvt);
ast_mutex_unlock(&pvt->lock);
}
}
static void cleanup_call_details(call_details_t *cd)
{
if (cd->call_token) {
ast_free(cd->call_token);
cd->call_token = NULL;
}
if (cd->call_source_aliases) {
ast_free(cd->call_source_aliases);
cd->call_source_aliases = NULL;
}
if (cd->call_dest_alias) {
ast_free(cd->call_dest_alias);
cd->call_dest_alias = NULL;
}
if (cd->call_source_name) {
ast_free(cd->call_source_name);
cd->call_source_name = NULL;
}
if (cd->call_source_e164) {
ast_free(cd->call_source_e164);
cd->call_source_e164 = NULL;
}
if (cd->call_dest_e164) {
ast_free(cd->call_dest_e164);
cd->call_dest_e164 = NULL;
}
if (cd->sourceIp) {
ast_free(cd->sourceIp);
cd->sourceIp = NULL;
}
if (cd->redirect_number) {
ast_free(cd->redirect_number);
cd->redirect_number = NULL;
}
}
static void __oh323_destroy(struct oh323_pvt *pvt)
{
struct oh323_pvt *cur, *prev = NULL;
AST_SCHED_DEL(sched, pvt->DTMFsched);
if (pvt->rtp) {
ast_rtp_instance_destroy(pvt->rtp);
}
/* Free dsp used for in-band DTMF detection */
if (pvt->vad) {
ast_dsp_free(pvt->vad);
}
cleanup_call_details(&pvt->cd);
/* Unlink us from the owner if we have one */
if (pvt->owner) {
ast_channel_lock(pvt->owner);
if (h323debug)
ast_debug(1, "Detaching from %s\n", pvt->owner->name);
pvt->owner->tech_pvt = NULL;
ast_channel_unlock(pvt->owner);
}
cur = iflist;
while(cur) {
if (cur == pvt) {
if (prev)
prev->next = cur->next;
else
iflist = cur->next;
break;
}
prev = cur;
cur = cur->next;
}
if (!cur) {
ast_log(LOG_WARNING, "%p is not in list?!?! \n", cur);
} else {
ast_mutex_unlock(&pvt->lock);
ast_mutex_destroy(&pvt->lock);
ast_free(pvt);
}
}
static void oh323_destroy(struct oh323_pvt *pvt)
{
if (h323debug) {
ast_debug(1, "Destroying channel %s\n", (pvt->owner ? pvt->owner->name : "<unknown>"));
}
ast_mutex_lock(&iflock);
ast_mutex_lock(&pvt->lock);
__oh323_destroy(pvt);
ast_mutex_unlock(&iflock);
}
static int oh323_digit_begin(struct ast_channel *c, char digit)
{
struct oh323_pvt *pvt = (struct oh323_pvt *) c->tech_pvt;
char *token;
if (!pvt) {
ast_log(LOG_ERROR, "No private structure?! This is bad\n");
return -1;
}
ast_mutex_lock(&pvt->lock);
if (pvt->rtp &&
(((pvt->options.dtmfmode & H323_DTMF_RFC2833) && pvt->dtmf_pt[0])
/*|| ((pvt->options.dtmfmode & H323_DTMF_CISCO) && pvt->dtmf_pt[1]))*/)) {
/* out-of-band DTMF */
if (h323debug) {
ast_log(LOG_DTMF, "Begin sending out-of-band digit %c on %s\n", digit, c->name);
}
ast_rtp_instance_dtmf_begin(pvt->rtp, digit);
ast_mutex_unlock(&pvt->lock);
} else if (pvt->txDtmfDigit != digit) {
/* in-band DTMF */
if (h323debug) {
ast_log(LOG_DTMF, "Begin sending inband digit %c on %s\n", digit, c->name);
}
pvt->txDtmfDigit = digit;
token = pvt->cd.call_token ? ast_strdup(pvt->cd.call_token) : NULL;
ast_mutex_unlock(&pvt->lock);
h323_send_tone(token, digit);
if (token) {
ast_free(token);
}
} else
ast_mutex_unlock(&pvt->lock);
oh323_update_info(c);
return 0;
}
/*! \brief
* Send (play) the specified digit to the channel.
*
*/
2007-01-19 18:06:03 +00:00
static int oh323_digit_end(struct ast_channel *c, char digit, unsigned int duration)
{
struct oh323_pvt *pvt = (struct oh323_pvt *) c->tech_pvt;
char *token;
if (!pvt) {
ast_log(LOG_ERROR, "No private structure?! This is bad\n");
return -1;
}
ast_mutex_lock(&pvt->lock);
if (pvt->rtp && (pvt->options.dtmfmode & H323_DTMF_RFC2833) && ((pvt->dtmf_pt[0] > 0) || (pvt->dtmf_pt[0] > 0))) {
/* out-of-band DTMF */
if (h323debug) {
ast_log(LOG_DTMF, "End sending out-of-band digit %c on %s, duration %d\n", digit, c->name, duration);
}
ast_rtp_instance_dtmf_end(pvt->rtp, digit);
ast_mutex_unlock(&pvt->lock);
} else {
/* in-band DTMF */
if (h323debug) {
ast_log(LOG_DTMF, "End sending inband digit %c on %s, duration %d\n", digit, c->name, duration);
}
pvt->txDtmfDigit = ' ';
token = pvt->cd.call_token ? ast_strdup(pvt->cd.call_token) : NULL;
ast_mutex_unlock(&pvt->lock);
h323_send_tone(token, ' ');
if (token) {
ast_free(token);
}
}
oh323_update_info(c);
return 0;
}
/*! \brief
* Make a call over the specified channel to the specified
* destination.
* Returns -1 on error, 0 on success.
*/
static int oh323_call(struct ast_channel *c, char *dest, int timeout)
{
int res = 0;
struct oh323_pvt *pvt = (struct oh323_pvt *)c->tech_pvt;
const char *addr;
char called_addr[1024];
if (h323debug) {
ast_debug(1, "Calling to %s on %s\n", dest, c->name);
}
if ((c->_state != AST_STATE_DOWN) && (c->_state != AST_STATE_RESERVED)) {
ast_log(LOG_WARNING, "Line is already in use (%s)\n", c->name);
return -1;
}
ast_mutex_lock(&pvt->lock);
if (!gatekeeper_disable) {
if (ast_strlen_zero(pvt->exten)) {
ast_copy_string(called_addr, dest, sizeof(called_addr));
} else {
snprintf(called_addr, sizeof(called_addr), "%s@%s", pvt->exten, dest);
}
} else {
res = htons(pvt->sa.sin_port);
addr = ast_inet_ntoa(pvt->sa.sin_addr);
if (ast_strlen_zero(pvt->exten)) {
snprintf(called_addr, sizeof(called_addr), "%s:%d", addr, res);
} else {
snprintf(called_addr, sizeof(called_addr), "%s@%s:%d", pvt->exten, addr, res);
}
}
/* make sure null terminated */
called_addr[sizeof(called_addr) - 1] = '\0';
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
if (c->connected.id.number.valid && c->connected.id.number.str) {
ast_copy_string(pvt->options.cid_num, c->connected.id.number.str, sizeof(pvt->options.cid_num));
}
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
if (c->connected.id.name.valid && c->connected.id.name.str) {
ast_copy_string(pvt->options.cid_name, c->connected.id.name.str, sizeof(pvt->options.cid_name));
}
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
if (c->redirecting.from.number.valid && c->redirecting.from.number.str) {
ast_copy_string(pvt->options.cid_rdnis, c->redirecting.from.number.str, sizeof(pvt->options.cid_rdnis));
}
pvt->options.presentation = ast_party_id_presentation(&c->connected.id);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
pvt->options.type_of_number = c->connected.id.number.plan;
if ((addr = pbx_builtin_getvar_helper(c, "PRIREDIRECTREASON"))) {
if (!strcasecmp(addr, "UNKNOWN"))
pvt->options.redirect_reason = 0;
else if (!strcasecmp(addr, "BUSY"))
pvt->options.redirect_reason = 1;
else if (!strcasecmp(addr, "NO_REPLY"))
pvt->options.redirect_reason = 2;
else if (!strcasecmp(addr, "UNCONDITIONAL"))
pvt->options.redirect_reason = 15;
else
pvt->options.redirect_reason = -1;
} else
pvt->options.redirect_reason = -1;
pvt->options.transfer_capability = c->transfercapability;
/* indicate that this is an outgoing call */
pvt->outgoing = 1;
ast_verb(3, "Requested transfer capability: 0x%.2x - %s\n", c->transfercapability, ast_transfercapability2str(c->transfercapability));
if (h323debug)
ast_debug(1, "Placing outgoing call to %s, %d/%d\n", called_addr, pvt->options.dtmfcodec[0], pvt->options.dtmfcodec[1]);
ast_mutex_unlock(&pvt->lock);
res = h323_make_call(called_addr, &(pvt->cd), &pvt->options);
if (res) {
ast_log(LOG_NOTICE, "h323_make_call failed(%s)\n", c->name);
return -1;
}
oh323_update_info(c);
return 0;
}
static int oh323_answer(struct ast_channel *c)
{
int res;
struct oh323_pvt *pvt = (struct oh323_pvt *) c->tech_pvt;
char *token;
if (h323debug)
ast_debug(1, "Answering on %s\n", c->name);
ast_mutex_lock(&pvt->lock);
token = pvt->cd.call_token ? ast_strdup(pvt->cd.call_token) : NULL;
ast_mutex_unlock(&pvt->lock);
res = h323_answering_call(token, 0);
if (token)
ast_free(token);
oh323_update_info(c);
if (c->_state != AST_STATE_UP) {
ast_setstate(c, AST_STATE_UP);
}
return res;
}
static int oh323_hangup(struct ast_channel *c)
{
struct oh323_pvt *pvt = (struct oh323_pvt *) c->tech_pvt;
int q931cause = AST_CAUSE_NORMAL_CLEARING;
char *call_token;
if (h323debug)
ast_debug(1, "Hanging up and scheduling destroy of call %s\n", c->name);
if (!c->tech_pvt) {
ast_log(LOG_WARNING, "Asked to hangup channel not connected\n");
return 0;
}
ast_mutex_lock(&pvt->lock);
/* Determine how to disconnect */
if (pvt->owner != c) {
ast_log(LOG_WARNING, "Huh? We aren't the owner?\n");
ast_mutex_unlock(&pvt->lock);
return 0;
}
pvt->owner = NULL;
c->tech_pvt = NULL;
if (c->hangupcause) {
q931cause = c->hangupcause;
} else {
const char *cause = pbx_builtin_getvar_helper(c, "DIALSTATUS");
if (cause) {
if (!strcmp(cause, "CONGESTION")) {
q931cause = AST_CAUSE_NORMAL_CIRCUIT_CONGESTION;
} else if (!strcmp(cause, "BUSY")) {
q931cause = AST_CAUSE_USER_BUSY;
} else if (!strcmp(cause, "CHANISUNVAIL")) {
q931cause = AST_CAUSE_REQUESTED_CHAN_UNAVAIL;
} else if (!strcmp(cause, "NOANSWER")) {
q931cause = AST_CAUSE_NO_ANSWER;
} else if (!strcmp(cause, "CANCEL")) {
q931cause = AST_CAUSE_CALL_REJECTED;
}
}
}
/* Start the process if it's not already started */
if (!pvt->alreadygone && !pvt->hangupcause) {
call_token = pvt->cd.call_token ? ast_strdup(pvt->cd.call_token) : NULL;
if (call_token) {
/* Release lock to eliminate deadlock */
ast_mutex_unlock(&pvt->lock);
if (h323_clear_call(call_token, q931cause)) {
ast_log(LOG_WARNING, "ClearCall failed.\n");
}
ast_free(call_token);
ast_mutex_lock(&pvt->lock);
}
}
pvt->needdestroy = 1;
ast_mutex_unlock(&pvt->lock);
/* Update usage counter */
ast_module_unref(ast_module_info->self);
return 0;
}
/*! \brief Retrieve audio/etc from channel. Assumes pvt->lock is already held. */
static struct ast_frame *oh323_rtp_read(struct oh323_pvt *pvt)
{
struct ast_frame *f;
/* Only apply it for the first packet, we just need the correct ip/port */
if (pvt->options.nat) {
ast_rtp_instance_set_prop(pvt->rtp, AST_RTP_PROPERTY_NAT, pvt->options.nat);
pvt->options.nat = 0;
}
f = ast_rtp_instance_read(pvt->rtp, 0);
/* Don't send RFC2833 if we're not supposed to */
if (f && (f->frametype == AST_FRAME_DTMF) && !(pvt->options.dtmfmode & (H323_DTMF_RFC2833 | H323_DTMF_CISCO))) {
return &ast_null_frame;
}
if (pvt->owner) {
/* We already hold the channel lock */
if (f->frametype == AST_FRAME_VOICE) {
if (!ast_format_cap_iscompatible(pvt->owner->nativeformats, &f->subclass.format)) {
/* Try to avoid deadlock */
if (ast_channel_trylock(pvt->owner)) {
ast_log(LOG_NOTICE, "Format changed but channel is locked. Ignoring frame...\n");
return &ast_null_frame;
}
if (h323debug)
ast_debug(1, "Oooh, format changed to '%s'\n", ast_getformatname(&f->subclass.format));
ast_format_cap_set(pvt->owner->nativeformats, &f->subclass.format);
pvt->nativeformats = ast_format_to_old_bitfield(&f->subclass.format);
ast_set_read_format(pvt->owner, &pvt->owner->readformat);
ast_set_write_format(pvt->owner, &pvt->owner->writeformat);
ast_channel_unlock(pvt->owner);
