[asterisk-commits] russell: trunk r321546 - in /trunk: ./ channels/ channels/sip/include/ config...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jun 1 16:31:47 CDT 2011
Author: russell
Date: Wed Jun 1 16:31:40 2011
New Revision: 321546
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=321546
Log:
Support routing text messages outside of a call.
Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Added:
trunk/include/asterisk/message.h
- copied unchanged from r321545, team/russell/messaging/include/asterisk/message.h
trunk/main/message.c
- copied unchanged from r321545, team/russell/messaging/main/message.c
Modified:
trunk/CHANGES
trunk/channels/chan_sip.c
trunk/channels/sip/include/sip.h
trunk/configs/jabber.conf.sample
trunk/configs/sip.conf.sample
trunk/include/asterisk/_private.h
trunk/include/asterisk/channel.h
trunk/include/asterisk/jabber.h
trunk/main/asterisk.c
trunk/main/channel.c
trunk/res/res_jabber.c
Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Wed Jun 1 16:31:40 2011
@@ -11,6 +11,19 @@
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.8 to Asterisk 1.10 -----------------
------------------------------------------------------------------------------
+
+Text Messaging
+--------------
+ * Asterisk now has protocol independent support for processing text messages
+ outside of a call. Messages are routed through the Asterisk dialplan.
+ SIP MESSAGE and XMPP are currently supported. There are options in
+ jabber.conf and sip.conf to allow enabling these features.
+ -> jabber.conf: see the "sendtodialplan" and "context" options.
+ -> sip.conf: see the "accept_outofcall_message" and "auth_message_requests"
+ options.
+ The MESSAGE() dialplan function and MessageSend() application have been
+ added to go along with this functionality. More detailed usage information
+ can be found on the Asterisk wiki (http://wiki.asterisk.org/).
Parking
-------
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Jun 1 16:31:40 2011
@@ -263,6 +263,7 @@
#include "asterisk/cel.h"
#include "asterisk/data.h"
#include "asterisk/aoc.h"
+#include "asterisk/message.h"
#include "sip/include/sip.h"
#include "sip/include/globals.h"
#include "sip/include/config_parser.h"
@@ -1252,7 +1253,8 @@
static int transmit_info_with_aoc(struct sip_pvt *p, struct ast_aoc_decoded *decoded);
static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration);
static int transmit_info_with_vidupdate(struct sip_pvt *p);
-static int transmit_message_with_text(struct sip_pvt *p, const char *text);
+static int transmit_message_with_text(struct sip_pvt *p, const char *text, int init, int auth);
+static int transmit_message_with_msg(struct sip_pvt *p, const struct ast_msg *msg);
static int transmit_refer(struct sip_pvt *p, const char *dest);
static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten);
static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
@@ -1261,7 +1263,7 @@
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
static void copy_request(struct sip_request *dst, const struct sip_request *src);
-static void receive_message(struct sip_pvt *p, struct sip_request *req);
+static void receive_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward);
static int sip_send_mwi_to_peer(struct sip_peer *peer, const struct ast_event *event, int cache_only);
@@ -1532,7 +1534,7 @@
static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *sin, const char *e);
static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
-static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
+static int handle_request_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int seqno, const char *e);
static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
@@ -1547,6 +1549,7 @@
static void handle_response_refer(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
static void handle_response_subscribe(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
static int handle_response_register(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
+static void handle_response_message(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno);
