[asterisk-commits] oej: branch oej/earlyrtpfix r97624 - in /team/oej/earlyrtpfix: ./ apps/ chann...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jan 9 14:32:12 CST 2008


Author: oej
Date: Wed Jan  9 14:32:12 2008
New Revision: 97624

URL: http://svn.digium.com/view/asterisk?view=rev&rev=97624
Log:
Reset, resolve. go. 
Need to check if this branch is needed anymore, don't think this is the proper solution
to the problem.

Modified:
    team/oej/earlyrtpfix/   (props changed)
    team/oej/earlyrtpfix/apps/app_meetme.c
    team/oej/earlyrtpfix/apps/app_queue.c
    team/oej/earlyrtpfix/apps/app_voicemail.c
    team/oej/earlyrtpfix/channels/chan_gtalk.c
    team/oej/earlyrtpfix/channels/chan_mgcp.c
    team/oej/earlyrtpfix/channels/chan_sip.c
    team/oej/earlyrtpfix/channels/chan_zap.c
    team/oej/earlyrtpfix/codecs/codec_zap.c
    team/oej/earlyrtpfix/funcs/func_groupcount.c
    team/oej/earlyrtpfix/main/asterisk.c
    team/oej/earlyrtpfix/main/autoservice.c
    team/oej/earlyrtpfix/main/cli.c
    team/oej/earlyrtpfix/main/editline/readline.c
    team/oej/earlyrtpfix/main/utils.c
    team/oej/earlyrtpfix/res/res_features.c

Propchange: team/oej/earlyrtpfix/
------------------------------------------------------------------------------
    automerge = http://www.codename-pineapple.org/

Propchange: team/oej/earlyrtpfix/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Jan  9 14:32:12 2008
@@ -1,1 +1,1 @@
-/branches/1.4:1-96960
+/branches/1.4:1-97621

Modified: team/oej/earlyrtpfix/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/apps/app_meetme.c?view=diff&rev=97624&r1=97623&r2=97624
==============================================================================
--- team/oej/earlyrtpfix/apps/app_meetme.c (original)
+++ team/oej/earlyrtpfix/apps/app_meetme.c Wed Jan  9 14:32:12 2008
@@ -1578,7 +1578,7 @@
 		goto outrun;
 	}
 
-	retryzap = (strcasecmp(chan->tech->type, "Zap") || chan->spies ? 1 : 0);
+	retryzap = (strcasecmp(chan->tech->type, "Zap") || (chan->spies || chan->monitor) ? 1 : 0);
 	user->zapchannel = !retryzap;
 
  zapretry:
@@ -1896,14 +1896,14 @@
 				break;
 
 			if (c) {
-				if (c->fds[0] != origfd || (user->zapchannel && c->spies)) {
+				if (c->fds[0] != origfd || (user->zapchannel && (c->spies || c->monitor))) {
 					if (using_pseudo) {
 						/* Kill old pseudo */
 						close(fd);
 						using_pseudo = 0;
 					}
 					ast_log(LOG_DEBUG, "Ooh, something swapped out under us, starting over\n");
-					retryzap = (strcasecmp(c->tech->type, "Zap") || c->spies ? 1 : 0);
+					retryzap = (strcasecmp(c->tech->type, "Zap") || (c->spies || c->monitor) ? 1 : 0);
 					user->zapchannel = !retryzap;
 					goto zapretry;
 				}

Modified: team/oej/earlyrtpfix/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/apps/app_queue.c?view=diff&rev=97624&r1=97623&r2=97624
==============================================================================
--- team/oej/earlyrtpfix/apps/app_queue.c (original)
+++ team/oej/earlyrtpfix/apps/app_queue.c Wed Jan  9 14:32:12 2008
@@ -445,7 +445,7 @@
 		warned = 1;
 	}
 }
-
+/*! \brief sets the QUEUESTATUS channel variable */
 static void set_queue_result(struct ast_channel *chan, enum queue_result res)
 {
 	int i;
@@ -508,6 +508,12 @@
 	QUEUE_NORMAL
 };
 
+/*! \brief Check if members are available
+ *
+ * This function checks to see if members are available to be called. If any member
+ * is available, the function immediately returns QUEUE_NORMAL. If no members are available,
+ * the appropriate reason why is returned
+ */
 static enum queue_member_status get_member_status(struct call_queue *q, int max_penalty)
 {
 	struct member *member;
@@ -552,7 +558,7 @@
 	int state;
 	char dev[0];
 };
-
+/*! \brief set a member's status based on device state of that member's interface*/
 static void *handle_statechange(struct statechange *sc)
 {
 	struct call_queue *q;
@@ -656,6 +662,7 @@
 	.thread = AST_PTHREADT_NULL,
 };
 
+/*! \brief Consumer of the statechange queue */
 static void *device_state_thread(void *data)
 {
 	struct statechange *sc = NULL;
@@ -689,7 +696,7 @@
 
 	return NULL;
 }
-
+/*! \brief Producer of the statechange queue */
 static int statechange_queue(const char *dev, int state, void *ign)
 {
 	struct statechange *sc;
@@ -707,7 +714,7 @@
 
 	return 0;
 }
-
+/*! \brief allocate space for new queue member and set fields based on parameters passed */
 static struct member *create_queue_member(const char *interface, const char *membername, int penalty, int paused)
 {
 	struct member *cur;
@@ -716,7 +723,7 @@
 		cur->penalty = penalty;
 		cur->paused = paused;
 		ast_copy_string(cur->interface, interface, sizeof(cur->interface));
-		if(!ast_strlen_zero(membername))
+		if (!ast_strlen_zero(membername))
 			ast_copy_string(cur->membername, membername, sizeof(cur->membername));
 		else
 			ast_copy_string(cur->membername, interface, sizeof(cur->membername));
@@ -788,7 +795,7 @@
 	q->context[0] = '\0';
 	q->monfmt[0] = '\0';
 	q->periodicannouncefrequency = 0;
-	if(!q->members)
+	if (!q->members)
 		q->members = ao2_container_alloc(37, member_hash_fn, member_cmp_fn);
 	q->membercount = 0;
 	q->found = 1;
@@ -1246,14 +1253,14 @@
 	struct ast_variable *var;
 	int ret = -1;
 