}
/* Do in-band DTMF detection */
if ((pvt->options.dtmfmode & H323_DTMF_INBAND) && pvt->vad) {
if ((pvt->nativeformats & (AST_FORMAT_SLINEAR | AST_FORMAT_ALAW | AST_FORMAT_ULAW))) {
if (!ast_channel_trylock(pvt->owner)) {
f = ast_dsp_process(pvt->owner, pvt->vad, f);
ast_channel_unlock(pvt->owner);
}
else
ast_log(LOG_NOTICE, "Unable to process inband DTMF while channel is locked\n");
} else if (pvt->nativeformats && !pvt->noInbandDtmf) {
ast_log(LOG_NOTICE, "Inband DTMF is not supported on codec %s. Use RFC2833\n", ast_getformatname(&f->subclass.format));
pvt->noInbandDtmf = 1;
}
if (f &&(f->frametype == AST_FRAME_DTMF)) {
if (h323debug)
ast_log(LOG_DTMF, "Received in-band digit %c.\n", f->subclass.integer);
}
}
}
}
return f;
}
static struct ast_frame *oh323_read(struct ast_channel *c)
{
struct ast_frame *fr;
struct oh323_pvt *pvt = (struct oh323_pvt *)c->tech_pvt;
ast_mutex_lock(&pvt->lock);
__oh323_update_info(c, pvt);
switch(c->fdno) {
case 0:
fr = oh323_rtp_read(pvt);
break;
case 1:
if (pvt->rtp)
fr = ast_rtp_instance_read(pvt->rtp, 1);
else
fr = &ast_null_frame;
break;
default:
ast_log(LOG_ERROR, "Unable to handle fd %d on channel %s\n", c->fdno, c->name);
fr = &ast_null_frame;
break;
}
ast_mutex_unlock(&pvt->lock);
return fr;
}
static int oh323_write(struct ast_channel *c, struct ast_frame *frame)
{
struct oh323_pvt *pvt = (struct oh323_pvt *) c->tech_pvt;
int res = 0;
if (frame->frametype != AST_FRAME_VOICE) {
if (frame->frametype == AST_FRAME_IMAGE) {
return 0;
} else {
ast_log(LOG_WARNING, "Can't send %d type frames with H323 write\n", frame->frametype);
return 0;
}
} else {
if (!(ast_format_cap_iscompatible(c->nativeformats, &frame->subclass.format))) {
char tmp[256];
ast_log(LOG_WARNING, "Asked to transmit frame type '%s', while native formats is '%s' (read/write = %s/%s)\n",
ast_getformatname(&frame->subclass.format), ast_getformatname_multiple(tmp, sizeof(tmp), c->nativeformats), ast_getformatname(&c->readformat), ast_getformatname(&c->writeformat));
return 0;
}
}
if (pvt) {
ast_mutex_lock(&pvt->lock);
if (pvt->rtp && !pvt->recvonly)
res = ast_rtp_instance_write(pvt->rtp, frame);
__oh323_update_info(c, pvt);
ast_mutex_unlock(&pvt->lock);
}
return res;
}
static int oh323_indicate(struct ast_channel *c, int condition, const void *data, size_t datalen)
{
struct oh323_pvt *pvt = (struct oh323_pvt *) c->tech_pvt;
char *token = (char *)NULL;
int res = -1;
int got_progress;
ast_mutex_lock(&pvt->lock);
token = (pvt->cd.call_token ? ast_strdup(pvt->cd.call_token) : NULL);
got_progress = pvt->got_progress;
if (condition == AST_CONTROL_PROGRESS)
pvt->got_progress = 1;
else if ((condition == AST_CONTROL_BUSY) || (condition == AST_CONTROL_CONGESTION))
pvt->alreadygone = 1;
ast_mutex_unlock(&pvt->lock);
if (h323debug)
ast_debug(1, "OH323: Indicating %d on %s (%s)\n", condition, token, c->name);
switch(condition) {
case AST_CONTROL_RINGING:
if (c->_state == AST_STATE_RING || c->_state == AST_STATE_RINGING) {
h323_send_alerting(token);
res = (got_progress ? 0 : -1); /* Do not simulate any audio tones if we got PROGRESS message */
}
break;
case AST_CONTROL_PROGRESS:
if (c->_state != AST_STATE_UP) {
/* Do not send PROGRESS message more than once */
if (!got_progress)
h323_send_progress(token);
res = 0;
}
break;
case AST_CONTROL_BUSY:
if (c->_state != AST_STATE_UP) {
h323_answering_call(token, 1);
ast_softhangup_nolock(c, AST_SOFTHANGUP_DEV);
res = 0;
}
break;
case AST_CONTROL_CONGESTION:
if (c->_state != AST_STATE_UP) {
h323_answering_call(token, 1);
ast_softhangup_nolock(c, AST_SOFTHANGUP_DEV);
res = 0;
}
break;
case AST_CONTROL_HOLD:
h323_hold_call(token, 1);
/* We should start MOH only if remote party isn't provide audio for us */
ast_moh_start(c, data, NULL);
res = 0;
break;
case AST_CONTROL_UNHOLD:
h323_hold_call(token, 0);
ast_moh_stop(c);
res = 0;
break;
case AST_CONTROL_SRCUPDATE:
ast_rtp_instance_update_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_instance_change_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_PROCEEDING:
case -1:
break;
default:
ast_log(LOG_WARNING, "OH323: Don't know how to indicate condition %d on %s\n", condition, token);
break;
}
if (h323debug)
ast_debug(1, "OH323: Indicated %d on %s, res=%d\n", condition, token, res);
if (token)
ast_free(token);
oh323_update_info(c);
return res;
}
static int oh323_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
struct oh323_pvt *pvt = (struct oh323_pvt *) newchan->tech_pvt;
ast_mutex_lock(&pvt->lock);
if (pvt->owner != oldchan) {
ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, pvt->owner);
return -1;
}
pvt->owner = newchan;
ast_mutex_unlock(&pvt->lock);
return 0;
}
static int __oh323_rtp_create(struct oh323_pvt *pvt)
{
struct ast_sockaddr our_addr;
if (pvt->rtp)
return 0;
{
struct ast_sockaddr tmp;
ast_sockaddr_from_sin(&tmp, &bindaddr);
if (ast_find_ourip(&our_addr, &tmp, AF_INET)) {
ast_mutex_unlock(&pvt->lock);
ast_log(LOG_ERROR, "Unable to locate local IP address for RTP stream\n");
return -1;
}
}
pvt->rtp = ast_rtp_instance_new("asterisk", sched, &our_addr, NULL);
if (!pvt->rtp) {
ast_mutex_unlock(&pvt->lock);
ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno));
return -1;
}
if (h323debug)
ast_debug(1, "Created RTP channel\n");
ast_rtp_instance_set_qos(pvt->rtp, tos, cos, "H323 RTP");
if (h323debug)
ast_debug(1, "Setting NAT on RTP to %d\n", pvt->options.nat);
ast_rtp_instance_set_prop(pvt->rtp, AST_RTP_PROPERTY_NAT, pvt->options.nat);
if (pvt->dtmf_pt[0] > 0)
ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pvt->dtmf_pt[0], "audio", "telephone-event", 0);
if (pvt->dtmf_pt[1] > 0)
ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pvt->dtmf_pt[1], "audio", "cisco-telephone-event", 0);
if (pvt->peercapability)
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->peer_prefs);
if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
ast_jb_configure(pvt->owner, &global_jbconf);
ast_channel_set_fd(pvt->owner, 0, ast_rtp_instance_fd(pvt->rtp, 0));
ast_channel_set_fd(pvt->owner, 1, ast_rtp_instance_fd(pvt->rtp, 1));
ast_queue_frame(pvt->owner, &ast_null_frame); /* Tell Asterisk to apply changes */
ast_channel_unlock(pvt->owner);
} else
pvt->update_rtp_info = 1;
return 0;
}
/*! \brief Private structure should be locked on a call */
static struct ast_channel *__oh323_new(struct oh323_pvt *pvt, int state, const char *host, const char *linkedid)
{
struct ast_channel *ch;
char *cid_num, *cid_name;
h323_format fmt;
struct ast_format tmpfmt;
if (!ast_strlen_zero(pvt->options.cid_num))
cid_num = pvt->options.cid_num;
else
cid_num = pvt->cd.call_source_e164;
if (!ast_strlen_zero(pvt->options.cid_name))
cid_name = pvt->options.cid_name;
else
cid_name = pvt->cd.call_source_name;
/* Don't hold a oh323_pvt lock while we allocate a chanel */
ast_mutex_unlock(&pvt->lock);
ch = ast_channel_alloc(1, state, cid_num, cid_name, pvt->accountcode, pvt->exten, pvt->context, linkedid, pvt->amaflags, "H323/%s", host);
/* Update usage counter */
ast_module_ref(ast_module_info->self);
ast_mutex_lock(&pvt->lock);
if (ch) {
ch->tech = &oh323_tech;
if (!(fmt = pvt->jointcapability) && !(fmt = pvt->options.capability))
fmt = global_options.capability;
ast_format_cap_from_old_bitfield(ch->nativeformats, fmt);
ast_codec_choose(&pvt->options.prefs, ch->nativeformats, 1, &tmpfmt)/* | (pvt->jointcapability & AST_FORMAT_VIDEO_MASK)*/;
ast_format_cap_set(ch->nativeformats, &tmpfmt);
pvt->nativeformats = ast_format_cap_to_old_bitfield(ch->nativeformats);
ast_best_codec(ch->nativeformats, &tmpfmt);
ast_format_copy(&ch->writeformat, &tmpfmt);
ast_format_copy(&ch->rawwriteformat, &tmpfmt);
ast_format_copy(&ch->readformat, &tmpfmt);
ast_format_copy(&ch->rawreadformat, &tmpfmt);
if (!pvt->rtp)
__oh323_rtp_create(pvt);
#if 0
ast_channel_set_fd(ch, 0, ast_rtp_instance_fd(pvt->rtp, 0));
ast_channel_set_fd(ch, 1, ast_rtp_instance_fd(pvt->rtp, 1));
#endif
#ifdef VIDEO_SUPPORT
if (pvt->vrtp) {
ast_channel_set_fd(ch, 2, ast_rtp_instance_fd(pvt->vrtp, 0));
ast_channel_set_fd(ch, 3, ast_rtp_instance_fd(pvt->vrtp, 1));
}
#endif
#ifdef T38_SUPPORT
if (pvt->udptl) {
ast_channel_set_fd(ch, 4, ast_udptl_fd(pvt->udptl));
}
#endif
if (state == AST_STATE_RING) {
ch->rings = 1;
}
/* Allocate dsp for in-band DTMF support */
if (pvt->options.dtmfmode & H323_DTMF_INBAND) {
pvt->vad = ast_dsp_new();
ast_dsp_set_features(pvt->vad, DSP_FEATURE_DIGIT_DETECT);
}
/* Register channel functions. */
ch->tech_pvt = pvt;
/* Set the owner of this channel */
pvt->owner = ch;
ast_copy_string(ch->context, pvt->context, sizeof(ch->context));
ast_copy_string(ch->exten, pvt->exten, sizeof(ch->exten));
ch->priority = 1;
if (!ast_strlen_zero(pvt->accountcode)) {
ast_string_field_set(ch, accountcode, pvt->accountcode);
}
if (pvt->amaflags) {
ch->amaflags = pvt->amaflags;
}
/* Don't use ast_set_callerid() here because it will
* generate a needless NewCallerID event */
if (!ast_strlen_zero(cid_num)) {
ch->caller.ani.number.valid = 1;
ch->caller.ani.number.str = ast_strdup(cid_num);
}
if (pvt->cd.redirect_reason >= 0) {
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
ch->redirecting.from.number.valid = 1;
ch->redirecting.from.number.str = ast_strdup(pvt->cd.redirect_number);
pbx_builtin_setvar_helper(ch, "PRIREDIRECTREASON", redirectingreason2str(pvt->cd.redirect_reason));
}
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
ch->caller.id.name.presentation = pvt->cd.presentation;
ch->caller.id.number.presentation = pvt->cd.presentation;
ch->caller.id.number.plan = pvt->cd.type_of_number;
if (!ast_strlen_zero(pvt->exten) && strcmp(pvt->exten, "s")) {
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
ch->dialed.number.str = ast_strdup(pvt->exten);
}
if (pvt->cd.transfer_capability >= 0)
ch->transfercapability = pvt->cd.transfer_capability;
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(ch)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ch->name);
ast_hangup(ch);
ch = NULL;
}
}
} else {
ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
}
return ch;
}
static struct oh323_pvt *oh323_alloc(int callid)
{
struct oh323_pvt *pvt;
pvt = ast_calloc(1, sizeof(*pvt));
if (!pvt) {
ast_log(LOG_ERROR, "Couldn't allocate private structure. This is bad\n");
return NULL;
}
pvt->cd.redirect_reason = -1;
pvt->cd.transfer_capability = -1;
/* Ensure the call token is allocated for outgoing call */
if (!callid) {
if ((pvt->cd).call_token == NULL) {
(pvt->cd).call_token = ast_calloc(1, 128);
}