/*------ SRTP Support -------- */
@@ -4444,7 +4447,7 @@
}
if (debug)
ast_verbose("Sending text %s on %s\n", text, ast->name);
- transmit_message_with_text(dialog, text);
+ transmit_message_with_text(dialog, text, 0, 0);
return 0;
}
@@ -13117,14 +13120,47 @@
return res;
}
+/*! \brief Transmit text with SIP MESSAGE method based on an ast_msg */
+static int transmit_message_with_msg(struct sip_pvt *p, const struct ast_msg *msg)
+{
+ struct sip_request req;
+ struct ast_msg_var_iterator *i;
+ const char *var, *val;
+
+ initreqprep(&req, p, SIP_MESSAGE, NULL);
+ ast_string_field_set(p, msg_body, ast_msg_get_body(msg));
+ initialize_initreq(p, &req);
+
+ i = ast_msg_var_iterator_init(msg);
+ while (ast_msg_var_iterator_next(msg, i, &var, &val)) {
+ add_header(&req, var, val);
+ ast_msg_var_unref_current(i);
+ }
+ ast_msg_var_iterator_destroy(i);
+
+ add_text(&req, ast_msg_get_body(msg));
+
+ return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
+}
+
/*! \brief Transmit text with SIP MESSAGE method */
-static int transmit_message_with_text(struct sip_pvt *p, const char *text)
+static int transmit_message_with_text(struct sip_pvt *p, const char *text, int init, int auth)
{
struct sip_request req;
-
- reqprep(&req, p, SIP_MESSAGE, 0, 1);
- add_text(&req, text);
- return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
+
+ if (init) {
+ initreqprep(&req, p, SIP_MESSAGE, NULL);
+ ast_string_field_set(p, msg_body, text);
+ initialize_initreq(p, &req);
+ } else {
+ reqprep(&req, p, SIP_MESSAGE, 0, 1);
+ }
+ if (auth) {
+ return transmit_request_with_auth(p, SIP_MESSAGE, p->ocseq, XMIT_RELIABLE, 0);
+ } else {
+ add_text(&req, text);
+ return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
+ }
}
/*! \brief Allocate SIP refer structure */
@@ -13355,6 +13391,10 @@
add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->hangupcause));
snprintf(buf, sizeof(buf), "%d", p->hangupcause);
add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
+ }
+
+ if (sipmethod == SIP_MESSAGE) {
+ add_text(&resp, p->msg_body);
}
return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
@@ -15912,15 +15952,52 @@
return 0;
}
+static int get_msg_text2(struct ast_str **buf, struct sip_request *req, int addnewline)
+{
+ int i, res = 0;
+
+ ast_str_reset(*buf);
+
+ for (i = 0; res >= 0 && i < req->lines; i++) {
+ const char *line = REQ_OFFSET_TO_STR(req, line[i]);
+
+ res = ast_str_append(buf, 0, "%s%s", line, addnewline ? "\n" : "");
+ }
+
+ return res < 0 ? -1 : 0;
+}
+
+static void set_message_vars_from_req(struct ast_msg *msg, struct sip_request *req)
+{
+ size_t x;
+ char name_buf[1024] = "";
+ char val_buf[1024] = "";
+ char *c;
+
+ for (x = 0; x < req->headers; x++) {
+ const char *header = REQ_OFFSET_TO_STR(req, header[x]);
+ if ((c = strchr(header, ':'))) {
+ ast_copy_string(name_buf, header, MIN((c - header + 1), sizeof(name_buf)));
+ ast_copy_string(val_buf, ast_skip_blanks(c + 1), sizeof(val_buf));
+ ast_trim_blanks(name_buf);
+ ast_msg_set_var(msg, name_buf, val_buf);
+ }
+ }
+}
+
+AST_THREADSTORAGE(sip_msg_buf);
/*! \brief Receive SIP MESSAGE method messages
\note We only handle messages within current calls currently
Reference: RFC 3428 */
-static void receive_message(struct sip_pvt *p, struct sip_request *req)
-{
- char buf[1400];
+static void receive_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
+{
+ struct ast_str *buf;
struct ast_frame f;
const char *content_type = get_header(req, "Content-Type");
+ struct ast_msg *msg;
+ int res;
+ char *from, *to;
if (strncmp(content_type, "text/plain", strlen("text/plain"))) { /* No text/plain attachment */
transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
@@ -15929,7 +16006,15 @@
return;
}
- if (get_msg_text(buf, sizeof(buf), req, FALSE)) {
+ if (!(buf = ast_str_thread_get(&sip_msg_buf, 128))) {
+ transmit_response(p, "500 Internal Server Error", req);
+ if (!p->owner) {
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ }
+ return;
+ }
+
+ if (get_msg_text2(&buf, req, FALSE)) {
ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid);