-	if(!(var = ast_load_realtime("queue_members", "interface", mem->interface, "queue_name", queue_name, NULL))) 
+	if (!(var = ast_load_realtime("queue_members", "interface", mem->interface, "queue_name", queue_name, NULL))) 
 		return ret;
 	while (var) {
-		if(!strcmp(var->name, "uniqueid"))
+		if (!strcmp(var->name, "uniqueid"))
 			break;
 		var = var->next;
 	}
-	if(var && !ast_strlen_zero(var->value)) {
+	if (var && !ast_strlen_zero(var->value)) {
 		if ((ast_update_realtime("queue_members", "uniqueid", var->value, field, value, NULL)) > -1)
 			ret = 0;
 	}
@@ -1775,6 +1782,11 @@
 	return vars;
 }
 
+/*! \brief Part 2 of ring_one
+ *
+ * Does error checking before attempting to request a channel and call a member. This
+ * function is only called from ring_one
+ */
 static int ring_entry(struct queue_ent *qe, struct callattempt *tmp, int *busies)
 {
 	int res;
@@ -1861,6 +1873,16 @@
 	/* Presense of ADSI CPE on outgoing channel follows ours */
 	tmp->chan->adsicpe = qe->chan->adsicpe;
 
+	/* Inherit context and extension */
+	if (!ast_strlen_zero(qe->chan->macrocontext))
+		ast_copy_string(tmp->chan->dialcontext, qe->chan->macrocontext, sizeof(tmp->chan->dialcontext));
+	else
+		ast_copy_string(tmp->chan->dialcontext, qe->chan->context, sizeof(tmp->chan->dialcontext));
+	if (!ast_strlen_zero(qe->chan->macroexten))
+		ast_copy_string(tmp->chan->exten, qe->chan->macroexten, sizeof(tmp->chan->exten));
+	else
+		ast_copy_string(tmp->chan->exten, qe->chan->exten, sizeof(tmp->chan->exten));
+
 	/* Place the call, but don't wait on the answer */
 	if ((res = ast_call(tmp->chan, location, 0))) {
 		/* Again, keep going even if there's an error */
@@ -1912,6 +1934,14 @@
 	return best;
 }
 
+/*! \brief Place a call to a queue member
+ *
+ * Once metrics have been calculated for each member, this function is used
+ * to place a call to the appropriate member (or members). The low-level
+ * channel-handling and error detection is handled in ring_entry
+ *
+ * Returns 1 if a member was called successfully, 0 otherwise
+ */
 static int ring_one(struct queue_ent *qe, struct callattempt *outgoing, int *busies)
 {
 	int ret = 0;
@@ -2044,7 +2074,16 @@
 }
 
 #define AST_MAX_WATCHERS 256
-
+/*! \brief Wait for a member to answer the call
+ *
+ * \param[in] qe the queue_ent corresponding to the caller in the queue
+ * \param[in] outgoing the list of callattempts. Relevant ones will have their chan and stillgoing parameters non-zero
+ * \param[in] to the amount of time (in milliseconds) to wait for a response
+ * \param[out] digit if a user presses a digit to exit the queue, this is the digit the caller pressed
+ * \param[in] prebusies number of busy members calculated prior to calling wait_for_answer
+ * \param[in] caller_disconnect if the 'H' option is used when calling Queue(), this is used to detect if the caller pressed * to disconnect the call
+ * \param[in] forwardsallowed used to detect if we should allow call forwarding, based on the 'i' option to Queue()
+ */
 static struct callattempt *wait_for_answer(struct queue_ent *qe, struct callattempt *outgoing, int *to, char *digit, int prebusies, int caller_disconnect, int forwardsallowed)
 {
 	char *queue = qe->parent->name;
@@ -2283,7 +2322,15 @@
 
 	return peer;
 }
-
+/*! \brief Check if we should start attempting to call queue members
+ *
+ * The behavior of this function is dependent first on whether autofill is enabled
+ * and second on whether the ring strategy is ringall. If autofill is not enabled,
+ * then return true if we're the head of the queue. If autofill is enabled, then
+ * we count the available members and see if the number of available members is enough
+ * that given our position in the queue, we would theoretically be able to connect to
+ * one of those available members
+ */
 static int is_our_turn(struct queue_ent *qe)
 {
 	struct queue_ent *ch;
@@ -2354,7 +2401,16 @@
 
 	return res;
 }
-
+/*! \brief The waiting areas for callers who are not actively calling members
+ *
+ * This function is one large loop. This function will return if a caller
+ * either exits the queue or it becomes that caller's turn to attempt calling
+ * queue members. Inside the loop, we service the caller with periodic announcements,
+ * holdtime announcements, etc. as configured in queues.conf
+ *
+ * \retval  0 if the caller's turn has arrived
+ * \retval -1 if the caller should exit the queue.
+ */
 static int wait_our_turn(struct queue_ent *qe, int ringing, enum queue_result *reason)
 {
 	int res = 0;
@@ -2424,6 +2480,12 @@
 	return 0;
 }
 