if (!pvt->cd.call_token) {
ast_log(LOG_ERROR, "Not enough memory to alocate call token\n");
ast_rtp_instance_destroy(pvt->rtp);
ast_free(pvt);
return NULL;
}
memset((char *)(pvt->cd).call_token, 0, 128);
pvt->cd.call_reference = callid;
}
memcpy(&pvt->options, &global_options, sizeof(pvt->options));
pvt->jointcapability = pvt->options.capability;
if (pvt->options.dtmfmode & (H323_DTMF_RFC2833 | H323_DTMF_CISCO)) {
pvt->nonCodecCapability |= AST_RTP_DTMF;
} else {
pvt->nonCodecCapability &= ~AST_RTP_DTMF;
}
ast_copy_string(pvt->context, default_context, sizeof(pvt->context));
pvt->newstate = pvt->newcontrol = pvt->newdigit = pvt->update_rtp_info = pvt->DTMFsched = -1;
ast_mutex_init(&pvt->lock);
/* Add to interface list */
ast_mutex_lock(&iflock);
pvt->next = iflist;
iflist = pvt;
ast_mutex_unlock(&iflock);
return pvt;
}
static struct oh323_pvt *find_call_locked(int call_reference, const char *token)
{
struct oh323_pvt *pvt;
ast_mutex_lock(&iflock);
pvt = iflist;
while(pvt) {
if (!pvt->needdestroy && ((signed int)pvt->cd.call_reference == call_reference)) {
/* Found the call */
if ((token != NULL) && (pvt->cd.call_token != NULL) && (!strcmp(pvt->cd.call_token, token))) {
ast_mutex_lock(&pvt->lock);
ast_mutex_unlock(&iflock);
return pvt;
} else if (token == NULL) {
ast_log(LOG_WARNING, "Call Token is NULL\n");
ast_mutex_lock(&pvt->lock);
ast_mutex_unlock(&iflock);
return pvt;
}
}
pvt = pvt->next;
}
ast_mutex_unlock(&iflock);
return NULL;
}
static int update_state(struct oh323_pvt *pvt, int state, int signal)
{
if (!pvt)
return 0;
if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
if (state >= 0)
ast_setstate(pvt->owner, state);
if (signal >= 0)
ast_queue_control(pvt->owner, signal);
ast_channel_unlock(pvt->owner);
return 1;
}
else {
if (state >= 0)
pvt->newstate = state;
if (signal >= 0)
pvt->newcontrol = signal;
return 0;
}
}
static struct oh323_alias *build_alias(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime)
{
struct oh323_alias *alias;
int found = 0;
alias = ASTOBJ_CONTAINER_FIND_UNLINK_FULL(&aliasl, name, name, 0, 0, strcasecmp);
if (alias)
found++;
else {
if (!(alias = ast_calloc(1, sizeof(*alias))))
return NULL;
ASTOBJ_INIT(alias);
}
if (!found && name)
ast_copy_string(alias->name, name, sizeof(alias->name));
for (; v || ((v = alt) && !(alt = NULL)); v = v->next) {
if (!strcasecmp(v->name, "e164")) {
ast_copy_string(alias->e164, v->value, sizeof(alias->e164));
} else if (!strcasecmp(v->name, "prefix")) {
ast_copy_string(alias->prefix, v->value, sizeof(alias->prefix));
} else if (!strcasecmp(v->name, "context")) {
ast_copy_string(alias->context, v->value, sizeof(alias->context));
} else if (!strcasecmp(v->name, "secret")) {
ast_copy_string(alias->secret, v->value, sizeof(alias->secret));
} else {
if (strcasecmp(v->value, "h323")) {
ast_log(LOG_WARNING, "Keyword %s does not make sense in type=h323\n", v->name);
}
}
}
ASTOBJ_UNMARK(alias);
return alias;
}
static struct oh323_alias *realtime_alias(const char *alias)
{
struct ast_variable *var, *tmp;
struct oh323_alias *a;
var = ast_load_realtime("h323", "name", alias, SENTINEL);
if (!var)
return NULL;
for (tmp = var; tmp; tmp = tmp->next) {
if (!strcasecmp(tmp->name, "type") &&
!(!strcasecmp(tmp->value, "alias") || !strcasecmp(tmp->value, "h323"))) {
ast_variables_destroy(var);
return NULL;
}
}
a = build_alias(alias, var, NULL, 1);
ast_variables_destroy(var);
return a;
}
static int h323_parse_allow_disallow(struct ast_codec_pref *pref, h323_format *formats, const char *list, int allowing)
{
int res;
struct ast_format_cap *cap = ast_format_cap_alloc_nolock();
if (!cap) {
return 1;
}
ast_format_cap_from_old_bitfield(cap, *formats);
res = ast_parse_allow_disallow(pref, cap, list, allowing);
*formats = ast_format_cap_to_old_bitfield(cap);
cap = ast_format_cap_destroy(cap);
return res;
}
static int update_common_options(struct ast_variable *v, struct call_options *options)
{
int tmp = 0;
char *val, *opt;
if (!strcasecmp(v->name, "allow")) {
h323_parse_allow_disallow(&options->prefs, &options->capability, v->value, 1);
} else if (!strcasecmp(v->name, "autoframing")) {
options->autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "disallow")) {
h323_parse_allow_disallow(&options->prefs, &options->capability, v->value, 0);
} else if (!strcasecmp(v->name, "dtmfmode")) {
val = ast_strdupa(v->value);
if ((opt = strchr(val, ':')) != (char *)NULL) {
*opt++ = '\0';
tmp = atoi(opt);
}
if (!strcasecmp(v->value, "inband")) {
options->dtmfmode |= H323_DTMF_INBAND;
} else if (!strcasecmp(val, "rfc2833")) {
options->dtmfmode |= H323_DTMF_RFC2833;
if (!opt) {
options->dtmfcodec[0] = H323_DTMF_RFC2833_PT;
} else if ((tmp >= 96) && (tmp < 128)) {
options->dtmfcodec[0] = tmp;
} else {
options->dtmfcodec[0] = H323_DTMF_RFC2833_PT;
ast_log(LOG_WARNING, "Unknown rfc2833 payload %s specified at line %d, using default %d\n", opt, v->lineno, options->dtmfcodec[0]);
}
} else if (!strcasecmp(val, "cisco")) {
options->dtmfmode |= H323_DTMF_CISCO;
if (!opt) {
options->dtmfcodec[1] = H323_DTMF_CISCO_PT;
} else if ((tmp >= 96) && (tmp < 128)) {
options->dtmfcodec[1] = tmp;
} else {
options->dtmfcodec[1] = H323_DTMF_CISCO_PT;
ast_log(LOG_WARNING, "Unknown Cisco DTMF payload %s specified at line %d, using default %d\n", opt, v->lineno, options->dtmfcodec[1]);
}
} else if (!strcasecmp(v->value, "h245-signal")) {
options->dtmfmode |= H323_DTMF_SIGNAL;
} else {
ast_log(LOG_WARNING, "Unknown dtmf mode '%s' at line %d\n", v->value, v->lineno);
}
} else if (!strcasecmp(v->name, "dtmfcodec")) {
ast_log(LOG_NOTICE, "Option %s at line %d is deprecated. Use dtmfmode=rfc2833[:<payload>] instead.\n", v->name, v->lineno);
tmp = atoi(v->value);
if (tmp < 96)
ast_log(LOG_WARNING, "Invalid %s value %s at line %d\n", v->name, v->value, v->lineno);
else
options->dtmfcodec[0] = tmp;
} else if (!strcasecmp(v->name, "bridge")) {
options->bridge = ast_true(v->value);
} else if (!strcasecmp(v->name, "nat")) {
options->nat = ast_true(v->value);
} else if (!strcasecmp(v->name, "fastStart")) {
options->fastStart = ast_true(v->value);
} else if (!strcasecmp(v->name, "h245Tunneling")) {
options->h245Tunneling = ast_true(v->value);
} else if (!strcasecmp(v->name, "silenceSuppression")) {
options->silenceSuppression = ast_true(v->value);
} else if (!strcasecmp(v->name, "progress_setup")) {
tmp = atoi(v->value);
if ((tmp != 0) && (tmp != 1) && (tmp != 3) && (tmp != 8)) {
ast_log(LOG_WARNING, "Invalid value %s for %s at line %d, assuming 0\n", v->value, v->name, v->lineno);
tmp = 0;
}
options->progress_setup = tmp;
} else if (!strcasecmp(v->name, "progress_alert")) {
tmp = atoi(v->value);
if ((tmp != 0) && (tmp != 1) && (tmp != 8)) {
ast_log(LOG_WARNING, "Invalid value %s for %s at line %d, assuming 0\n", v->value, v->name, v->lineno);
tmp = 0;
}
options->progress_alert = tmp;
} else if (!strcasecmp(v->name, "progress_audio")) {
options->progress_audio = ast_true(v->value);
} else if (!strcasecmp(v->name, "callerid")) {
ast_callerid_split(v->value, options->cid_name, sizeof(options->cid_name), options->cid_num, sizeof(options->cid_num));
} else if (!strcasecmp(v->name, "fullname")) {
ast_copy_string(options->cid_name, v->value, sizeof(options->cid_name));
} else if (!strcasecmp(v->name, "cid_number")) {
ast_copy_string(options->cid_num, v->value, sizeof(options->cid_num));
} else if (!strcasecmp(v->name, "tunneling")) {
if (!strcasecmp(v->value, "none"))
options->tunnelOptions = 0;
else if (!strcasecmp(v->value, "cisco"))
options->tunnelOptions |= H323_TUNNEL_CISCO;
else if (!strcasecmp(v->value, "qsig"))
options->tunnelOptions |= H323_TUNNEL_QSIG;
else
ast_log(LOG_WARNING, "Invalid value %s for %s at line %d\n", v->value, v->name, v->lineno);
} else if (!strcasecmp(v->name, "hold")) {
if (!strcasecmp(v->value, "none"))
options->holdHandling = ~0;
else if (!strcasecmp(v->value, "notify"))
options->holdHandling |= H323_HOLD_NOTIFY;
else if (!strcasecmp(v->value, "q931only"))
options->holdHandling |= H323_HOLD_NOTIFY | H323_HOLD_Q931ONLY;
else if (!strcasecmp(v->value, "h450"))
options->holdHandling |= H323_HOLD_H450;
else
ast_log(LOG_WARNING, "Invalid value %s for %s at line %d\n", v->value, v->name, v->lineno);
} else
return 1;
return 0;
}
static struct oh323_user *build_user(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime)
{
struct oh323_user *user;
struct ast_ha *oldha;
int found = 0;
int format;
user = ASTOBJ_CONTAINER_FIND_UNLINK_FULL(&userl, name, name, 0, 0, strcmp);
if (user)
found++;
else {
if (!(user = ast_calloc(1, sizeof(*user))))
return NULL;
ASTOBJ_INIT(user);
}
oldha = user->ha;
user->ha = (struct ast_ha *)NULL;
memcpy(&user->options, &global_options, sizeof(user->options));
user->options.dtmfmode = 0;
user->options.holdHandling = 0;
/* Set default context */
ast_copy_string(user->context, default_context, sizeof(user->context));
if (user && !found)
ast_copy_string(user->name, name, sizeof(user->name));
#if 0 /* XXX Port channel variables functionality from chan_sip XXX */
if (user->chanvars) {
ast_variables_destroy(user->chanvars);
user->chanvars = NULL;
}
#endif
for (; v || ((v = alt) && !(alt = NULL)); v = v->next) {
if (!update_common_options(v, &user->options))
continue;
if (!strcasecmp(v->name, "context")) {
ast_copy_string(user->context, v->value, sizeof(user->context));
} else if (!strcasecmp(v->name, "secret")) {
ast_copy_string(user->secret, v->value, sizeof(user->secret));
} else if (!strcasecmp(v->name, "accountcode")) {
ast_copy_string(user->accountcode, v->value, sizeof(user->accountcode));
} else if (!strcasecmp(v->name, "host")) {
if (!strcasecmp(v->value, "dynamic")) {
ast_log(LOG_ERROR, "A dynamic host on a type=user does not make any sense\n");
ASTOBJ_UNREF(user, oh323_destroy_user);
return NULL;
} else {
struct ast_sockaddr tmp;
if (ast_get_ip(&tmp, v->value)) {
ASTOBJ_UNREF(user, oh323_destroy_user);
return NULL;
}
ast_sockaddr_to_sin(&tmp, &user->addr);
}
/* Let us know we need to use ip authentication */
user->host = 1;
} else if (!strcasecmp(v->name, "amaflags")) {
format = ast_cdr_amaflags2int(v->value);
if (format < 0) {
ast_log(LOG_WARNING, "Invalid AMA Flags: %s at line %d\n", v->value, v->lineno);
} else {
user->amaflags = format;
}
} else if (!strcasecmp(v->name, "permit") ||
!strcasecmp(v->name, "deny")) {
int ha_error = 0;
user->ha = ast_append_ha(v->name, v->value, user->ha, &ha_error);
if (ha_error)
ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
}
}