transmit_response(p, "202 Accepted", req);
if (!p->owner)
@@ -15939,23 +16024,93 @@
if (p->owner) {
if (sip_debug_test_pvt(p))
- ast_verbose("SIP Text message received: '%s'\n", buf);
+ ast_verbose("SIP Text message received: '%s'\n", ast_str_buffer(buf));
memset(&f, 0, sizeof(f));
f.frametype = AST_FRAME_TEXT;
f.subclass.integer = 0;
f.offset = 0;
- f.data.ptr = buf;
- f.datalen = strlen(buf) + 1;
+ f.data.ptr = ast_str_buffer(buf);
+ f.datalen = ast_str_strlen(buf) + 1;
ast_queue_frame(p->owner, &f);
transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */
return;
}
- /* Message outside of a call, we do not support that */
- ast_log(LOG_WARNING, "Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req, "To"), get_header(req, "From"), content_type, buf);
- transmit_response(p, "405 Method Not Allowed", req);
+ if (!sip_cfg.accept_outofcall_message) {
+ /* Message outside of a call, we do not support that */
+ ast_debug(1, "MESSAGE outside of a call administratively disabled.\n");
+ transmit_response(p, "405 Method Not Allowed", req);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ return;
+ }
+
+ if (sip_cfg.auth_message_requests) {
+ int res;
+
+ copy_request(&p->initreq, req);
+ set_pvt_allowed_methods(p, req);
+ res = check_user(p, req, SIP_MESSAGE, e, XMIT_UNRELIABLE, addr);
+ if (res == AUTH_CHALLENGE_SENT) {
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ return;
+ }
+ if (res < 0) { /* Something failed in authentication */
+ if (res == AUTH_FAKE_AUTH) {
+ ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
+ transmit_fake_auth_response(p, SIP_OPTIONS, req, XMIT_UNRELIABLE);
+ } else {
+ ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
+ transmit_response(p, "403 Forbidden", req);
+ }
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ return;
+ }
+ /* Auth was successful. Proceed. */
+ } else {
+ struct sip_peer *peer;
+
+ /*
+ * MESSAGE outside of a call, not authenticating it.
+ * Check to see if we match a peer anyway so that we can direct
+ * it to the right context.
+ */
+
+ peer = find_peer(NULL, &p->recv, TRUE, FINDPEERS, 0, p->socket.type);
+ if (peer) {
+ /* Only if no auth is required. */
+ if (ast_strlen_zero(peer->secret) && ast_strlen_zero(peer->md5secret)) {
+ ast_string_field_set(p, context, peer->context);
+ }
+ peer = unref_peer(peer, "from find_peer() in receive_message");
+ }
+ }
+
+ if (!(msg = ast_msg_alloc())) {
+ transmit_response(p, "500 Internal Server Error", req);
+ if (!p->owner) {
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ }
+ return;
+ }
+
+ to = ast_strdupa(REQ_OFFSET_TO_STR(req, rlPart2));
+ from = ast_strdupa(get_header(req, "From"));
+
+ res = ast_msg_set_to(msg, "%s", to);
+ res |= ast_msg_set_from(msg, "%s", get_in_brackets(from));
+ res |= ast_msg_set_body(msg, "%s", ast_str_buffer(buf));
+ res |= ast_msg_set_context(msg, "%s", p->context);
+ res |= ast_msg_set_exten(msg, "%s", p->exten);
+
+ if (res) {
+ ast_msg_destroy(msg);
+ } else {
+ set_message_vars_from_req(msg, req);
+ ast_msg_queue(msg);
+ }
+
+ transmit_response(p, "202 Accepted", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return;
}
/*! \brief CLI Command to show calls within limits set by call_limit */
@@ -20549,6 +20704,8 @@
handle_response_register(p, resp, rest, req, seqno);
else if (sipmethod == SIP_UPDATE) {
handle_response_update(p, resp, rest, req, seqno);
+ } else if (sipmethod == SIP_MESSAGE) {
+ handle_response_message(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) {
if (p->options)
p->options->auth_type = resp;
@@ -20894,11 +21051,11 @@
#ifdef WHEN_WE_KNOW_THAT_THE_CLIENT_SUPPORTS_MESSAGE
if (!res) {
- transmit_message_with_text(transferer->tech_pvt, "Unable to park call.\n");
+ transmit_message_with_text(transferer->tech_pvt, "Unable to park call.\n", 0, 0);
} else {
/* Then tell the transferer what happened */
sprintf(buf, "Call parked on extension '%d'", ext);
- transmit_message_with_text(transferer->tech_pvt, buf);
+ transmit_message_with_text(transferer->tech_pvt, buf, 0, 0);
}
#endif
@@ -23378,16 +23535,127 @@
return 1;
}
+/*!