+/*! \brief Calculate the metric of each member in the outgoing callattempts
+ *
+ * A numeric metric is given to each member depending on the ring strategy used
+ * by the queue. Members with lower metrics will be called before members with
+ * higher metrics
+ */
 static int calc_metric(struct call_queue *q, struct member *mem, int pos, struct queue_ent *qe, struct callattempt *tmp)
 {
 	if (qe->max_penalty && (mem->penalty > qe->max_penalty))
@@ -2478,6 +2540,29 @@
 	}
 	return 0;
 }
+/*! \brief A large function which calls members, updates statistics, and bridges the caller and a member
+ * 
+ * Here is the process of this function
+ * 1. Process any options passed to the Queue() application. Options here mean the third argument to Queue()
+ * 2. Iterate trough the members of the queue, creating a callattempt corresponding to each member. During this
+ *    iteration, we also check the dialed_interfaces datastore to see if we have already attempted calling this
+ *    member. If we have, we do not create a callattempt. This is in place to prevent call forwarding loops. Also
+ *    during each iteration, we call calc_metric to determine which members should be rung when.
+ * 3. Call ring_one to place a call to the appropriate member(s)
+ * 4. Call wait_for_answer to wait for an answer. If no one answers, return.
+ * 5. Take care of any holdtime announcements, member delays, or other options which occur after a call has been answered.
+ * 6. Start the monitor or mixmonitor if the option is set
+ * 7. Remove the caller from the queue to allow other callers to advance
+ * 8. Bridge the call.
+ * 9. Do any post processing after the call has disconnected.
+ *
+ * \param[in] qe the queue_ent structure which corresponds to the caller attempting to reach members
+ * \param[in] options the options passed as the third parameter to the Queue() application
+ * \param[in] url the url passed as the fourth parameter to the Queue() application
+ * \param[in,out] tries the number of times we have tried calling queue members
+ * \param[out] noption set if the call to Queue() has the 'n' option set.
+ * \param[in] agi the agi passed as the fifth parameter to the Queue() application
+ */
 
 static int try_calling(struct queue_ent *qe, const char *options, char *announceoverride, const char *url, int *tries, int *noption, const char *agi)
 {
@@ -3073,7 +3158,7 @@
 
 		if ((mem = ao2_find(q->members, &tmpmem, OBJ_POINTER))) {
 			/* XXX future changes should beware of this assumption!! */
-			if(!mem->dynamic) {
+			if (!mem->dynamic) {
 				res = RES_NOT_DYNAMIC;
 				ao2_ref(mem, -1);
 				ast_mutex_unlock(&q->lock);
@@ -3187,7 +3272,7 @@
 				if (queue_persistent_members)
 					dump_queue_members(q);
 
-				if(mem->realtime)
+				if (mem->realtime)
 					update_realtime_member_field(mem, q->name, "paused", paused ? "1" : "0");
 
 				ast_queue_log(q->name, "NONE", mem->membername, (paused ? "PAUSE" : "UNPAUSE"), "%s", "");
@@ -3590,6 +3675,18 @@
 	return 0;
 }
 
+/*!\brief The starting point for all queue calls
+ *
+ * The process involved here is to 
+ * 1. Parse the options specified in the call to Queue()
+ * 2. Join the queue
+ * 3. Wait in a loop until it is our turn to try calling a queue member
+ * 4. Attempt to call a queue member
+ * 5. If 4. did not result in a bridged call, then check for between
+ *    call options such as periodic announcements etc.
+ * 6. Try 4 again uless some condition (such as an expiration time) causes us to 
+ *    exit the queue.
+ */
 static int queue_exec(struct ast_channel *chan, void *data)
 {
 	int res=-1;
@@ -3853,7 +3950,7 @@
 
 	lu = ast_module_user_add(chan);
 
-	if((q = load_realtime_queue(data))) {
+	if ((q = load_realtime_queue(data))) {
 		ast_mutex_lock(&q->lock);
 		mem_iter = ao2_iterator_init(q->members, 0);
 		while ((m = ao2_iterator_next(&mem_iter))) {
@@ -4029,7 +4126,7 @@
 	use_weight=0;
 	/* Mark all non-realtime queues as dead for the moment */
 	AST_LIST_TRAVERSE(&queues, q, list) {
-		if(!q->realtime) {
+		if (!q->realtime) {
 			q->dead = 1;
 			q->found = 0;
 		}
@@ -4070,7 +4167,7 @@
 				/* Check if a queue with this name already exists */
 				if (q->found) {
 					ast_log(LOG_WARNING, "Queue '%s' already defined! Skipping!\n", cat);
-					if(!new)
+					if (!new)
 						ast_mutex_unlock(&q->lock);
 					continue;
 				}
@@ -4095,7 +4192,7 @@
 						AST_NONSTANDARD_APP_ARGS(args, parse, ',');
 
 						interface = args.interface;
-						if(!ast_strlen_zero(args.penalty)) {
+						if (!ast_strlen_zero(args.penalty)) {
 							tmp = args.penalty;
 							while (*tmp && *tmp < 33) tmp++;
 							penalty = atoi(tmp);

Modified: team/oej/earlyrtpfix/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/apps/app_voicemail.c?view=diff&rev=97624&r1=97623&r2=97624
==============================================================================
--- team/oej/earlyrtpfix/apps/app_voicemail.c (original)
+++ team/oej/earlyrtpfix/apps/app_voicemail.c Wed Jan  9 14:32:12 2008
@@ -2569,11 +2569,11 @@
 		ast_log(LOG_WARNING, "Unable to copy mail, mailbox %s is full\n", recip->mailbox);
 		return -1;
 	}
-	if (!(sendvms = get_vm_state_by_imapuser(vmu->imapuser, 2))) {
+	if (!(sendvms = get_vm_state_by_imapuser(vmu->imapuser, 0))) {
 		ast_log(LOG_ERROR, "Couldn't get vm_state for originator's mailbox!!\n");
 		return -1;
 	}
-	if (!(destvms = get_vm_state_by_imapuser(recip->imapuser, 2))) {
+	if (!(destvms = get_vm_state_by_imapuser(recip->imapuser, 0))) {
 		ast_log(LOG_ERROR, "Couldn't get vm_state for destination mailbox!\n");
 		return -1;
 	}
@@ -4721,7 +4721,7 @@
 		stream = mail_open (stream, tmp, debug ? OP_DEBUG : NIL);
 		if (stream == NIL) {
 			ast_log (LOG_ERROR, "Can't connect to imap server %s\n", tmp);
-			return NIL;
+			return -1;
 		}
 		get_mailbox_delimiter(stream);
 		/* update delimiter in imapfolder */
@@ -8805,6 +8805,7 @@
 {
 	struct vmstate *vlist = NULL;
 
