mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-26 14:27:14 +00:00
Merge patches for 1.8.11-cert6
This includes the following * r369351 for AST-883 * r368807 for AST-884 * r356604, r356650, r364203 for AST-890 * r370618 for AST-896 * r370205, r370273, r370360 for AST-916 * r371469 for AST-932 git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/1.8.11@371651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
@@ -9034,6 +9034,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
|
||||
int video = FALSE;
|
||||
int image = FALSE;
|
||||
int text = FALSE;
|
||||
int processed_crypto = FALSE;
|
||||
char protocol[5] = {0,};
|
||||
int x;
|
||||
|
||||
@@ -9208,28 +9209,34 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
|
||||
case 'a':
|
||||
/* Audio specific scanning */
|
||||
if (audio) {
|
||||
if (process_sdp_a_sendonly(value, &sendonly))
|
||||
if (process_sdp_a_sendonly(value, &sendonly)) {
|
||||
processed = TRUE;
|
||||
else if (process_crypto(p, p->rtp, &p->srtp, value))
|
||||
} else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
|
||||
processed_crypto = TRUE;
|
||||
processed = TRUE;
|
||||
else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
|
||||
} else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
|
||||
processed = TRUE;
|
||||
}
|
||||
}
|
||||
/* Video specific scanning */
|
||||
else if (video) {
|
||||
if (process_sdp_a_sendonly(value, &vsendonly))
|
||||
if (process_sdp_a_sendonly(value, &vsendonly)) {
|
||||
processed = TRUE;
|
||||
else if (process_crypto(p, p->vrtp, &p->vsrtp, value))
|
||||
} else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value)) {
|
||||
processed_crypto = TRUE;
|
||||
processed = TRUE;
|
||||
else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
|
||||
} else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
|
||||
processed = TRUE;
|
||||
}
|
||||
}
|
||||
/* Text (T.140) specific scanning */
|
||||
else if (text) {
|
||||
if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
|
||||
if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
|
||||
processed = TRUE;
|
||||
else if (process_crypto(p, p->trtp, &p->tsrtp, value))
|
||||
} else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value)) {
|
||||
processed_crypto = TRUE;
|
||||
processed = TRUE;
|
||||
}
|
||||
}
|
||||
/* Image (T.38 FAX) specific scanning */
|
||||
else if (image) {
|
||||
@@ -16251,11 +16258,12 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
|
||||
}
|
||||
if (!ast_strlen_zero(peer->cid_name))
|
||||
ast_string_field_set(p, cid_name, peer->cid_name);
|
||||
if (!ast_strlen_zero(peer->cid_tag))
|
||||
ast_string_field_set(p, cid_tag, peer->cid_tag);
|
||||
if (peer->callingpres)
|
||||
p->callingpres = peer->callingpres;
|
||||
}
|
||||
if (!ast_strlen_zero(peer->cid_tag)) {
|
||||
ast_string_field_set(p, cid_tag, peer->cid_tag);
|
||||
}
|
||||
ast_string_field_set(p, fullcontact, peer->fullcontact);
|
||||
|
||||
if (!ast_strlen_zero(peer->context)) {
|
||||
@@ -27318,6 +27326,10 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
|
||||
} else if (!strcasecmp(word, "nonat")) {
|
||||
ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
|
||||
ast_clear_flag(&flags[0], SIP_DIRECT_MEDIA_NAT);
|
||||
} else if (!strcasecmp(word, "outgoing")) {
|
||||
ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
|
||||
ast_set_flag(&mask[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
|
||||
ast_set_flag(&flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
|
||||
} else {
|
||||
ast_log(LOG_WARNING, "Unknown directmedia mode '%s' on line %d\n", v->value, v->lineno);
|
||||
}
|
||||
@@ -29747,6 +29759,18 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
|
||||
p->redircodecs = codecs;
|
||||
changed = 1;
|
||||
}
|
||||
|
||||
if (ast_test_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING) && !p->outgoing_call) {
|
||||
/* We only wish to withhold sending the initial direct media reinvite on the incoming dialog.
|
||||
* Further direct media reinvites beyond the initial should be sent. In order to allow further
|
||||
* direct media reinvites to be sent, we clear this flag.
|
||||
*/
|
||||
ast_clear_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
|
||||
sip_pvt_unlock(p);
|
||||
ast_channel_unlock(chan);
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
|
||||
if (chan->_state != AST_STATE_UP) { /* We are in early state */
|
||||
if (p->do_history)
|
||||
@@ -30115,12 +30139,6 @@ static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struc
|
||||
}
|
||||
}
|
||||
|
||||
/* For now, when we receive an INVITE just take the first successful crypto line */
|
||||
if ((*srtp)->crypto && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
|
||||
ast_debug(3, "We've already processed a crypto attribute, skipping '%s'\n", a);
|
||||
return FALSE;
|
||||
}
|
||||
|
||||
if (!(*srtp)->crypto && !((*srtp)->crypto = sdp_crypto_setup())) {
|
||||
return FALSE;
|
||||
}
|
||||
|
||||
@@ -359,9 +359,10 @@
|
||||
|
||||
|
||||
#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
|
||||
#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 1) /*!< DP: Only send direct media reinvites on outgoing calls */
|
||||
|
||||
#define SIP_PAGE3_FLAGS_TO_COPY \
|
||||
(SIP_PAGE3_SNOM_AOC)
|
||||
(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_DIRECT_MEDIA_OUTGOING)
|
||||
|
||||
/*@}*/
|
||||
|
||||
|
||||
@@ -889,6 +889,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
||||
; instead of INVITE. This can be combined with 'nonat', as
|
||||
; 'directmedia=update,nonat'. It implies 'yes'.