+ * \internal
+ * \brief Handle auth requests to a MESSAGE request
+ */
+static void handle_response_message(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
+{
+ char *header, *respheader;
+ char digest[1024];
+
+ if (p->options) {
+ p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
+ }
+
+ if ((p->authtries == MAX_AUTHTRIES)) {
+ ast_log(LOG_NOTICE, "Failed to authenticate on MESSAGE to '%s'\n", get_header(&p->initreq, "From"));
+ pvt_set_needdestroy(p, "MESSAGE authentication failed");
+ return;
+ }
+
+ p->authtries++;
+ auth_headers((resp == 401 ? WWW_AUTH : PROXY_AUTH), &header, &respheader);
+ memset(digest, 0, sizeof(digest));
+ if (reply_digest(p, req, header, SIP_MESSAGE, digest, sizeof(digest))) {
+ /* There's nothing to use for authentication */
+ ast_debug(1, "Nothing to use for MESSAGE authentication\n");
+ pvt_set_needdestroy(p, "MESSAGE authentication failed");
+ return;
+ }
+
+ if (p->do_history) {
+ append_history(p, "MessageAuth", "Try: %d", p->authtries);
+ }
+
+ transmit_message_with_text(p, p->msg_body, 0, 1);
+}
+
/*! \brief Handle incoming MESSAGE request */
-static int handle_request_message(struct sip_pvt *p, struct sip_request *req)
+static int handle_request_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
{
if (!req->ignore) {
if (req->debug)
ast_verbose("Receiving message!\n");
- receive_message(p, req);
+ receive_message(p, req, addr, e);
} else
transmit_response(p, "202 Accepted", req);
return 1;
+}
+
+static int sip_msg_send(const struct ast_msg *msg, const char *to, const char *from);
+
+static const struct ast_msg_tech sip_msg_tech = {
+ .name = "sip",
+ .msg_send = sip_msg_send,
+};
+
+static int sip_msg_send(const struct ast_msg *msg, const char *to, const char *from)
+{
+ struct sip_pvt *pvt;
+ int res;
+ char *peer;
+ struct sip_peer *peer_ptr;
+
+ if (!(pvt = sip_alloc(NULL, NULL, 0, SIP_MESSAGE, NULL))) {
+ return -1;
+ }
+
+ peer = ast_strdupa(to);
+ if (strchr(peer, '@')) {
+ strsep(&peer, "@");
+ } else {
+ strsep(&peer, ":");
+ }
+ if (ast_strlen_zero(peer)) {
+ ast_log(LOG_WARNING, "MESSAGE(to) is invalid for SIP - '%s'\n", to);
+ return -1;
+ }
+
+ if (!ast_strlen_zero(from)) {
+ if ((peer_ptr = find_peer(from, NULL, 0, 1, 0, 0))) {
+ ast_string_field_set(pvt, fromname, S_OR(peer_ptr->cid_name, peer_ptr->name));
+ ast_string_field_set(pvt, fromuser, S_OR(peer_ptr->cid_num, peer_ptr->name));
+ unref_peer(peer_ptr, "unref_peer, from sip_msg_send, find_peer");
+ } else if (strchr(from, '<')) { /* from is callerid-style */
+ char *sender;
+ char *name = NULL, *location = NULL, *user = NULL, *domain = NULL;
+
+ sender = ast_strdupa(from);
+ ast_callerid_parse(sender, &name, &location);
+ ast_string_field_set(pvt, fromname, name);
+ if (strchr(location, ':')) { /* Must be a URI */
+ parse_uri(location, "sip:,sips:", &user, NULL, &domain, NULL);
+ ast_string_field_set(pvt, fromuser, user);
+ ast_string_field_set(pvt, fromdomain, domain);
+ } else { /* Treat it as an exten/user */
+ ast_string_field_set(pvt, fromuser, location);
+ }