+	ast_mutex_lock(&vmstate_lock);
 	vlist = vmstates;
 	while (vlist) {
 		if (vlist->vms) {
@@ -8826,6 +8827,7 @@
 		}
 		vlist = vlist->next;
 	}
+	ast_mutex_unlock(&vmstate_lock);
 	if (option_debug > 2)
 		ast_log(LOG_DEBUG, "%s not found in vmstates\n",user);
 	return NULL;
@@ -8834,7 +8836,8 @@
 static struct vm_state *get_vm_state_by_mailbox(const char *mailbox, int interactive)
 { 
 	struct vmstate *vlist = NULL;
-	
+
+	ast_mutex_lock(&vmstate_lock);
 	vlist = vmstates;
 	if (option_debug > 2) 
 		ast_log(LOG_DEBUG, "Mailbox set to %s\n",mailbox);
@@ -8858,6 +8861,7 @@
 		}
 		vlist = vlist->next;
 	}
+	ast_mutex_unlock(&vmstate_lock);
 	if (option_debug > 2)
 		ast_log(LOG_DEBUG, "%s not found in vmstates\n",mailbox);
 	return NULL;

Modified: team/oej/earlyrtpfix/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_gtalk.c?view=diff&rev=97624&r1=97623&r2=97624
==============================================================================
--- team/oej/earlyrtpfix/channels/chan_gtalk.c (original)
+++ team/oej/earlyrtpfix/channels/chan_gtalk.c Wed Jan  9 14:32:12 2008
@@ -919,6 +919,9 @@
 		return NULL;
 	}
 
+	/* Set CALLERID(name) to the full JID of the remote peer */
+	ast_copy_string(tmp->cid_name, tmp->them, sizeof(tmp->cid_name));
+
 	if(strchr(tmp->us, '/')) {
 		data = ast_strdupa(tmp->us);
 		exten = strsep(&data, "/");
@@ -940,7 +943,6 @@
 	int fmt;
 	int what;
 	const char *n2;
-	char *data = NULL, *cid = NULL;
 
 	if (title)
 		n2 = title;
@@ -999,20 +1001,7 @@
 	ast_module_ref(ast_module_info->self);
 	ast_copy_string(tmp->context, client->context, sizeof(tmp->context));
 	ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
-	/* Don't use ast_set_callerid() here because it will
-	 * generate a needless NewCallerID event */
-	if (!strcasecmp(client->name, "guest")) {
-		data = ast_strdupa(i->them);
-		if (strchr(data, '/')) {
-			cid = strsep(&data, "/");
-		} else
-			cid = data;
-	} else {
-		data =  ast_strdupa(client->user);
-		cid = data;
-	}
-	cid = strsep(&cid, "@");
-	tmp->cid.cid_ani = ast_strdup(cid);
+
 	if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
 		tmp->cid.cid_dnid = ast_strdup(i->exten);
 	tmp->priority = 1;

Modified: team/oej/earlyrtpfix/channels/chan_mgcp.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_mgcp.c?view=diff&rev=97624&r1=97623&r2=97624
==============================================================================
--- team/oej/earlyrtpfix/channels/chan_mgcp.c (original)
+++ team/oej/earlyrtpfix/channels/chan_mgcp.c Wed Jan  9 14:32:12 2008
@@ -434,6 +434,7 @@
 static int mgcp_senddigit_begin(struct ast_channel *ast, char digit);
 static int mgcp_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
 static int mgcp_devicestate(void *data);
+static void add_header_offhook(struct mgcp_subchannel *sub, struct mgcp_request *resp);
 
 static const struct ast_channel_tech mgcp_tech = {
 	.type = "MGCP",
@@ -1276,23 +1277,50 @@
 
 static int mgcp_senddigit_begin(struct ast_channel *ast, char digit)
 {
-	/* Let asterisk play inband indications */
-	return -1;
+	struct mgcp_subchannel *sub = ast->tech_pvt;
+	struct mgcp_endpoint *p = sub->parent;
+	int res = 0;
+
+	ast_mutex_lock(&sub->lock);
+	if (p->dtmfmode & MGCP_DTMF_INBAND || p->dtmfmode & MGCP_DTMF_HYBRID) {
+		ast_log(LOG_DEBUG, "Sending DTMF using inband/hybrid\n");
+		res = -1; /* Let asterisk play inband indications */
+	} else if (p->dtmfmode & MGCP_DTMF_RFC2833) {
+		ast_log(LOG_DEBUG, "Sending DTMF using RFC2833");
+		ast_rtp_senddigit_begin(sub->rtp, digit);
+	} else {
+		ast_log(LOG_ERROR, "Don't know about DTMF_MODE %d\n", p->dtmfmode);
+	}
+	ast_mutex_unlock(&sub->lock);
+
+	return res;
 }
 
 static int mgcp_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration)
 {
 	struct mgcp_subchannel *sub = ast->tech_pvt;
+	struct mgcp_endpoint *p = sub->parent;
+	int res = 0;
 	char tmp[4];
 
