chan_h323.c

Asterisk中信道部分的源码 。。。。

	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 = digit,
				.samples = duration * 8,
				.len = duration,
				.src = "SEND_DIGIT",
			};
			if (digit == ' ') {		/* signalUpdate message */
				f.subclass = 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 = pvt->curDTMF;
					f.samples = f.len = 0;
					ast_queue_frame(pvt->owner, &f);
					/* Restore values */
					f.subclass = 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;
}

/**
  * Callback function used to inform the H.323 stack of the local rtp ip/port details
  *
  * 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 = (struct rtp_info *)malloc(sizeof(struct rtp_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) {
		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);
		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 */
	ast_rtp_get_us(pvt->rtp, &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_log(LOG_DEBUG, "Sending RTP 'US' %s:%d\n", info->addr, info->port);
	return info;
}

/**
 * Definition taken from rtp.c for rtpPayloadType because we need it here.
 */
struct rtpPayloadType {
	int isAstFormat;	/* whether the following code is an AST_FORMAT */
	int code;
};

/**
  * 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;
	struct rtpPayloadType rtptype;
	int nativeformats_changed;
	enum { NEED_NONE, NEED_HOLD, NEED_UNHOLD } rtp_change = NEED_NONE;

	if (h323debug)
		ast_log(LOG_DEBUG, "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);

	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) {
		ast_rtp_set_peer(pvt->rtp, &them);
		if (pvt->recvonly) {
			pvt->recvonly = 0;
			rtp_change = NEED_UNHOLD;
		}
	} else {
		ast_rtp_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 */
		rtptype = ast_rtp_lookup_pt(pvt->rtp, pt);
		if (h323debug)
			ast_log(LOG_DEBUG, "Native format is set to %d from %d by RTP payload type %d\n", rtptype.code, pvt->nativeformats, pt);
		if (pvt->nativeformats != rtptype.code) {
			pvt->nativeformats = rtptype.code;
			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)) {
			/* Re-build translation path only if native format(s) has been changed */
			if (pvt->owner->nativeformats != pvt->nativeformats) {
				if (h323debug)
					ast_log(LOG_DEBUG, "Native format changed to %d from %d, read format is %d, write format is %d\n", pvt->nativeformats, pvt->owner->nativeformats, pvt->owner->readformat, pvt->owner->writeformat);
				pvt->owner->nativeformats = pvt->nativeformats;
				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);
		}
		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_log(LOG_DEBUG, "RTP connection preparation for %s is pending...\n", token);
		}
	}
	ast_mutex_unlock(&pvt->lock);

	if (h323debug)
		ast_log(LOG_DEBUG, "RTP connection prepared for %s\n", token);

	return;
}

/**
  *	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_log(LOG_DEBUG, "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_log(LOG_DEBUG, "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;
}

/**
 *  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_log(LOG_DEBUG, "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_verbose(VERBOSE_PREFIX_3 "Setting up Call\n");
		ast_verbose(VERBOSE_PREFIX_3 " \tCall token:  [%s]\n", pvt->cd.call_token);
		ast_verbose(VERBOSE_PREFIX_3 " \tCalling party name:  [%s]\n", pvt->cd.call_source_name);
		ast_verbose(VERBOSE_PREFIX_3 " \tCalling party number:  [%s]\n", pvt->cd.call_source_e164);
		ast_verbose(VERBOSE_PREFIX_3 " \tCalled party name:  [%s]\n", pvt->cd.call_dest_alias);
		ast_verbose(VERBOSE_PREFIX_3 " \tCalled party number:  [%s]\n", pvt->cd.call_dest_e164);
		if (pvt->cd.redirect_reason >= 0)
			ast_verbose(VERBOSE_PREFIX_3 " \tRedirecting party number:  [%s] (reason %d)\n", pvt->cd.redirect_number, pvt->cd.redirect_reason);
		ast_verbose(VERBOSE_PREFIX_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_log(LOG_DEBUG, "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;
}

/**
 * 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_log(LOG_DEBUG, "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