if (!user->options.dtmfmode)
user->options.dtmfmode = global_options.dtmfmode;
if (user->options.holdHandling == ~0)
user->options.holdHandling = 0;
else if (!user->options.holdHandling)
user->options.holdHandling = global_options.holdHandling;
ASTOBJ_UNMARK(user);
ast_free_ha(oldha);
return user;
}
static struct oh323_user *realtime_user(const call_details_t *cd)
{
struct ast_variable *var, *tmp;
struct oh323_user *user;
const char *username;
if (userbyalias)
var = ast_load_realtime("h323", "name", username = cd->call_source_aliases, SENTINEL);
else {
username = (char *)NULL;
var = ast_load_realtime("h323", "host", cd->sourceIp, SENTINEL);
}
if (!var)
return NULL;
for (tmp = var; tmp; tmp = tmp->next) {
if (!strcasecmp(tmp->name, "type") &&
!(!strcasecmp(tmp->value, "user") || !strcasecmp(tmp->value, "friend"))) {
ast_variables_destroy(var);
return NULL;
} else if (!username && !strcasecmp(tmp->name, "name"))
username = tmp->value;
}
if (!username) {
ast_log(LOG_WARNING, "Cannot determine user name for IP address %s\n", cd->sourceIp);
ast_variables_destroy(var);
return NULL;
}
user = build_user(username, var, NULL, 1);
ast_variables_destroy(var);
return user;
}
static struct oh323_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime)
{
struct oh323_peer *peer;
struct ast_ha *oldha;
int found = 0;
peer = ASTOBJ_CONTAINER_FIND_UNLINK_FULL(&peerl, name, name, 0, 0, strcmp);
if (peer)
found++;
else {
if (!(peer = ast_calloc(1, sizeof(*peer))))
return NULL;
ASTOBJ_INIT(peer);
}
oldha = peer->ha;
peer->ha = NULL;
memcpy(&peer->options, &global_options, sizeof(peer->options));
peer->options.dtmfmode = 0;
peer->options.holdHandling = 0;
peer->addr.sin_port = htons(h323_signalling_port);
peer->addr.sin_family = AF_INET;
if (!found && name)
ast_copy_string(peer->name, name, sizeof(peer->name));
#if 0 /* XXX Port channel variables functionality from chan_sip XXX */
if (peer->chanvars) {
ast_variables_destroy(peer->chanvars);
peer->chanvars = NULL;
}
#endif
/* Default settings for mailbox */
peer->mailbox[0] = '\0';
for (; v || ((v = alt) && !(alt = NULL)); v = v->next) {
if (!update_common_options(v, &peer->options))
continue;
if (!strcasecmp(v->name, "host")) {
if (!strcasecmp(v->value, "dynamic")) {
ast_log(LOG_ERROR, "Dynamic host configuration not implemented.\n");
ASTOBJ_UNREF(peer, oh323_destroy_peer);
return NULL;
}
{
struct ast_sockaddr tmp;
if (ast_get_ip(&tmp, v->value)) {
ast_log(LOG_ERROR, "Could not determine IP for %s\n", v->value);
ASTOBJ_UNREF(peer, oh323_destroy_peer);
return NULL;
}
ast_sockaddr_to_sin(&tmp, &peer->addr);
}
} else if (!strcasecmp(v->name, "port")) {
peer->addr.sin_port = htons(atoi(v->value));
} else if (!strcasecmp(v->name, "permit") ||
!strcasecmp(v->name, "deny")) {
int ha_error = 0;
peer->ha = ast_append_ha(v->name, v->value, peer->ha, &ha_error);
if (ha_error)
ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
} else if (!strcasecmp(v->name, "mailbox")) {
ast_copy_string(peer->mailbox, v->value, sizeof(peer->mailbox));
} else if (!strcasecmp(v->name, "hasvoicemail")) {
if (ast_true(v->value) && ast_strlen_zero(peer->mailbox)) {
ast_copy_string(peer->mailbox, name, sizeof(peer->mailbox));
}
}
}
if (!peer->options.dtmfmode)
peer->options.dtmfmode = global_options.dtmfmode;
if (peer->options.holdHandling == ~0)
peer->options.holdHandling = 0;
else if (!peer->options.holdHandling)
peer->options.holdHandling = global_options.holdHandling;
ASTOBJ_UNMARK(peer);
ast_free_ha(oldha);
return peer;
}
static struct oh323_peer *realtime_peer(const char *peername, struct sockaddr_in *sin)
{
struct oh323_peer *peer;
struct ast_variable *var;
struct ast_variable *tmp;
const char *addr = NULL;
/* First check on peer name */
if (peername)
var = ast_load_realtime("h323", "name", peername, SENTINEL);
else if (sin) /* Then check on IP address for dynamic peers */
var = ast_load_realtime("h323", "host", addr = ast_inet_ntoa(sin->sin_addr), SENTINEL);
else
return NULL;
if (!var)
return NULL;
for (tmp = var; tmp; tmp = tmp->next) {
/* If this is type=user, then skip this object. */
if (!strcasecmp(tmp->name, "type") &&
!(!strcasecmp(tmp->value, "peer") || !strcasecmp(tmp->value, "friend"))) {
ast_variables_destroy(var);
return NULL;
} else if (!peername && !strcasecmp(tmp->name, "name")) {
peername = tmp->value;
}
}
if (!peername) { /* Did not find peer in realtime */
ast_log(LOG_WARNING, "Cannot determine peer name for IP address %s\n", addr);
ast_variables_destroy(var);
return NULL;
}
/* Peer found in realtime, now build it in memory */
peer = build_peer(peername, var, NULL, 1);
ast_variables_destroy(var);
return peer;
}
static int oh323_addrcmp_str(struct in_addr inaddr, char *addr)
{
return strcmp(ast_inet_ntoa(inaddr), addr);
}
static struct oh323_user *find_user(const call_details_t *cd, int realtime)
{
struct oh323_user *u;
if (userbyalias)
u = ASTOBJ_CONTAINER_FIND(&userl, cd->call_source_aliases);
else
u = ASTOBJ_CONTAINER_FIND_FULL(&userl, cd->sourceIp, addr.sin_addr, 0, 0, oh323_addrcmp_str);
if (!u && realtime)
u = realtime_user(cd);
if (!u && h323debug)
ast_debug(1, "Could not find user by name %s or address %s\n", cd->call_source_aliases, cd->sourceIp);
return u;
}
static int oh323_addrcmp(struct sockaddr_in addr, struct sockaddr_in *sin)
{
int res;
if (!sin)
res = -1;
else
res = inaddrcmp(&addr , sin);
return res;
}
static struct oh323_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
{
struct oh323_peer *p;
if (peer)
p = ASTOBJ_CONTAINER_FIND(&peerl, peer);
else
p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, addr, 0, 0, oh323_addrcmp);
if (!p && realtime)
p = realtime_peer(peer, sin);
if (!p && h323debug)
ast_debug(1, "Could not find peer by name %s or address %s\n", (peer ? peer : "<NONE>"), (sin ? ast_inet_ntoa(sin->sin_addr) : "<NONE>"));
return p;
}
static int create_addr(struct oh323_pvt *pvt, char *opeer)
{
struct hostent *hp;
struct ast_hostent ahp;
struct oh323_peer *p;
int portno;
int found = 0;
char *port;
char *hostn;
char peer[256] = "";
ast_copy_string(peer, opeer, sizeof(peer));
port = strchr(peer, ':');
if (port) {
*port = '\0';
port++;
}
pvt->sa.sin_family = AF_INET;
p = find_peer(peer, NULL, 1);
if (p) {
found++;
memcpy(&pvt->options, &p->options, sizeof(pvt->options));
pvt->jointcapability = pvt->options.capability;
if (pvt->options.dtmfmode) {
if (pvt->options.dtmfmode & H323_DTMF_RFC2833) {
pvt->nonCodecCapability |= AST_RTP_DTMF;
} else {
pvt->nonCodecCapability &= ~AST_RTP_DTMF;
}
}
if (p->addr.sin_addr.s_addr) {
pvt->sa.sin_addr = p->addr.sin_addr;
pvt->sa.sin_port = p->addr.sin_port;
}
ASTOBJ_UNREF(p, oh323_destroy_peer);
}
if (!p && !found) {
hostn = peer;
if (port) {
portno = atoi(port);
} else {
portno = h323_signalling_port;
}
hp = ast_gethostbyname(hostn, &ahp);
if (hp) {
memcpy(&pvt->sa.sin_addr, hp->h_addr, sizeof(pvt->sa.sin_addr));
pvt->sa.sin_port = htons(portno);
/* Look peer by address */
p = find_peer(NULL, &pvt->sa, 1);
memcpy(&pvt->options, (p ? &p->options : &global_options), sizeof(pvt->options));
pvt->jointcapability = pvt->options.capability;
if (p) {
ASTOBJ_UNREF(p, oh323_destroy_peer);
}
if (pvt->options.dtmfmode) {
if (pvt->options.dtmfmode & H323_DTMF_RFC2833) {
pvt->nonCodecCapability |= AST_RTP_DTMF;
} else {
pvt->nonCodecCapability &= ~AST_RTP_DTMF;
}
}
return 0;
} else {
ast_log(LOG_WARNING, "No such host: %s\n", peer);
return -1;
}
} else if (!found) {
return -1;
} else {
return 0;
}
}
static struct ast_channel *oh323_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause)
{
struct oh323_pvt *pvt;
struct ast_channel *tmpc = NULL;
char *dest = (char *)data;
char *ext, *host;
char *h323id = NULL;
char tmp[256], tmp1[256];
if (h323debug)
ast_debug(1, "type=%s, format=%s, data=%s.\n", type, ast_getformatname_multiple(tmp, sizeof(tmp), cap), (char *)data);
pvt = oh323_alloc(0);
if (!pvt) {
ast_log(LOG_WARNING, "Unable to build pvt data for '%s'\n", (char *)data);
return NULL;
}
if (!(ast_format_cap_has_type(cap, AST_FORMAT_TYPE_AUDIO))) {
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%s'\n", ast_getformatname_multiple(tmp, sizeof(tmp), cap));
oh323_destroy(pvt);
if (cause)
*cause = AST_CAUSE_INCOMPATIBLE_DESTINATION;
return NULL;
}
ast_copy_string(tmp, dest, sizeof(tmp));
host = strchr(tmp, '@');
if (host) {
*host = '\0';
host++;
ext = tmp;
} else {
ext = strrchr(tmp, '/');
if (ext)
*ext++ = '\0';
host = tmp;
}
strtok_r(host, "/", &(h323id));
if (!ast_strlen_zero(h323id)) {
h323_set_id(h323id);
}
if (ext) {
ast_copy_string(pvt->exten, ext, sizeof(pvt->exten));
}
if (h323debug)
ast_debug(1, "Extension: %s Host: %s\n", pvt->exten, host);
if (gatekeeper_disable) {
if (create_addr(pvt, host)) {
oh323_destroy(pvt);
if (cause)
*cause = AST_CAUSE_DESTINATION_OUT_OF_ORDER;
return NULL;
}
}
else {
memcpy(&pvt->options, &global_options, sizeof(pvt->options));
pvt->jointcapability = pvt->options.capability;
if (pvt->options.dtmfmode) {
if (pvt->options.dtmfmode & H323_DTMF_RFC2833) {
pvt->nonCodecCapability |= AST_RTP_DTMF;
} else {
pvt->nonCodecCapability &= ~AST_RTP_DTMF;
}
}
}
ast_mutex_lock(&caplock);
/* Generate unique channel identifier */
snprintf(tmp1, sizeof(tmp1)-1, "%s-%u", host, ++unique);
tmp1[sizeof(tmp1)-1] = '\0';
ast_mutex_unlock(&caplock);
ast_mutex_lock(&pvt->lock);
tmpc = __oh323_new(pvt, AST_STATE_DOWN, tmp1, requestor ? requestor->linkedid : NULL);
ast_mutex_unlock(&pvt->lock);
if (!tmpc) {
oh323_destroy(pvt);
if (cause)
*cause = AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
}
ast_update_use_count();
restart_monitor();
return tmpc;
}
/*! \brief Find a call by alias */
static struct oh323_alias *find_alias(const char *source_aliases, int realtime)
{
struct oh323_alias *a;
a = ASTOBJ_CONTAINER_FIND(&aliasl, source_aliases);
if (!a && realtime)
a = realtime_alias(source_aliases);
return a;
}
/*! \brief
* Callback for sending digits from H.323 up to asterisk
*
*/
static int receive_digit(unsigned call_reference, char digit, const char *token, int duration)
{
struct oh323_pvt *pvt;
int res;
pvt = find_call_locked(call_reference, token);
if (!pvt) {
ast_log(LOG_ERROR, "Received digit '%c' (%u ms) for call %s without private structure\n", digit, duration, token);
return -1;
}
if (h323debug)
ast_log(LOG_DTMF, "Received %s digit '%c' (%u ms) for call %s\n", (digit == ' ' ? "update for" : "new"), (digit == ' ' ? pvt->curDTMF : digit), duration, token);
if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
if (digit == '!')