-	tmp[0] = 'D';
-	tmp[1] = '/';
-	tmp[2] = digit;
-	tmp[3] = '\0';
 	ast_mutex_lock(&sub->lock);
-	transmit_notify_request(sub, tmp);
+	if (p->dtmfmode & MGCP_DTMF_INBAND || p->dtmfmode & MGCP_DTMF_HYBRID) {
+		ast_log(LOG_DEBUG, "Stopping DTMF using inband/hybrid\n");
+		res = -1; /* Tell Asterisk to stop inband indications */
+	} else if (p->dtmfmode & MGCP_DTMF_RFC2833) {
+		ast_log(LOG_DEBUG, "Stopping DTMF using RFC2833\n");
+		tmp[0] = 'D';
+		tmp[1] = '/';
+		tmp[2] = digit;
+		tmp[3] = '\0';
+		transmit_notify_request(sub, tmp);
+                ast_rtp_senddigit_end(sub->rtp, digit);
+	} else {
+		ast_log(LOG_ERROR, "Don't know about DTMF_MODE %d\n", p->dtmfmode);
+	}
 	ast_mutex_unlock(&sub->lock);
-	return -1; /* Return non-zero so that Asterisk will stop the inband indications */
+
+	return res;
 }
 
 /*!
@@ -2193,7 +2221,7 @@
 		add_header(&resp, "R", "L/hd(N)");
 		break;
 	case MGCP_OFFHOOK:
-		add_header(&resp, "R", (sub->rtp && (p->dtmfmode & MGCP_DTMF_INBAND)) ? "L/hu(N),L/hf(N)" : "L/hu(N),L/hf(N),D/[0-9#*](N)");
+		add_header_offhook(sub, &resp);
 		break;
 	}
 	if (!ast_strlen_zero(tone)) {
@@ -2236,7 +2264,7 @@
 		add_header(&resp, "R", "L/hd(N)");
 		break;
 	case MGCP_OFFHOOK:
-		add_header(&resp, "R",  (sub->rtp && (p->dtmfmode & MGCP_DTMF_INBAND)) ? "L/hu(N),L/hf(N)" : "L/hu(N),L/hf(N),D/[0-9#*](N)");
+		add_header_offhook(sub, &resp);
 		break;
 	}
 	if (!ast_strlen_zero(tone2)) {
@@ -2277,7 +2305,7 @@
 		add_header(&resp, "R", "L/hd(N)");
 		break;
 	case MGCP_OFFHOOK:
-		add_header(&resp, "R",  (sub->rtp && (p->dtmfmode & MGCP_DTMF_INBAND)) ? "L/hu(N), L/hf(N)" : "L/hu(N),L/hf(N),D/[0-9#*](N)");
+		add_header_offhook(sub, &resp);
 		break;
 	}
 	/* fill in new fields */
@@ -2286,6 +2314,16 @@
 	return send_request(p, sub, &resp, oseq); /* SC */
 }
 
+
+static void add_header_offhook(struct mgcp_subchannel *sub, struct mgcp_request *resp)
+{
+	struct mgcp_endpoint *p = sub->parent;
+
+	if (p && p->sub && p->sub->owner && p->sub->owner->_state >= AST_STATE_RINGING && (p->dtmfmode & (MGCP_DTMF_INBAND | MGCP_DTMF_HYBRID)))
+		add_header(resp, "R", "L/hu(N),L/hf(N)");
+	else
+		add_header(resp, "R", "L/hu(N),L/hf(N),D/[0-9#*](N)");
+}
 
 static int transmit_audit_endpoint(struct mgcp_endpoint *p)
 {

Modified: team/oej/earlyrtpfix/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_sip.c?view=diff&rev=97624&r1=97623&r2=97624
==============================================================================
--- team/oej/earlyrtpfix/channels/chan_sip.c (original)
+++ team/oej/earlyrtpfix/channels/chan_sip.c Wed Jan  9 14:32:12 2008
@@ -150,6 +150,7 @@
 #include "asterisk/threadstorage.h"
 #include "asterisk/translate.h"
 #include "asterisk/dnsmgr.h"
+#include "asterisk/astobj2.h"
 
 #ifndef FALSE
 #define FALSE    0
@@ -1011,7 +1012,7 @@
 	struct ast_rtp *remotertp;		/*!< Remote RTP Session (for direct RTP setup) */
 	struct ast_rtp *remotevrtp;		/*!< Remote Video RTP session (for direct RTP setup) */
 
-	struct sip_pkt *packets;		/*!< Packets scheduled for re-transmission */
+	struct ao2_container *packets;		/*!< Packets scheduled for re-transmission */
 	struct sip_history_head *history;	/*!< History of this SIP dialog */
 	size_t history_entries;			/*!< Number of entires in the history */
 	struct ast_variable *chanvars;		/*!< Channel variables to set for inbound call */
@@ -1028,7 +1029,6 @@
 
 /*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission */
 struct sip_pkt {
-	struct sip_pkt *next;			/*!< Next packet in linked list */
 	int retrans;				/*!< Retransmission number */
 	int method;				/*!< SIP method for this packet */
 	int seqno;				/*!< Sequence number */
@@ -1856,7 +1856,7 @@
 	if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
 		return;
 	if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
-		free(hist);
+		ast_free(hist);
 		return;
 	}
 	memcpy(hist->event, buf, l);
@@ -1893,12 +1893,15 @@
 /*! \brief Retransmit SIP message if no answer (Called from scheduler) */
 static int retrans_pkt(const void *data)
 {
-	struct sip_pkt *pkt = (struct sip_pkt *)data, *prev, *cur = NULL;
+	struct sip_pkt *pkt = (struct sip_pkt *)data, *prev;
 	int reschedule = DEFAULT_RETRANS;
 	int xmitres = 0;
 