+ } else { /* assume we just have the name, use defaults for the rest */
+ ast_string_field_set(pvt, fromname, from);
+ }
+ }
+
+ sip_pvt_lock(pvt);
+
+ if (create_addr(pvt, peer, NULL, TRUE, NULL)) {
+ sip_pvt_unlock(pvt);
+ dialog_unlink_all(pvt, TRUE, TRUE);
+ dialog_unref(pvt, "create_addr failed sending a MESSAGE");
+ return -1;
+ }
+ ast_sip_ouraddrfor(&pvt->sa, &pvt->ourip, pvt);
+ ast_set_flag(&pvt->flags[0], SIP_OUTGOING);
+
+ /* XXX Does pvt->expiry need to be set? */
+
+ res = transmit_message_with_msg(pvt, msg);
+
+ sip_pvt_unlock(pvt);
+ sip_scheddestroy(pvt, DEFAULT_TRANS_TIMEOUT);
+ dialog_unref(pvt, "sent a MESSAGE");
+
+ return res;
}
static enum sip_publish_type determine_sip_publish_type(struct sip_request *req, const char * const event, const char * const etag, const char * const expires, int *expires_int)
@@ -24589,7 +24857,7 @@
res = handle_request_bye(p, req);
break;
case SIP_MESSAGE:
- res = handle_request_message(p, req);
+ res = handle_request_message(p, req, addr, e);
break;
case SIP_PUBLISH:
res = handle_request_publish(p, req, addr, seqno, e);
@@ -27368,6 +27636,8 @@
sip_cfg.directrtpsetup = FALSE; /* Experimental feature, disabled by default */
sip_cfg.alwaysauthreject = DEFAULT_ALWAYSAUTHREJECT;
sip_cfg.auth_options_requests = DEFAULT_AUTH_OPTIONS;
+ sip_cfg.auth_message_requests = DEFAULT_AUTH_MESSAGE;
+ sip_cfg.accept_outofcall_message = DEFAULT_ACCEPT_OUTOFCALL_MESSAGE;
sip_cfg.allowsubscribe = FALSE;
sip_cfg.disallowed_methods = SIP_UNKNOWN;
sip_cfg.contact_ha = NULL; /* Reset the contact ACL */
@@ -27616,6 +27886,10 @@
if (ast_true(v->value)) {
sip_cfg.auth_options_requests = 1;
}
+ } else if (!strcasecmp(v->name, "auth_message_requests")) {
+ sip_cfg.auth_message_requests = ast_true(v->value) ? 1 : 0;
+ } else if (!strcasecmp(v->name, "accept_outofcall_message")) {
+ sip_cfg.accept_outofcall_message = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "mohinterpret")) {
ast_copy_string(default_mohinterpret, v->value, sizeof(default_mohinterpret));
} else if (!strcasecmp(v->name, "mohsuggest")) {
@@ -29586,6 +29860,11 @@
memcpy(&sip_tech_info, &sip_tech, sizeof(sip_tech));
memset((void *) &sip_tech_info.send_digit_begin, 0, sizeof(sip_tech_info.send_digit_begin));
+ if (ast_msg_tech_register(&sip_msg_tech)) {
+ /* LOAD_FAILURE stops Asterisk, so cleanup is a moot point. */
+ return AST_MODULE_LOAD_FAILURE;
+ }
+
/* Make sure we can register our sip channel type */
if (ast_channel_register(&sip_tech)) {
ast_log(LOG_ERROR, "Unable to register channel type 'SIP'\n");
@@ -29694,6 +29973,8 @@
/* First, take us out of the channel type list */
ast_channel_unregister(&sip_tech);
+ ast_msg_tech_unregister(&sip_msg_tech);
+
/* Unregister dial plan functions */
ast_custom_function_unregister(&sipchaninfo_function);
ast_custom_function_unregister(&sippeer_function);
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Wed Jun 1 16:31:40 2011
@@ -211,6 +211,8 @@
#define DEFAULT_CALLEVENTS FALSE /*!< Extra manager SIP call events */
#define DEFAULT_ALWAYSAUTHREJECT TRUE /*!< Don't reject authentication requests always */