res = ast_queue_control(pvt->owner, AST_CONTROL_FLASH);
else {
struct ast_frame f = {
.frametype = AST_FRAME_DTMF_END,
.subclass.integer = digit,
.samples = duration * 8,
.len = duration,
.src = "SEND_DIGIT",
};
if (digit == ' ') { /* signalUpdate message */
f.subclass.integer = pvt->curDTMF;
AST_SCHED_DEL(sched, pvt->DTMFsched);
} else { /* Regular input or signal message */
if (pvt->DTMFsched >= 0) {
/* We still don't send DTMF END from previous event, send it now */
AST_SCHED_DEL(sched, pvt->DTMFsched);
f.subclass.integer = pvt->curDTMF;
f.samples = f.len = 0;
ast_queue_frame(pvt->owner, &f);
/* Restore values */
f.subclass.integer = digit;
f.samples = duration * 8;
f.len = duration;
}
if (duration) { /* This is a signal, signalUpdate follows */
f.frametype = AST_FRAME_DTMF_BEGIN;
pvt->DTMFsched = ast_sched_add(sched, duration, oh323_simulate_dtmf_end, pvt);
if (h323debug)
ast_log(LOG_DTMF, "Scheduled DTMF END simulation for %d ms, id=%d\n", duration, pvt->DTMFsched);
}
pvt->curDTMF = digit;
}
res = ast_queue_frame(pvt->owner, &f);
}
ast_channel_unlock(pvt->owner);
} else {
if (digit == '!')
pvt->newcontrol = AST_CONTROL_FLASH;
else {
pvt->newduration = duration;
pvt->newdigit = digit;
}
res = 0;
}
ast_mutex_unlock(&pvt->lock);
return res;
}
/*! \brief
* Callback function used to inform the H.323 stack of the local rtp ip/port details
*
* \return Returns the local RTP information
*/
static struct rtp_info *external_rtp_create(unsigned call_reference, const char * token)
{
struct oh323_pvt *pvt;
struct sockaddr_in us;
struct rtp_info *info;
info = ast_calloc(1, sizeof(*info));
if (!info) {
ast_log(LOG_ERROR, "Unable to allocated info structure, this is very bad\n");
return NULL;
}
pvt = find_call_locked(call_reference, token);
if (!pvt) {
ast_free(info);
ast_log(LOG_ERROR, "Unable to find call %s(%d)\n", token, call_reference);
return NULL;
}
if (!pvt->rtp)
__oh323_rtp_create(pvt);
if (!pvt->rtp) {
ast_mutex_unlock(&pvt->lock);
ast_free(info);
ast_log(LOG_ERROR, "No RTP stream is available for call %s (%d)", token, call_reference);
return NULL;
}
/* figure out our local RTP port and tell the H.323 stack about it */
{
struct ast_sockaddr tmp;
ast_rtp_instance_get_local_address(pvt->rtp, &tmp);
ast_sockaddr_to_sin(&tmp, &us);
}
ast_mutex_unlock(&pvt->lock);
ast_copy_string(info->addr, ast_inet_ntoa(us.sin_addr), sizeof(info->addr));
info->port = ntohs(us.sin_port);
if (h323debug)
ast_debug(1, "Sending RTP 'US' %s:%d\n", info->addr, info->port);
return info;
}
/*! \brief
* Call-back function passing remote ip/port information from H.323 to asterisk
*
* Returns nothing
*/
static void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int remotePort, const char *token, int pt)
{
struct oh323_pvt *pvt;
struct sockaddr_in them;
int nativeformats_changed;
enum { NEED_NONE, NEED_HOLD, NEED_UNHOLD } rtp_change = NEED_NONE;
if (h323debug)
ast_debug(1, "Setting up RTP connection for %s\n", token);
/* Find the call or allocate a private structure if call not found */
pvt = find_call_locked(call_reference, token);
if (!pvt) {
ast_log(LOG_ERROR, "Something is wrong: rtp\n");
return;
}
if (pvt->alreadygone) {
ast_mutex_unlock(&pvt->lock);
return;
}
if (!pvt->rtp)
__oh323_rtp_create(pvt);
if ((pt == 2) && (pvt->jointcapability & AST_FORMAT_G726_AAL2)) {
ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pt, "audio", "G726-32", AST_RTP_OPT_G726_NONSTANDARD);
}
them.sin_family = AF_INET;
/* only works for IPv4 */
them.sin_addr.s_addr = inet_addr(remoteIp);
them.sin_port = htons(remotePort);
if (them.sin_addr.s_addr) {
{
struct ast_sockaddr tmp;
ast_sockaddr_from_sin(&tmp, &them);
ast_rtp_instance_set_remote_address(pvt->rtp, &tmp);
}
if (pvt->recvonly) {
pvt->recvonly = 0;
rtp_change = NEED_UNHOLD;
}
} else {
ast_rtp_instance_stop(pvt->rtp);
if (!pvt->recvonly) {
pvt->recvonly = 1;
rtp_change = NEED_HOLD;
}
}
/* Change native format to reflect information taken from OLC/OLCAck */
nativeformats_changed = 0;
if (pt != 128 && pvt->rtp) { /* Payload type is invalid, so try to use previously decided */
struct ast_rtp_payload_type rtptype = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(pvt->rtp), pt);
if (rtptype.asterisk_format) {
if (pvt->nativeformats != ast_format_to_old_bitfield(&rtptype.format)) {
pvt->nativeformats = ast_format_to_old_bitfield(&rtptype.format);
nativeformats_changed = 1;
}
}
} else if (h323debug)
ast_log(LOG_NOTICE, "Payload type is unknown, formats isn't changed\n");
/* Don't try to lock the channel if nothing changed */
if (nativeformats_changed || pvt->options.progress_audio || (rtp_change != NEED_NONE)) {
if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
struct ast_format_cap *pvt_native = ast_format_cap_alloc_nolock();
ast_format_cap_from_old_bitfield(pvt_native, pvt->nativeformats);
/* Re-build translation path only if native format(s) has been changed */
if (!(ast_format_cap_identical(pvt->owner->nativeformats, pvt_native))) {
if (h323debug) {
char tmp[256], tmp2[256];
ast_debug(1, "Native format changed to '%s' from '%s', read format is %s, write format is %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), pvt_native), ast_getformatname_multiple(tmp2, sizeof(tmp2), pvt->owner->nativeformats), ast_getformatname(&pvt->owner->readformat), ast_getformatname(&pvt->owner->writeformat));
}
ast_format_cap_copy(pvt->owner->nativeformats, pvt_native);
ast_set_read_format(pvt->owner, &pvt->owner->readformat);
ast_set_write_format(pvt->owner, &pvt->owner->writeformat);
}
if (pvt->options.progress_audio)
ast_queue_control(pvt->owner, AST_CONTROL_PROGRESS);
switch (rtp_change) {
case NEED_HOLD:
ast_queue_control(pvt->owner, AST_CONTROL_HOLD);
break;
case NEED_UNHOLD:
ast_queue_control(pvt->owner, AST_CONTROL_UNHOLD);
break;
default:
break;
}
ast_channel_unlock(pvt->owner);
pvt_native = ast_format_cap_destroy(pvt_native);
}
else {
if (pvt->options.progress_audio)
pvt->newcontrol = AST_CONTROL_PROGRESS;
else if (rtp_change == NEED_HOLD)
pvt->newcontrol = AST_CONTROL_HOLD;
else if (rtp_change == NEED_UNHOLD)
pvt->newcontrol = AST_CONTROL_UNHOLD;
if (h323debug)
ast_debug(1, "RTP connection preparation for %s is pending...\n", token);
}
}
ast_mutex_unlock(&pvt->lock);
if (h323debug)
ast_debug(1, "RTP connection prepared for %s\n", token);
return;
}
/*! \brief
* Call-back function to signal asterisk that the channel has been answered
* Returns nothing
*/
static void connection_made(unsigned call_reference, const char *token)
{
struct oh323_pvt *pvt;
if (h323debug)
ast_debug(1, "Call %s answered\n", token);
pvt = find_call_locked(call_reference, token);
if (!pvt) {
ast_log(LOG_ERROR, "Something is wrong: connection\n");
return;
}
/* Inform asterisk about remote party connected only on outgoing calls */
if (!pvt->outgoing) {
ast_mutex_unlock(&pvt->lock);
return;
}
/* Do not send ANSWER message more than once */
if (!pvt->connection_established) {
pvt->connection_established = 1;
update_state(pvt, -1, AST_CONTROL_ANSWER);
}
ast_mutex_unlock(&pvt->lock);
return;
}
static int progress(unsigned call_reference, const char *token, int inband)
{
struct oh323_pvt *pvt;
if (h323debug)
ast_debug(1, "Received ALERT/PROGRESS message for %s tones\n", (inband ? "inband" : "self-generated"));
pvt = find_call_locked(call_reference, token);
if (!pvt) {
ast_log(LOG_ERROR, "Private structure not found in progress.\n");
return -1;
}
if (!pvt->owner) {
ast_mutex_unlock(&pvt->lock);
ast_log(LOG_ERROR, "No Asterisk channel associated with private structure.\n");
return -1;
}
update_state(pvt, -1, (inband ? AST_CONTROL_PROGRESS : AST_CONTROL_RINGING));
ast_mutex_unlock(&pvt->lock);
return 0;
}
/*! \brief
* Call-back function for incoming calls
*
* Returns 1 on success
*/
static call_options_t *setup_incoming_call(call_details_t *cd)
{
struct oh323_pvt *pvt;
struct oh323_user *user = NULL;
struct oh323_alias *alias = NULL;
if (h323debug)
ast_debug(1, "Setting up incoming call for %s\n", cd->call_token);
/* allocate the call*/
pvt = oh323_alloc(cd->call_reference);
if (!pvt) {
ast_log(LOG_ERROR, "Unable to allocate private structure, this is bad.\n");
cleanup_call_details(cd);
return NULL;
}
/* Populate the call details in the private structure */
memcpy(&pvt->cd, cd, sizeof(pvt->cd));
memcpy(&pvt->options, &global_options, sizeof(pvt->options));
pvt->jointcapability = pvt->options.capability;
if (h323debug) {
ast_verb(3, "Setting up Call\n");
ast_verb(3, " \tCall token: [%s]\n", pvt->cd.call_token);
ast_verb(3, " \tCalling party name: [%s]\n", pvt->cd.call_source_name);
ast_verb(3, " \tCalling party number: [%s]\n", pvt->cd.call_source_e164);
ast_verb(3, " \tCalled party name: [%s]\n", pvt->cd.call_dest_alias);
ast_verb(3, " \tCalled party number: [%s]\n", pvt->cd.call_dest_e164);
if (pvt->cd.redirect_reason >= 0)
ast_verb(3, " \tRedirecting party number: [%s] (reason %d)\n", pvt->cd.redirect_number, pvt->cd.redirect_reason);
ast_verb(3, " \tCalling party IP: [%s]\n", pvt->cd.sourceIp);
}
/* Decide if we are allowing Gatekeeper routed calls*/
if ((!strcasecmp(cd->sourceIp, gatekeeper)) && (gkroute == -1) && !gatekeeper_disable) {
if (!ast_strlen_zero(cd->call_dest_e164)) {
ast_copy_string(pvt->exten, cd->call_dest_e164, sizeof(pvt->exten));
ast_copy_string(pvt->context, default_context, sizeof(pvt->context));
} else {
alias = find_alias(cd->call_dest_alias, 1);
if (!alias) {
ast_log(LOG_ERROR, "Call for %s rejected, alias not found\n", cd->call_dest_alias);
oh323_destroy(pvt);
return NULL;
}
ast_copy_string(pvt->exten, alias->name, sizeof(pvt->exten));
ast_copy_string(pvt->context, alias->context, sizeof(pvt->context));
}
} else {
/* Either this call is not from the Gatekeeper
or we are not allowing gk routed calls */
user = find_user(cd, 1);
if (!user) {
if (!acceptAnonymous) {
ast_log(LOG_NOTICE, "Anonymous call from '%s@%s' rejected\n", pvt->cd.call_source_aliases, pvt->cd.sourceIp);
oh323_destroy(pvt);
return NULL;
}
if (ast_strlen_zero(default_context)) {
ast_log(LOG_ERROR, "Call from '%s@%s' rejected due to no default context\n", pvt->cd.call_source_aliases, pvt->cd.sourceIp);
oh323_destroy(pvt);
return NULL;
}
ast_copy_string(pvt->context, default_context, sizeof(pvt->context));
if (!ast_strlen_zero(pvt->cd.call_dest_e164)) {
ast_copy_string(pvt->exten, cd->call_dest_e164, sizeof(pvt->exten));
} else {
ast_copy_string(pvt->exten, cd->call_dest_alias, sizeof(pvt->exten));
}
if (h323debug)
ast_debug(1, "Sending %s@%s to context [%s] extension %s\n", cd->call_source_aliases, cd->sourceIp, pvt->context, pvt->exten);
} else {
if (user->host) {
if (strcasecmp(cd->sourceIp, ast_inet_ntoa(user->addr.sin_addr))) {