+	ao2_ref(pkt, 1); /* Make sure this cannot go away while we're using it */
+
 	/* Lock channel PVT */
-	ast_mutex_lock(&pkt->owner->lock);
+	if (pkt->owner)
+		ast_mutex_lock(&pkt->owner->lock);
 
 	if (pkt->retrans < MAX_RETRANS) {
 		pkt->retrans++;
@@ -1926,7 +1929,7 @@
  				ast_log(LOG_DEBUG, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid);
  		} 
 
-		if (sip_debug_test_pvt(pkt->owner)) {
+		if (pkt->owner && sip_debug_test_pvt(pkt->owner)) {
 			const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
 			ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
 				pkt->retrans, sip_nat_mode(pkt->owner),
@@ -1936,38 +1939,43 @@
 
 		append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data);
 		xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
-		ast_mutex_unlock(&pkt->owner->lock);
+		if (pkt->owner)
+			ast_mutex_unlock(&pkt->owner->lock);
 		if (xmitres == XMIT_ERROR)
-			ast_log(LOG_WARNING, "Network error on retransmit in dialog %s\n", pkt->owner->callid);
-		else
+			ast_log(LOG_WARNING, "Network error on retransmit in dialog %s\n", pkt->owner ? pkt->owner->callid : "<unknown>");
+		else {
+			ao2_ref(pkt, -1);
 			return  reschedule;
+		}
 	} 
 	/* Too many retries */
 	if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) {
 		if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug)	/* Tell us if it's critical or if we're debugging */
 			ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request");
-	} else if ((pkt->method == SIP_OPTIONS) && sipdebug) {
-			ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
-	}
-	if (xmitres == XMIT_ERROR) {
-		ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission of transaction in call id %s \n", pkt->owner->callid);
-		append_history(pkt->owner, "XmitErr", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
-	} else
-		append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
- 		
+	} else if (pkt->owner && (pkt->method == SIP_OPTIONS) && sipdebug) {
+		ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
+	}
+	if (pkt->owner) {
+		if (xmitres == XMIT_ERROR) {
+			ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission of transaction in call id %s \n", pkt->owner->callid);
+			append_history(pkt->owner, "XmitErr", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
+		} else
+			append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
+ 	}
 	pkt->retransid = -1;
 
 	if (ast_test_flag(pkt, FLAG_FATAL)) {
-		while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
+		while (pkt->owner && pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
 			ast_mutex_unlock(&pkt->owner->lock);	/* SIP_PVT, not channel */
 			usleep(1);
-			ast_mutex_lock(&pkt->owner->lock);
-		}
-
-		if (pkt->owner->owner && !pkt->owner->owner->hangupcause) 
+			if (pkt->owner)
+				ast_mutex_lock(&pkt->owner->lock);
+		}
+
+		if (pkt->owner && pkt->owner->owner && !pkt->owner->owner->hangupcause) 
 			pkt->owner->owner->hangupcause = AST_CAUSE_NO_USER_RESPONSE;
 		
-		if (pkt->owner->owner) {
+		if (pkt->owner && pkt->owner->owner) {
 			sip_alreadygone(pkt->owner);
 			ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
 			ast_queue_hangup(pkt->owner->owner);
@@ -1976,7 +1984,7 @@
 			/* If no channel owner, destroy now */
 
 			/* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
-			if (pkt->method != SIP_OPTIONS) {
+			if (pkt->owner && pkt->method != SIP_OPTIONS) {
 				ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY);	
 				sip_alreadygone(pkt->owner);
 				if (option_debug)
@@ -1985,7 +1993,7 @@
 		}
 	}
 
-	if (pkt->method == SIP_BYE) {
+	if (pkt->owner && pkt->method == SIP_BYE) {
 		/* We're not getting answers on SIP BYE's.  Tear down the call anyway. */
 		if (pkt->owner->owner) 
 			ast_channel_unlock(pkt->owner->owner);
@@ -1994,23 +2002,23 @@
 	}
 
 	/* In any case, go ahead and remove the packet */
-	for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
-		if (cur == pkt)
-			break;
-	}
-	if (cur) {
-		if (prev)
-			prev->next = cur->next;
-		else
-			pkt->owner->packets = cur->next;
+	if (pkt->owner && (prev = ao2_find(pkt->owner->packets, pkt, OBJ_UNLINK | OBJ_POINTER))) {
+		/* Destroy the container's reference (inherited) */
+		ao2_ref(prev, -1);
 		ast_mutex_unlock(&pkt->owner->lock);
-		free(cur);
-		pkt = NULL;
-	} else
+		/* Now destroy our initial reference */
+		ao2_ref(pkt, -1);
+		/* And destroy the sched ref */
+		ao2_ref(pkt, -1);
+		return 0;
+	} else {
 		ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
-	if (pkt)
-		ast_mutex_unlock(&pkt->owner->lock);
-	return 0;
+		if (pkt->owner)
+			ast_mutex_unlock(&pkt->owner->lock);
+		ao2_ref(pkt, -1); /* Initial ref */
+		ao2_ref(pkt, -1); /* Sched ref */
+		return 0;
+	}
 }
 
 /*! \brief Transmit packet with retransmits 
@@ -2022,12 +2030,11 @@
 	int siptimer_a = DEFAULT_RETRANS;
 	int xmitres = 0;
 
-	if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
+	if (!(pkt = ao2_alloc(sizeof(*pkt) + len + 1, ast_free)))
 		return AST_FAILURE;
 	memcpy(pkt->data, data, len);
 	pkt->method = sipmethod;
 	pkt->packetlen = len;
-	pkt->next = p->packets;
 	pkt->owner = p;
 	pkt->seqno = seqno;
 	if (resp)
@@ -2039,12 +2046,9 @@
 	if (pkt->timer_t1)
 		siptimer_a = pkt->timer_t1 * 2;
 