|
||||
|
||||
;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
|
||||
; reinvite on an incoming call leg. This option is useful when
|
||||
; peered with another SIP user agent that is known to send
|
||||
; immediate direct media reinvites upon call establishment. Setting
|
||||
; the option in this situation helps to prevent potential glares.
|
||||
; Setting this option implies 'yes'.
|
||||
|
||||
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
|
||||
; the call directly with media peer-2-peer without re-invites.
|
||||
; Will not work for video and cases where the callee sends
|
||||
|
||||
46
main/cel.c
46
main/cel.c
@@ -398,6 +398,14 @@ void ast_cel_check_retire_linkedid(struct ast_channel *chan)
|
||||
}
|
||||
}
|
||||
|
||||
/* Note that no 'chan_fixup' function is provided for this datastore type,
|
||||
* because the channels that will use it will never be involved in masquerades.
|
||||
*/
|
||||
static const struct ast_datastore_info fabricated_channel_datastore = {
|
||||
.type = "CEL fabricated channel",
|
||||
.destroy = ast_free_ptr,
|
||||
};
|
||||
|
||||
struct ast_channel *ast_cel_fabricate_channel_from_event(const struct ast_event *event)
|
||||
{
|
||||
struct varshead *headp;
|
||||
@@ -407,6 +415,8 @@ struct ast_channel *ast_cel_fabricate_channel_from_event(const struct ast_event
|
||||
struct ast_cel_event_record record = {
|
||||
.version = AST_CEL_EVENT_RECORD_VERSION,
|
||||
};
|
||||
struct ast_datastore *datastore;
|
||||
char *app_data;
|
||||
|
||||
/* do not call ast_channel_alloc because this is not really a real channel */
|
||||
if (!(tchan = ast_dummy_channel_alloc())) {
|
||||
@@ -469,10 +479,42 @@ struct ast_channel *ast_cel_fabricate_channel_from_event(const struct ast_event
|
||||
AST_LIST_INSERT_HEAD(headp, newvariable, entries);
|
||||
}
|
||||
|
||||
tchan->appl = ast_strdup(record.application_name);
|
||||
tchan->data = ast_strdup(record.application_data);
|
||||
tchan->amaflags = record.amaflag;
|
||||
|
||||
/* We need to store an 'application name' and 'application
|
||||
* data' on the channel for logging purposes, but the channel
|
||||
* structure only provides a place to store pointers, and it
|
||||
* expects these pointers to be pointing to data that does not
|
||||
* need to be freed. This means that the channel's destructor
|
||||
* does not attempt to free any storage that these pointers
|
||||
* point to. However, we can't provide data in that form directly for
|
||||
* these structure members. In order to ensure that these data
|
||||
* elements have a lifetime that matches the channel's
|
||||
* lifetime, we'll put them in a datastore attached to the
|
||||
* channel, and set's the channel's pointers to point into the
|
||||
* datastore. The datastore will then be automatically destroyed
|
||||
* when the channel is destroyed.
|
||||
*/
|
||||
|
||||
if (!(datastore = ast_datastore_alloc(&fabricated_channel_datastore, NULL))) {
|
||||
ast_channel_unref(tchan);
|
||||
return NULL;
|
||||
}
|
||||
|
||||
if (!(app_data = ast_malloc(strlen(record.application_name) + strlen(record.application_data) + 2))) {
|
||||
ast_datastore_free(datastore);
|
||||
ast_channel_unref(tchan);
|
||||
return NULL;
|
||||
}
|
||||
|
||||
tchan->appl = app_data;
|
||||
tchan->data = app_data + strlen(record.application_name) + 1;
|
||||
|
||||
strcpy((char *) tchan->appl, record.application_name);
|
||||
strcpy((char *) tchan->data, record.application_data);
|
||||
datastore->data = app_data;
|
||||
ast_channel_datastore_add(tchan, datastore);
|
||||
|
||||
return tchan;
|
||||
}
|
||||
|
||||
|
||||
@@ -2484,6 +2484,13 @@ static void ast_dummy_channel_destructor(void *obj)
|
||||
struct ast_channel *chan = obj;
|
||||
struct ast_var_t *vardata;
|
||||
struct varshead *headp;
|
||||
struct ast_datastore *datastore;
|
||||
|
||||
/* Get rid of each of the data stores on the channel */
|
||||
while ((datastore = AST_LIST_REMOVE_HEAD(&chan->datastores, entry))) {
|
||||
/* Free the data store */
|
||||
ast_datastore_free(datastore);
|
||||
}
|
||||
|
||||
headp = &chan->varshead;
|
||||
|
||||
|
||||
@@ -1784,14 +1784,16 @@ static struct ast_str *xmldoc_get_formatted(struct ast_xml_node *node, int raw_o
|
||||
{
|
||||
struct ast_xml_node *tmp;
|
||||
const char *notcleanret, *tmpstr;
|
||||
struct ast_str *ret = ast_str_create(128);
|
||||
struct ast_str *ret;
|
||||
|
||||
if (raw_output) {
|
||||
/* xmldoc_string_cleanup will allocate the ret object */
|
||||
notcleanret = ast_xml_get_text(node);
|
||||
tmpstr = notcleanret;
|
||||
xmldoc_string_cleanup(ast_skip_blanks(notcleanret), &ret, 0);
|
||||
ast_xml_free_text(tmpstr);
|
||||
} else {
|
||||
ret = ast_str_create(128);
|
||||
for (tmp = ast_xml_node_get_children(node); tmp; tmp = ast_xml_node_get_next(tmp)) {
|
||||
/* if found, parse a <para> element. */
|
||||
if (xmldoc_parse_para(tmp, "", "\n", &ret)) {
|
||||
|
||||
Reference in New Issue
Block a user