if (ast_strlen_zero(user->context)) {
if (ast_strlen_zero(default_context)) {
ast_log(LOG_ERROR, "Call from '%s' rejected due to non-matching IP address (%s) and no default context\n", user->name, cd->sourceIp);
oh323_destroy(pvt);
ASTOBJ_UNREF(user, oh323_destroy_user);
return NULL;
}
ast_copy_string(pvt->context, default_context, sizeof(pvt->context));
} else {
ast_copy_string(pvt->context, user->context, sizeof(pvt->context));
}
pvt->exten[0] = 'i';
pvt->exten[1] = '\0';
ast_log(LOG_ERROR, "Call from '%s' rejected due to non-matching IP address (%s)s\n", user->name, cd->sourceIp);
oh323_destroy(pvt);
ASTOBJ_UNREF(user, oh323_destroy_user);
return NULL; /* XXX: Hmmm... Why to setup context if we drop connection immediately??? */
}
}
ast_copy_string(pvt->context, user->context, sizeof(pvt->context));
memcpy(&pvt->options, &user->options, sizeof(pvt->options));
pvt->jointcapability = pvt->options.capability;
if (!ast_strlen_zero(pvt->cd.call_dest_e164)) {
ast_copy_string(pvt->exten, cd->call_dest_e164, sizeof(pvt->exten));
} else {
ast_copy_string(pvt->exten, cd->call_dest_alias, sizeof(pvt->exten));
}
if (!ast_strlen_zero(user->accountcode)) {
ast_copy_string(pvt->accountcode, user->accountcode, sizeof(pvt->accountcode));
}
if (user->amaflags) {
pvt->amaflags = user->amaflags;
}
ASTOBJ_UNREF(user, oh323_destroy_user);
}
}
return &pvt->options;
}
/*! \brief
* Call-back function to start PBX when OpenH323 ready to serve incoming call
*
* Returns 1 on success
*/
static int answer_call(unsigned call_reference, const char *token)
{
struct oh323_pvt *pvt;
struct ast_channel *c = NULL;
enum {ext_original, ext_s, ext_i, ext_notexists} try_exten;
char tmp_exten[sizeof(pvt->exten)];
if (h323debug)
ast_debug(1, "Preparing Asterisk to answer for %s\n", token);
/* Find the call or allocate a private structure if call not found */
pvt = find_call_locked(call_reference, token);
if (!pvt) {
ast_log(LOG_ERROR, "Something is wrong: answer_call\n");
return 0;
}
/* Check if requested extension@context pair exists in the dialplan */
ast_copy_string(tmp_exten, pvt->exten, sizeof(tmp_exten));
/* Try to find best extension in specified context */
if ((tmp_exten[0] != '\0') && (tmp_exten[1] == '\0')) {
if (tmp_exten[0] == 's')
try_exten = ext_s;
else if (tmp_exten[0] == 'i')
try_exten = ext_i;
else
try_exten = ext_original;
} else
try_exten = ext_original;
do {
if (ast_exists_extension(NULL, pvt->context, tmp_exten, 1, NULL))
break;
switch (try_exten) {
case ext_original:
tmp_exten[0] = 's';
tmp_exten[1] = '\0';
try_exten = ext_s;
break;
case ext_s:
tmp_exten[0] = 'i';
try_exten = ext_i;
break;
case ext_i:
try_exten = ext_notexists;
break;
default:
break;
}
} while (try_exten != ext_notexists);
/* Drop the call if we don't have <exten>, s and i extensions */
if (try_exten == ext_notexists) {
ast_log(LOG_NOTICE, "Dropping call because extensions '%s', 's' and 'i' doesn't exists in context [%s]\n", pvt->exten, pvt->context);
ast_mutex_unlock(&pvt->lock);
h323_clear_call(token, AST_CAUSE_UNALLOCATED);
return 0;
} else if ((try_exten != ext_original) && (strcmp(pvt->exten, tmp_exten) != 0)) {
if (h323debug)
ast_debug(1, "Going to extension %s@%s because %s@%s isn't exists\n", tmp_exten, pvt->context, pvt->exten, pvt->context);
ast_copy_string(pvt->exten, tmp_exten, sizeof(pvt->exten));
}
/* allocate a channel and tell asterisk about it */
c = __oh323_new(pvt, AST_STATE_RINGING, pvt->cd.call_token, NULL);
/* And release when done */
ast_mutex_unlock(&pvt->lock);
if (!c) {
ast_log(LOG_ERROR, "Couldn't create channel. This is bad\n");
return 0;
}
return 1;
}
/*! \brief
* Call-back function to establish an outgoing H.323 call
*
* Returns 1 on success
*/
static int setup_outgoing_call(call_details_t *cd)
{
/* Use argument here or free it immediately */
cleanup_call_details(cd);
return 1;
}
/*! \brief
* Call-back function to signal asterisk that the channel is ringing
* Returns nothing
*/
static void chan_ringing(unsigned call_reference, const char *token)
{
struct oh323_pvt *pvt;
if (h323debug)
ast_debug(1, "Ringing on %s\n", token);
pvt = find_call_locked(call_reference, token);
if (!pvt) {
ast_log(LOG_ERROR, "Something is wrong: ringing\n");
return;
}
if (!pvt->owner) {
ast_mutex_unlock(&pvt->lock);
ast_log(LOG_ERROR, "Channel has no owner\n");
return;
}
update_state(pvt, AST_STATE_RINGING, AST_CONTROL_RINGING);
ast_mutex_unlock(&pvt->lock);
return;
}
/*! \brief
* Call-back function to cleanup communication
* Returns nothing,
*/
static void cleanup_connection(unsigned call_reference, const char *call_token)
{
struct oh323_pvt *pvt;
if (h323debug)
ast_debug(1, "Cleaning connection to %s\n", call_token);
while (1) {
pvt = find_call_locked(call_reference, call_token);
if (!pvt) {
if (h323debug)
ast_debug(1, "No connection for %s\n", call_token);
return;
}
if (!pvt->owner || !ast_channel_trylock(pvt->owner))
break;
#if 1
ast_log(LOG_NOTICE, "Avoiding H.323 destory deadlock on %s\n", call_token);
#ifdef DEBUG_THREADS
/* XXX to be completed
* If we want to print more info on who is holding the lock,
* implement the relevant code in lock.h and use the routines
* supplied there.
*/
#endif
#endif
ast_mutex_unlock(&pvt->lock);
usleep(1);
}
if (pvt->rtp) {
/* Immediately stop RTP */
ast_rtp_instance_destroy(pvt->rtp);
pvt->rtp = NULL;
}
/* Free dsp used for in-band DTMF detection */
if (pvt->vad) {
ast_dsp_free(pvt->vad);
pvt->vad = NULL;
}
cleanup_call_details(&pvt->cd);
pvt->alreadygone = 1;
/* Send hangup */
if (pvt->owner) {
pvt->owner->_softhangup |= AST_SOFTHANGUP_DEV;
ast_queue_hangup(pvt->owner);
ast_channel_unlock(pvt->owner);
}
ast_mutex_unlock(&pvt->lock);
if (h323debug)
ast_debug(1, "Connection to %s cleaned\n", call_token);
return;
}
static void hangup_connection(unsigned int call_reference, const char *token, int cause)
{
struct oh323_pvt *pvt;
if (h323debug)
ast_debug(1, "Hanging up connection to %s with cause %d\n", token, cause);
pvt = find_call_locked(call_reference, token);
if (!pvt) {
if (h323debug)
ast_debug(1, "Connection to %s already cleared\n", token);
return;
}
if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
pvt->owner->_softhangup |= AST_SOFTHANGUP_DEV;
pvt->owner->hangupcause = pvt->hangupcause = cause;
ast_queue_hangup_with_cause(pvt->owner, cause);
ast_channel_unlock(pvt->owner);
}
else {
pvt->needhangup = 1;
pvt->hangupcause = cause;
if (h323debug)
ast_debug(1, "Hangup for %s is pending\n", token);
}
ast_mutex_unlock(&pvt->lock);
}
static void set_dtmf_payload(unsigned call_reference, const char *token, int payload, int is_cisco)
{
struct oh323_pvt *pvt;
if (h323debug)
ast_debug(1, "Setting %s DTMF payload to %d on %s\n", (is_cisco ? "Cisco" : "RFC2833"), payload, token);
pvt = find_call_locked(call_reference, token);
if (!pvt) {
return;
}
if (pvt->rtp) {
ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, payload, "audio", (is_cisco ? "cisco-telephone-event" : "telephone-event"), 0);
}
pvt->dtmf_pt[is_cisco ? 1 : 0] = payload;
ast_mutex_unlock(&pvt->lock);
if (h323debug)
ast_debug(1, "DTMF payload on %s set to %d\n", token, payload);
}
static void set_peer_capabilities(unsigned call_reference, const char *token, int capabilities, struct ast_codec_pref *prefs)
{
struct oh323_pvt *pvt;
if (h323debug)
ast_debug(1, "Got remote capabilities from connection %s\n", token);
pvt = find_call_locked(call_reference, token);
if (!pvt)
return;
pvt->peercapability = capabilities;
pvt->jointcapability = pvt->options.capability & capabilities;
if (prefs) {
memcpy(&pvt->peer_prefs, prefs, sizeof(pvt->peer_prefs));
if (h323debug) {
int i;
for (i = 0; i < 32; ++i) {
if (!prefs->order[i])
break;
ast_debug(1, "prefs[%d]=%s:%d\n", i, (prefs->order[i] ? ast_getformatname(&prefs->formats[i]) : "<none>"), prefs->framing[i]);
}
}
if (pvt->rtp) {
if (pvt->options.autoframing) {
ast_debug(2, "Autoframing option set, using peer's packetization settings\n");
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->peer_prefs);
} else {
ast_debug(2, "Autoframing option not set, ignoring peer's packetization settings\n");
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->options.prefs);
}
}
}
ast_mutex_unlock(&pvt->lock);
}
static void set_local_capabilities(unsigned call_reference, const char *token)
{
struct oh323_pvt *pvt;
int capability, dtmfmode, pref_codec;
struct ast_codec_pref prefs;
if (h323debug)
ast_debug(1, "Setting capabilities for connection %s\n", token);
pvt = find_call_locked(call_reference, token);
if (!pvt)
return;
capability = (pvt->jointcapability) ? pvt->jointcapability : pvt->options.capability;
dtmfmode = pvt->options.dtmfmode;
prefs = pvt->options.prefs;
pref_codec = pvt->pref_codec;
ast_mutex_unlock(&pvt->lock);
h323_set_capabilities(token, capability, dtmfmode, &prefs, pref_codec);
if (h323debug) {
int i;
for (i = 0; i < 32; i++) {
if (!prefs.order[i])
break;
ast_debug(1, "local prefs[%d]=%s:%d\n", i, (prefs.order[i] ? ast_getformatname(&prefs.formats[i]) : "<none>"), prefs.framing[i]);
}
ast_debug(1, "Capabilities for connection %s is set\n", token);
}
}
static void remote_hold(unsigned call_reference, const char *token, int is_hold)
{
struct oh323_pvt *pvt;
if (h323debug)
ast_debug(1, "Setting %shold status for connection %s\n", (is_hold ? "" : "un"), token);
pvt = find_call_locked(call_reference, token);
if (!pvt)
return;
if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
if (is_hold)
ast_queue_control(pvt->owner, AST_CONTROL_HOLD);
else
ast_queue_control(pvt->owner, AST_CONTROL_UNHOLD);
ast_channel_unlock(pvt->owner);
}
else {
if (is_hold)
pvt->newcontrol = AST_CONTROL_HOLD;
else
pvt->newcontrol = AST_CONTROL_UNHOLD;
}
ast_mutex_unlock(&pvt->lock);
}
static void *do_monitor(void *data)
{
int res;
int reloading;
struct oh323_pvt *oh323 = NULL;
for(;;) {
/* Check for a reload request */
ast_mutex_lock(&h323_reload_lock);
reloading = h323_reloading;
h323_reloading = 0;
ast_mutex_unlock(&h323_reload_lock);
if (reloading) {
ast_verb(1, "Reloading H.323\n");
h323_do_reload();
}
/* Check for interfaces needing to be killed */
if (!ast_mutex_trylock(&iflock)) {
#if 1
do {
for (oh323 = iflist; oh323; oh323 = oh323->next) {
if (!ast_mutex_trylock(&oh323->lock)) {
if (oh323->needdestroy) {
__oh323_destroy(oh323);
break;
}
ast_mutex_unlock(&oh323->lock);
}
}
} while (/*oh323*/ 0);
#else
restartsearch:
oh323 = iflist;
while(oh323) {
if (!ast_mutex_trylock(&oh323->lock)) {
if (oh323->needdestroy) {
__oh323_destroy(oh323);
goto restartsearch;
}
ast_mutex_unlock(&oh323->lock);
oh323 = oh323->next;
}
}
#endif
ast_mutex_unlock(&iflock);
} else
oh323 = (struct oh323_pvt *)1; /* Force fast loop */
pthread_testcancel();
/* Wait for sched or io */