-	/* Schedule retransmission */
-	pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
 	if (option_debug > 3 && sipdebug)
 		ast_log(LOG_DEBUG, "*** SIP TIMER: Initializing retransmit timer on packet: Id  #%d\n", pkt->retransid);
-	pkt->next = p->packets;
-	p->packets = pkt;
+
 	if (sipmethod == SIP_INVITE) {
 		/* Note this is a pending invite */
 		p->pendinginvite = seqno;
@@ -2054,11 +2058,25 @@
 
 	if (xmitres == XMIT_ERROR) {	/* Serious network trouble, no need to try again */
 		append_history(pkt->owner, "XmitErr", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
-		ast_sched_del(sched, pkt->retransid);	/* No more retransmission */
 		pkt->retransid = -1;
+		ao2_ref(pkt, -1);	/* and deallocate */
 		return AST_FAILURE;
-	} else
+	} else {
+		/* Add refcount for scheduler pointer */
+		ao2_ref(pkt, 1);
+		/* Schedule retransmission */
+		pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
+		/* Link into the list of packets */
+		ao2_link(p->packets, pkt);
 		return AST_SUCCESS;
+	}
+}
+
+static int __deref_ao2_owner_cb(void *obj, void *unused, int flags)
+{
+	struct sip_pkt *pkt = obj;
+	pkt->owner = NULL;
+	return 0;
 }
 
 /*! \brief Kill a SIP dialog (called by scheduler) */
@@ -2072,16 +2090,15 @@
 		p->subscribed = NONE;
 		append_history(p, "Subscribestatus", "timeout");
 		if (option_debug > 2)
-			ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>");
+			ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subscription %s\n", p->callid ? p->callid : "<unknown>");
 		return 10000;	/* Reschedule this destruction so that we know that it's gone */
 	}
 
-	/* If there are packets still waiting for delivery, delay the destruction */
-	if (p->packets) {
-		if (option_debug > 2)
-			ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>");
-		append_history(p, "ReliableXmit", "timeout");
-		return 10000;
+	/* If there are packets still waiting for delivery, make sure they can't callback to us anymore. */
+	if (ao2_container_count(p->packets)) {
+		ast_mutex_lock(&p->lock);
+		ao2_callback(p->packets, 0, __deref_ao2_owner_cb, NULL);
+		ast_mutex_unlock(&p->lock);
 	}
 
 	/* If we're destroying a subscription, dereference peer object too */
@@ -2138,7 +2155,8 @@
 /*! \brief Acknowledges receipt of a packet and stops retransmission */
 static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
 {
-	struct sip_pkt *cur, *prev = NULL;
+	struct sip_pkt *cur;
+	struct ao2_iterator ao2i;
 
 	/* Just in case... */
 	char *msg;
@@ -2147,7 +2165,8 @@
 	msg = sip_methods[sipmethod].text;
 
 	ast_mutex_lock(&p->lock);
-	for (cur = p->packets; cur; prev = cur, cur = cur->next) {
+	ao2i = ao2_iterator_init(p->packets, 0);
+	while ((cur = ao2_iterator_next(&ao2i))) {
 		if ((cur->seqno == seqno) && ((ast_test_flag(cur, FLAG_RESPONSE)) == resp) &&
 			((ast_test_flag(cur, FLAG_RESPONSE)) || 
 			 (!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
@@ -2158,59 +2177,68 @@
 			}
 			/* this is our baby */
 			res = TRUE;
-			UNLINK(cur, p->packets, prev);
 			if (cur->retransid > -1) {
 				if (sipdebug && option_debug > 3)
 					ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
-				ast_sched_del(sched, cur->retransid);
+				if (!ast_sched_del(sched, cur->retransid))
+					ao2_ref(cur, -1); /* scheduler deref */
 				cur->retransid = -1;
 			}
-			free(cur);
+
+			/* Remove it from the list */
+			ao2_unlink(p->packets, cur);
+			ao2_ref(cur, -1); /* iterator deref */
 			break;
 		}
+
+		ao2_ref(cur, -1); /* iterator deref */
 	}
 	ast_mutex_unlock(&p->lock);
 	if (option_debug)
 		ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
 }
 
-/*! \brief Pretend to ack all packets
- * maybe the lock on p is not strictly necessary but there might be a race */
+static int __sip_pretend_ack_cb(void *obj, void *vp, int flags)
+{
+	struct sip_pvt *p = vp;
+	struct sip_pkt *pkt = obj;
+	__sip_ack(p, pkt->seqno, ast_test_flag(pkt, FLAG_RESPONSE), pkt->method ? pkt->method : find_sip_method(pkt->data));
+	return 0;
+}
+
+/*! \brief Pretend to ack all packets */
 static void __sip_pretend_ack(struct sip_pvt *p)
 {
-	struct sip_pkt *cur = NULL;
-
-	while (p->packets) {
-		int method;
-		if (cur == p->packets) {
-			ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
-			return;
-		}
-		cur = p->packets;
-		method = (cur->method) ? cur->method : find_sip_method(cur->data);
-		__sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method);
-	}
+	ao2_callback(p->packets, 0, __sip_pretend_ack_cb, p);
 }
 