res = ast_sched_wait(sched);
if ((res < 0) || (res > 1000)) {
res = 1000;
}
/* Do not wait if some channel(s) is destroyed, probably, more available too */
if (oh323)
res = 1;
res = ast_io_wait(io, res);
pthread_testcancel();
ast_mutex_lock(&monlock);
if (res >= 0) {
ast_sched_runq(sched);
}
ast_mutex_unlock(&monlock);
}
/* Never reached */
return NULL;
}
static int restart_monitor(void)
{
/* If we're supposed to be stopped -- stay stopped */
if (ast_mutex_lock(&monlock)) {
ast_log(LOG_WARNING, "Unable to lock monitor\n");
return -1;
}
if (monitor_thread == AST_PTHREADT_STOP) {
ast_mutex_unlock(&monlock);
return 0;
}
if (monitor_thread == pthread_self()) {
ast_mutex_unlock(&monlock);
ast_log(LOG_WARNING, "Cannot kill myself\n");
return -1;
}
if (monitor_thread && (monitor_thread != AST_PTHREADT_NULL)) {
/* Wake up the thread */
pthread_kill(monitor_thread, SIGURG);
} else {
/* Start a new monitor */
if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
monitor_thread = AST_PTHREADT_NULL;
ast_mutex_unlock(&monlock);
ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
return -1;
}
}
ast_mutex_unlock(&monlock);
return 0;
}
static char *handle_cli_h323_set_trace(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "h323 set trace [on|off]";
e->usage =
"Usage: h323 set trace (on|off|<trace level>)\n"
" Enable/Disable H.323 stack tracing for debugging purposes\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (!strcasecmp(a->argv[3], "off")) {
h323_debug(0, 0);
ast_cli(a->fd, "H.323 Trace Disabled\n");
} else if (!strcasecmp(a->argv[3], "on")) {
h323_debug(1, 1);
ast_cli(a->fd, "H.323 Trace Enabled\n");
} else {
int tracelevel = atoi(a->argv[3]);
h323_debug(1, tracelevel);
ast_cli(a->fd, "H.323 Trace Enabled (Trace Level: %d)\n", tracelevel);
}
return CLI_SUCCESS;
}
static char *handle_cli_h323_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "h323 set debug [on|off]";
e->usage =
"Usage: h323 set debug [on|off]\n"
" Enable/Disable H.323 debugging output\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (strcasecmp(a->argv[3], "on") && strcasecmp(a->argv[3], "off"))
return CLI_SHOWUSAGE;
h323debug = (strcasecmp(a->argv[3], "on")) ? 0 : 1;
ast_cli(a->fd, "H.323 Debugging %s\n", h323debug ? "Enabled" : "Disabled");
return CLI_SUCCESS;
}
static char *handle_cli_h323_cycle_gk(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "h323 cycle gk";
e->usage =
"Usage: h323 cycle gk\n"
" Manually re-register with the Gatekeper (Currently Disabled)\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
h323_gk_urq();
/* Possibly register with a GK */
if (!gatekeeper_disable) {
if (h323_set_gk(gatekeeper_discover, gatekeeper, secret)) {
ast_log(LOG_ERROR, "Gatekeeper registration failed.\n");
}
}
return CLI_SUCCESS;
}
static char *handle_cli_h323_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "h323 hangup";
e->usage =
"Usage: h323 hangup <token>\n"
" Manually try to hang up the call identified by <token>\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
if (h323_soft_hangup(a->argv[2])) {
ast_verb(3, "Hangup succeeded on %s\n", a->argv[2]);
} else {
ast_verb(3, "Hangup failed for %s\n", a->argv[2]);
}
return CLI_SUCCESS;
}
static char *handle_cli_h323_show_tokens(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "h323 show tokens";
e->usage =
"Usage: h323 show tokens\n"
" Print out all active call tokens\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
h323_show_tokens();
return CLI_SUCCESS;
}
static char *handle_cli_h323_show_version(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "h323 show version";
e->usage =
"Usage: h323 show version\n"
" Show the version of the H.323 library in use\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
h323_show_version();
return CLI_SUCCESS;
}
static struct ast_cli_entry cli_h323[] = {
AST_CLI_DEFINE(handle_cli_h323_set_trace, "Enable/Disable H.323 Stack Tracing"),
AST_CLI_DEFINE(handle_cli_h323_set_debug, "Enable/Disable H.323 Debugging"),
AST_CLI_DEFINE(handle_cli_h323_cycle_gk, "Manually re-register with the Gatekeper"),
AST_CLI_DEFINE(handle_cli_h323_hangup, "Manually try to hang up a call"),
AST_CLI_DEFINE(handle_cli_h323_show_tokens, "Show all active call tokens"),
AST_CLI_DEFINE(handle_cli_h323_show_version, "Show the version of the H.323 library in use"),
};
static void delete_users(void)
{
int pruned = 0;
/* Delete all users */
ASTOBJ_CONTAINER_WRLOCK(&userl);
ASTOBJ_CONTAINER_TRAVERSE(&userl, 1, do {
ASTOBJ_RDLOCK(iterator);
ASTOBJ_MARK(iterator);
++pruned;
ASTOBJ_UNLOCK(iterator);
} while (0) );
if (pruned) {
ASTOBJ_CONTAINER_PRUNE_MARKED(&userl, oh323_destroy_user);
}
ASTOBJ_CONTAINER_UNLOCK(&userl);
ASTOBJ_CONTAINER_WRLOCK(&peerl);
ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do {
ASTOBJ_RDLOCK(iterator);
ASTOBJ_MARK(iterator);
ASTOBJ_UNLOCK(iterator);
} while (0) );
ASTOBJ_CONTAINER_UNLOCK(&peerl);
}
static void delete_aliases(void)
{
int pruned = 0;
/* Delete all aliases */
ASTOBJ_CONTAINER_WRLOCK(&aliasl);
ASTOBJ_CONTAINER_TRAVERSE(&aliasl, 1, do {
ASTOBJ_RDLOCK(iterator);
ASTOBJ_MARK(iterator);
++pruned;
ASTOBJ_UNLOCK(iterator);
} while (0) );
if (pruned) {
ASTOBJ_CONTAINER_PRUNE_MARKED(&aliasl, oh323_destroy_alias);
}
ASTOBJ_CONTAINER_UNLOCK(&aliasl);
}
static void prune_peers(void)
{
/* Prune peers who still are supposed to be deleted */
ASTOBJ_CONTAINER_PRUNE_MARKED(&peerl, oh323_destroy_peer);
}
static int reload_config(int is_reload)
{
struct ast_config *cfg, *ucfg;
struct ast_variable *v;
struct oh323_peer *peer = NULL;
struct oh323_user *user = NULL;
struct oh323_alias *alias = NULL;
struct ast_hostent ahp; struct hostent *hp;
char *cat;
const char *utype;
int is_user, is_peer, is_alias;
char _gatekeeper[100];
int gk_discover, gk_disable, gk_changed;
struct ast_flags config_flags = { is_reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
cfg = ast_config_load(config, config_flags);
/* We *must* have a config file otherwise stop immediately */
if (!cfg) {
ast_log(LOG_NOTICE, "Unable to load config %s, H.323 disabled\n", config);
return 1;
} else if (cfg == CONFIG_STATUS_FILEUNCHANGED) {
ucfg = ast_config_load("users.conf", config_flags);
if (ucfg == CONFIG_STATUS_FILEUNCHANGED) {
return 0;
} else if (ucfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file users.conf is in an invalid format. Aborting.\n");
return 0;
}
ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
if ((cfg = ast_config_load(config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", config);
ast_config_destroy(ucfg);
return 0;
}
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", config);
return 0;
} else {
ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
if ((ucfg = ast_config_load("users.conf", config_flags)) == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file users.conf is in an invalid format. Aborting.\n");
ast_config_destroy(cfg);
return 0;
}
}
if (is_reload) {
delete_users();
delete_aliases();
prune_peers();
}
/* fire up the H.323 Endpoint */
if (!h323_end_point_exist()) {
h323_end_point_create();
}
ast_copy_string(_gatekeeper, gatekeeper, sizeof(_gatekeeper));
gk_discover = gatekeeper_discover;
gk_disable = gatekeeper_disable;
memset(&bindaddr, 0, sizeof(bindaddr));
memset(&global_options, 0, sizeof(global_options));
global_options.fastStart = 1;
global_options.h245Tunneling = 1;
global_options.dtmfcodec[0] = H323_DTMF_RFC2833_PT;
global_options.dtmfcodec[1] = H323_DTMF_CISCO_PT;
global_options.dtmfmode = 0;
global_options.holdHandling = 0;
global_options.capability = GLOBAL_CAPABILITY;
global_options.bridge = 1; /* Do native bridging by default */
global_options.autoframing = 0;
strcpy(default_context, "default");
h323_signalling_port = 1720;
gatekeeper_disable = 1;
gatekeeper_discover = 0;
gkroute = 0;
userbyalias = 1;
acceptAnonymous = 1;
tos = 0;
cos = 0;
/* Copy the default jb config over global_jbconf */
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
if (ucfg) {
struct ast_variable *gen;
int genhas_h323;
const char *has_h323;
genhas_h323 = ast_true(ast_variable_retrieve(ucfg, "general", "hash323"));
gen = ast_variable_browse(ucfg, "general");
for (cat = ast_category_browse(ucfg, NULL); cat; cat = ast_category_browse(ucfg, cat)) {
if (strcasecmp(cat, "general")) {
has_h323 = ast_variable_retrieve(ucfg, cat, "hash323");
if (ast_true(has_h323) || (!has_h323 && genhas_h323)) {
user = build_user(cat, gen, ast_variable_browse(ucfg, cat), 0);
if (user) {
ASTOBJ_CONTAINER_LINK(&userl, user);
ASTOBJ_UNREF(user, oh323_destroy_user);
}
peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0);
if (peer) {
ASTOBJ_CONTAINER_LINK(&peerl, peer);
ASTOBJ_UNREF(peer, oh323_destroy_peer);
}
}
}
}
ast_config_destroy(ucfg);
}
for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
/* handle jb conf */
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
continue;
/* Create the interface list */
if (!strcasecmp(v->name, "port")) {
h323_signalling_port = (int)strtol(v->value, NULL, 10);
} else if (!strcasecmp(v->name, "bindaddr")) {
if (!(hp = ast_gethostbyname(v->value, &ahp))) {
ast_log(LOG_WARNING, "Invalid address: %s\n", v->value);
} else {
memcpy(&bindaddr.sin_addr, hp->h_addr, sizeof(bindaddr.sin_addr));
}
} else if (!strcasecmp(v->name, "tos")) { /* Needs to be removed in next release */
ast_log(LOG_WARNING, "The \"tos\" setting is deprecated in this version of Asterisk. Please change to \"tos_audio\".\n");
if (ast_str2tos(v->value, &tos)) {
ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
}
} else if (!strcasecmp(v->name, "tos_audio")) {
if (ast_str2tos(v->value, &tos)) {
ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
}
} else if (!strcasecmp(v->name, "cos")) {
ast_log(LOG_WARNING, "The \"cos\" setting is deprecated in this version of Asterisk. Please change to \"cos_audio\".\n");
if (ast_str2cos(v->value, &cos)) {
ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
}
} else if (!strcasecmp(v->name, "cos_audio")) {
if (ast_str2cos(v->value, &cos)) {
ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
}
} else if (!strcasecmp(v->name, "gatekeeper")) {
if (!strcasecmp(v->value, "DISABLE")) {
gatekeeper_disable = 1;
} else if (!strcasecmp(v->value, "DISCOVER")) {
gatekeeper_disable = 0;
gatekeeper_discover = 1;
} else {
gatekeeper_disable = 0;
ast_copy_string(gatekeeper, v->value, sizeof(gatekeeper));
}
} else if (!strcasecmp(v->name, "secret")) {
ast_copy_string(secret, v->value, sizeof(secret));
} else if (!strcasecmp(v->name, "AllowGKRouted")) {
gkroute = ast_true(v->value);
} else if (!strcasecmp(v->name, "context")) {
ast_copy_string(default_context, v->value, sizeof(default_context));
ast_verb(2, "Setting default context to %s\n", default_context);
} else if (!strcasecmp(v->name, "UserByAlias")) {
userbyalias = ast_true(v->value);
} else if (!strcasecmp(v->name, "AcceptAnonymous")) {
acceptAnonymous = ast_true(v->value);
} else if (!update_common_options(v, &global_options)) {
/* dummy */
}
}
if (!global_options.dtmfmode)
global_options.dtmfmode = H323_DTMF_RFC2833;
if (global_options.holdHandling == ~0)
global_options.holdHandling = 0;
else if (!global_options.holdHandling)
global_options.holdHandling = H323_HOLD_H450;
for (cat = ast_category_browse(cfg, NULL); cat; cat = ast_category_browse(cfg, cat)) {
if (strcasecmp(cat, "general")) {
utype = ast_variable_retrieve(cfg, cat, "type");
if (utype) {
is_user = is_peer = is_alias = 0;
if (!strcasecmp(utype, "user"))
is_user = 1;
else if (!strcasecmp(utype, "peer"))
is_peer = 1;
else if (!strcasecmp(utype, "friend"))
is_user = is_peer = 1;
else if (!strcasecmp(utype, "h323") || !strcasecmp(utype, "alias"))
is_alias = 1;
else {
ast_log(LOG_WARNING, "Unknown type '%s' for '%s' in %s\n", utype, cat, config);
continue;
}
if (is_user) {
user = build_user(cat, ast_variable_browse(cfg, cat), NULL, 0);
if (user) {
ASTOBJ_CONTAINER_LINK(&userl, user);
ASTOBJ_UNREF(user, oh323_destroy_user);
}
}
if (is_peer) {
peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0);
if (peer) {
ASTOBJ_CONTAINER_LINK(&peerl, peer);
ASTOBJ_UNREF(peer, oh323_destroy_peer);
}
}
if (is_alias) {
alias = build_alias(cat, ast_variable_browse(cfg, cat), NULL, 0);
if (alias) {
ASTOBJ_CONTAINER_LINK(&aliasl, alias);
ASTOBJ_UNREF(alias, oh323_destroy_alias);
}
}
} else {
ast_log(LOG_WARNING, "Section '%s' lacks type\n", cat);
}
}
}
ast_config_destroy(cfg);
/* Register our H.323 aliases if any*/
ASTOBJ_CONTAINER_WRLOCK(&aliasl);
ASTOBJ_CONTAINER_TRAVERSE(&aliasl, 1, do {
ASTOBJ_RDLOCK(iterator);
if (h323_set_alias(iterator)) {
ast_log(LOG_ERROR, "Alias %s rejected by endpoint\n", alias->name);
ASTOBJ_UNLOCK(iterator);
continue;
}
ASTOBJ_UNLOCK(iterator);
} while (0) );
ASTOBJ_CONTAINER_UNLOCK(&aliasl);
/* Don't touch GK if nothing changed because URQ will drop all existing calls */
gk_changed = 0;
if (gatekeeper_disable != gk_disable)
gk_changed = is_reload;
else if(!gatekeeper_disable && (gatekeeper_discover != gk_discover))
gk_changed = is_reload;
else if(!gatekeeper_disable && (strncmp(_gatekeeper, gatekeeper, sizeof(_gatekeeper)) != 0))
gk_changed = is_reload;
if (gk_changed) {
if(!gk_disable)
h323_gk_urq();
if (!gatekeeper_disable) {
if (h323_set_gk(gatekeeper_discover, gatekeeper, secret)) {
ast_log(LOG_ERROR, "Gatekeeper registration failed.\n");
gatekeeper_disable = 1;
}
}
}
return 0;
}
static int h323_reload(void)
{
ast_mutex_lock(&h323_reload_lock);
if (h323_reloading) {
ast_verbose("Previous H.323 reload not yet done\n");
} else {
h323_reloading = 1;
}
ast_mutex_unlock(&h323_reload_lock);
restart_monitor();
return 0;
}
static char *handle_cli_h323_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "h323 reload";
e->usage =
"Usage: h323 reload\n"
" Reloads H.323 configuration from h323.conf\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 2)
return CLI_SHOWUSAGE;
h323_reload();
return CLI_SUCCESS;
}
static int h323_do_reload(void)
{
reload_config(1);
return 0;
}
static int reload(void)
{
if (!sched || !io) {
ast_log(LOG_NOTICE, "Unload and load chan_h323.so again in order to receive configuration changes.\n");
return 0;
}
return h323_reload();
}
static struct ast_cli_entry cli_h323_reload =
AST_CLI_DEFINE(handle_cli_h323_reload, "Reload H.323 configuration");
static enum ast_rtp_glue_result oh323_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct oh323_pvt *pvt;
enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;
if (!(pvt = (struct oh323_pvt *)chan->tech_pvt))
return AST_RTP_GLUE_RESULT_FORBID;
ast_mutex_lock(&pvt->lock);
*instance = pvt->rtp ? ao2_ref(pvt->rtp, +1), pvt->rtp : NULL;
#if 0
if (pvt->options.bridge) {
res = AST_RTP_GLUE_RESULT_REMOTE;
}
#endif
ast_mutex_unlock(&pvt->lock);
return res;
}
static char *convertcap(struct ast_format *format)
{
switch (format->id) {
case AST_FORMAT_G723_1:
return "G.723";
case AST_FORMAT_GSM:
return "GSM";
case AST_FORMAT_ULAW:
return "ULAW";
case AST_FORMAT_ALAW:
return "ALAW";
case AST_FORMAT_G722:
return "G.722";
case AST_FORMAT_ADPCM:
return "G.728";
case AST_FORMAT_G729A:
return "G.729";
case AST_FORMAT_SPEEX:
return "SPEEX";
case AST_FORMAT_ILBC:
return "ILBC";
default:
ast_log(LOG_NOTICE, "Don't know how to deal with mode %s\n", ast_getformatname(format));
return NULL;
}
}
static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, const struct ast_format_cap *codecs, int nat_active)
{
/* XXX Deal with Video */
struct oh323_pvt *pvt;
struct sockaddr_in them = { 0, };
struct sockaddr_in us = { 0, };
char *mode;
if (!rtp) {
return 0;
}
mode = convertcap(&chan->writeformat);
pvt = (struct oh323_pvt *) chan->tech_pvt;
if (!pvt) {
ast_log(LOG_ERROR, "No Private Structure, this is bad\n");
return -1;
}
{
struct ast_sockaddr tmp;
ast_rtp_instance_get_remote_address(rtp, &tmp);
ast_sockaddr_to_sin(&tmp, &them);
ast_rtp_instance_get_local_address(rtp, &tmp);
ast_sockaddr_to_sin(&tmp, &us);
}
#if 0 /* Native bridge still isn't ready */
h323_native_bridge(pvt->cd.call_token, ast_inet_ntoa(them.sin_addr), mode);
#endif
return 0;
}
static struct ast_rtp_glue oh323_rtp_glue = {
.type = "H323",
.get_rtp_info = oh323_get_rtp_peer,
.update_peer = oh323_set_rtp_peer,
};
static enum ast_module_load_result load_module(void)
{
int res;
if (!(oh323_tech.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_FAILURE;
}
ast_format_cap_add_all_by_type(oh323_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
h323debug = 0;
sched = ast_sched_context_create();
if (!sched) {
ast_log(LOG_WARNING, "Unable to create schedule context\n");
return AST_MODULE_LOAD_FAILURE;
}
io = io_context_create();
if (!io) {
ast_log(LOG_WARNING, "Unable to create I/O context\n");
return AST_MODULE_LOAD_FAILURE;
}
ast_cli_register(&cli_h323_reload);
ASTOBJ_CONTAINER_INIT(&userl);
ASTOBJ_CONTAINER_INIT(&peerl);
ASTOBJ_CONTAINER_INIT(&aliasl);
res = reload_config(0);
if (res) {
/* No config entry */
ast_log(LOG_NOTICE, "Unload and load chan_h323.so again in order to receive configuration changes.\n");
ast_cli_unregister(&cli_h323_reload);
io_context_destroy(io);
io = NULL;
ast_sched_context_destroy(sched);
sched = NULL;
ASTOBJ_CONTAINER_DESTROY(&userl);
ASTOBJ_CONTAINER_DESTROY(&peerl);
ASTOBJ_CONTAINER_DESTROY(&aliasl);
return AST_MODULE_LOAD_DECLINE;
} else {
/* Make sure we can register our channel type */
if (ast_channel_register(&oh323_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'H323'\n");
ast_cli_unregister(&cli_h323_reload);
h323_end_process();
io_context_destroy(io);
ast_sched_context_destroy(sched);
ASTOBJ_CONTAINER_DESTROYALL(&userl, oh323_destroy_user);
ASTOBJ_CONTAINER_DESTROY(&userl);
ASTOBJ_CONTAINER_DESTROYALL(&peerl, oh323_destroy_peer);
ASTOBJ_CONTAINER_DESTROY(&peerl);
ASTOBJ_CONTAINER_DESTROYALL(&aliasl, oh323_destroy_alias);
ASTOBJ_CONTAINER_DESTROY(&aliasl);
return AST_MODULE_LOAD_FAILURE;
}
ast_cli_register_multiple(cli_h323, sizeof(cli_h323) / sizeof(struct ast_cli_entry));
ast_rtp_glue_register(&oh323_rtp_glue);
/* Register our callback functions */
h323_callback_register(setup_incoming_call,
setup_outgoing_call,
external_rtp_create,
setup_rtp_connection,
cleanup_connection,
chan_ringing,
connection_made,
receive_digit,
answer_call,
progress,
set_dtmf_payload,
hangup_connection,
set_local_capabilities,
set_peer_capabilities,
remote_hold);
/* start the h.323 listener */
if (h323_start_listener(h323_signalling_port, bindaddr)) {
ast_log(LOG_ERROR, "Unable to create H323 listener.\n");
ast_rtp_glue_unregister(&oh323_rtp_glue);
ast_cli_unregister_multiple(cli_h323, sizeof(cli_h323) / sizeof(struct ast_cli_entry));
ast_cli_unregister(&cli_h323_reload);
h323_end_process();
io_context_destroy(io);
ast_sched_context_destroy(sched);
ASTOBJ_CONTAINER_DESTROYALL(&userl, oh323_destroy_user);
ASTOBJ_CONTAINER_DESTROY(&userl);
ASTOBJ_CONTAINER_DESTROYALL(&peerl, oh323_destroy_peer);
ASTOBJ_CONTAINER_DESTROY(&peerl);
ASTOBJ_CONTAINER_DESTROYALL(&aliasl, oh323_destroy_alias);
ASTOBJ_CONTAINER_DESTROY(&aliasl);
return AST_MODULE_LOAD_DECLINE;
}
/* Possibly register with a GK */
if (!gatekeeper_disable) {
if (h323_set_gk(gatekeeper_discover, gatekeeper, secret)) {
ast_log(LOG_ERROR, "Gatekeeper registration failed.\n");
gatekeeper_disable = 1;
res = AST_MODULE_LOAD_SUCCESS;
}
}
/* And start the monitor for the first time */
restart_monitor();
}
return res;
}
static int unload_module(void)
{
struct oh323_pvt *p, *pl;
/* unregister commands */
ast_cli_unregister_multiple(cli_h323, sizeof(cli_h323) / sizeof(struct ast_cli_entry));
ast_cli_unregister(&cli_h323_reload);
ast_channel_unregister(&oh323_tech);
ast_rtp_glue_unregister(&oh323_rtp_glue);
if (!ast_mutex_lock(&iflock)) {
/* hangup all interfaces if they have an owner */
p = iflist;
while(p) {
if (p->owner) {
ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
}
p = p->next;
}
iflist = NULL;
ast_mutex_unlock(&iflock);
} else {
ast_log(LOG_WARNING, "Unable to lock the interface list\n");
return -1;
}
if (!ast_mutex_lock(&monlock)) {
if ((monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
if (monitor_thread != pthread_self()) {
pthread_cancel(monitor_thread);
}
pthread_kill(monitor_thread, SIGURG);
pthread_join(monitor_thread, NULL);
}
monitor_thread = AST_PTHREADT_STOP;
ast_mutex_unlock(&monlock);
} else {
ast_log(LOG_WARNING, "Unable to lock the monitor\n");
return -1;
}
if (!ast_mutex_lock(&iflock)) {
/* destroy all the interfaces and free their memory */
p = iflist;
while(p) {
pl = p;
p = p->next;
/* free associated memory */
ast_mutex_destroy(&pl->lock);
ast_free(pl);
}
iflist = NULL;
ast_mutex_unlock(&iflock);
} else {
ast_log(LOG_WARNING, "Unable to lock the interface list\n");
return -1;
}
if (!gatekeeper_disable)
h323_gk_urq();
h323_end_process();
if (io)
io_context_destroy(io);
if (sched)
ast_sched_context_destroy(sched);
ASTOBJ_CONTAINER_DESTROYALL(&userl, oh323_destroy_user);
ASTOBJ_CONTAINER_DESTROY(&userl);
ASTOBJ_CONTAINER_DESTROYALL(&peerl, oh323_destroy_peer);
ASTOBJ_CONTAINER_DESTROY(&peerl);
ASTOBJ_CONTAINER_DESTROYALL(&aliasl, oh323_destroy_alias);
ASTOBJ_CONTAINER_DESTROY(&aliasl);
oh323_tech.capabilities = ast_format_cap_destroy(oh323_tech.capabilities);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "The NuFone Network's OpenH323 Channel Driver",
.load = load_module,
.unload = unload_module,
.reload = reload,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
);