 /*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
 static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
 {
-	struct sip_pkt *cur;
+	struct sip_pkt *cur, *found;
 	int res = -1;
-
-	for (cur = p->packets; cur; cur = cur->next) {
+	struct ao2_iterator ao2i;
+
+	ao2i = ao2_iterator_init(p->packets, 0);
+	while ((cur = ao2_iterator_next(&ao2i))) {
 		if (cur->seqno == seqno && ast_test_flag(cur, FLAG_RESPONSE) == resp &&
 			(ast_test_flag(cur, FLAG_RESPONSE) || method_match(sipmethod, cur->data))) {
 			/* this is our baby */
 			if (cur->retransid > -1) {
 				if (option_debug > 3 && sipdebug)
 					ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
-				ast_sched_del(sched, cur->retransid);
+				if (!ast_sched_del(sched, cur->retransid))
+					ao2_ref(cur, -1); /* scheduler deref */
 				cur->retransid = -1;
 			}
 			res = 0;
+			/* Now remove it from the packet list. */
+			if ((found = ao2_find(p->packets, cur, OBJ_UNLINK | OBJ_POINTER)))
+				ao2_ref(found, -1); /* container item deref */
+			ao2_ref(cur, -1); /* iterator deref */
 			break;
 		}
+		ao2_ref(cur, -1); /* iterator deref */
 	}
 	if (option_debug)
 		ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
@@ -3049,11 +3077,21 @@
 	
 }
 
+static int __sip_destroy_packet_cb(void *obj, void *unused, int flags)
+{
+	struct sip_pkt *pkt = obj;
+	if (pkt->retransid > -1) {
+		if (!ast_sched_del(sched, pkt->retransid))
+			ao2_ref(pkt, -1); /* scheduler deref */
+	}
+	pkt->owner = NULL;
+	return 0;
+}
+
 /*! \brief Execute destruction of SIP dialog structure, release memory */
 static void __sip_destroy(struct sip_pvt *p, int lockowner)
 {
 	struct sip_pvt *cur, *prev = NULL;
-	struct sip_pkt *cp;
 
 	if (sip_debug_test_pvt(p) || option_debug > 2)
 		ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
@@ -3072,7 +3110,7 @@
 		sip_dump_history(p);
 
 	if (p->options)
-		free(p->options);
+		ast_free(p->options);
 
 	if (p->stateid > -1)
 		ast_extension_state_del(p->stateid, NULL);
@@ -3090,7 +3128,7 @@
 	if (p->udptl)
 		ast_udptl_destroy(p->udptl);
 	if (p->refer)
-		free(p->refer);
+		ast_free(p->refer);
 	if (p->route) {
 		free_old_route(p->route);
 		p->route = NULL;
@@ -3115,7 +3153,7 @@
 	if (p->history) {
 		struct sip_history *hist;
 		while ( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) ) {
-			free(hist);
+			ast_free(hist);
 			p->history_entries--;
 		}
 		free(p->history);
@@ -3134,12 +3172,8 @@
 	} 
 
 	/* remove all current packets in this dialog */
-	while((cp = p->packets)) {
-		p->packets = p->packets->next;
-		if (cp->retransid > -1)
-			ast_sched_del(sched, cp->retransid);
-		free(cp);
-	}
+	ao2_callback(p->packets, 0, __sip_destroy_packet_cb, NULL);
+
 	if (p->chanvars) {
 		ast_variables_destroy(p->chanvars);
 		p->chanvars = NULL;
@@ -3148,7 +3182,7 @@
 
 	ast_string_field_free_memory(p);
 
-	free(p);
+	ast_free(p);
 }
 
 /*! \brief  update_call_counter: Handle call_limit for SIP users 
@@ -4399,17 +4433,40 @@
 	snprintf(tagbuf, len, "as%08lx", ast_random());
 }
 
+static int packet_hash_fn(const void *obj, const int flags)
+{
+	const struct sip_pkt *pkt = obj;
+	return pkt->seqno;
+}
+
+static int packet_cmp_fn(void *obj1, void *obj2, int flags)
+{
+	struct sip_pkt *p1 = obj1, *p2 = obj2;
+
+	if (flags & OBJ_POINTER)
+		return p1 == p2 ? CMP_MATCH : 0;
+	else
+		return p1->seqno == p2->seqno ? CMP_MATCH : 0;
+}
+
 /*! \brief Allocate SIP_PVT structure and set defaults */
 static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
 				 int useglobal_nat, const int intended_method)
 {
 	struct sip_pvt *p;
-
-	if (!(p = ast_calloc(1, sizeof(*p))))
+	struct ao2_container *aoc;
+
+	if (!(aoc = ao2_container_alloc(37, packet_hash_fn, packet_cmp_fn)))
 		return NULL;
 
+	if (!(p = ast_calloc(1, sizeof(*p)))) {
+		ao2_ref(aoc, -1);
+		return NULL;
+	}
+
 	if (ast_string_field_init(p, 512)) {
-		free(p);
+		ao2_ref(aoc, -1);
+		ast_free(p);
 		return NULL;
 	}
 
@@ -4422,6 +4479,7 @@
 	p->subscribed = NONE;
 	p->stateid = -1;
 	p->prefs = default_prefs;		/* Set default codecs for this call */
+	p->packets = aoc;
 
 	if (intended_method != SIP_OPTIONS)	/* Peerpoke has it's own system */
 		p->timer_t1 = 500;	/* Default SIP retransmission timer T1 (RFC 3261) */
@@ -15588,7 +15646,7 @@
 				}
 			}
 			/* If we have sessions that needs to be destroyed, do it now */
-			if (ast_test_flag(&sip->flags[0], SIP_NEEDDESTROY) && !sip->packets &&
+			if (ast_test_flag(&sip->flags[0], SIP_NEEDDESTROY) && !ao2_container_count(sip->packets) &&
 			    !sip->owner) {
 				ast_mutex_unlock(&sip->lock);
 				__sip_destroy(sip, 1);

Modified: team/oej/earlyrtpfix/channels/chan_zap.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_zap.c?view=diff&rev=97624&r1=97623&r2=97624
==============================================================================

[... 379 lines stripped ...]



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