#define DEFAULT_AUTH_OPTIONS FALSE
+#define DEFAULT_AUTH_MESSAGE TRUE
+#define DEFAULT_ACCEPT_OUTOFCALL_MESSAGE TRUE
#define DEFAULT_REGEXTENONQUALIFY FALSE
#define DEFAULT_LEGACY_USEROPTION_PARSING FALSE
#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
@@ -680,6 +682,8 @@
int allowguest; /*!< allow unauthenticated peers to connect? */
int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
int auth_options_requests; /*!< Authenticate OPTIONS requests */
+ int auth_message_requests; /*!< Authenticate MESSAGE requests */
+ int accept_outofcall_message; /*!< Accept MESSAGE outside of a call */
int compactheaders; /*!< send compact sip headers */
int allow_external_domains; /*!< Accept calls to external SIP domains? */
int callevents; /*!< Whether we send manager events or not */
@@ -966,6 +970,7 @@
AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
AST_STRING_FIELD(engine); /*!< RTP engine to use */
AST_STRING_FIELD(dialstring); /*!< The dialstring used to call this SIP endpoint */
+ AST_STRING_FIELD(msg_body); /*!< Text for a MESSAGE body */
);
char via[128]; /*!< Via: header */
int maxforwards; /*!< SIP Loop prevention */
Modified: trunk/configs/jabber.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/jabber.conf.sample?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/configs/jabber.conf.sample (original)
+++ trunk/configs/jabber.conf.sample Wed Jun 1 16:31:40 2011
@@ -34,3 +34,6 @@
; Messages stored longer than this value will be deleted by Asterisk.
; This option applies to incoming messages only, which are intended to
; be processed by the JABBER_RECEIVE dialplan function.
+;sendtodialplan=yes ; Send incoming messages into the dialplan. Off by default.
+;context=messages ; Dialplan context to send incoming messages to. If not set,
+ ; "default" will be used.
Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Wed Jun 1 16:31:40 2011
@@ -384,6 +384,16 @@
;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
; INVITE requests are. By default this option is disabled.
+
+;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
+ ; call. By default, this option is enabled. When enabled, MESSAGE
+ ; requests are passed in to the dialplan.
+
+;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
+ ; By default this option is enabled. However, it can be disabled
+ ; should an application desire to not load the Asterisk server with
+ ; doing authentication and implement end to end security in the
+ ; message body.
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
Modified: trunk/include/asterisk/_private.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/_private.h?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/include/asterisk/_private.h (original)
+++ trunk/include/asterisk/_private.h Wed Jun 1 16:31:40 2011
@@ -47,6 +47,7 @@
int ast_cel_engine_reload(void); /*!< Provided by cel.c */
int ast_ssl_init(void); /*!< Provided by ssl.c */
int ast_test_init(void); /*!< Provided by test.c */
+int ast_msg_init(void); /*!< Provided by message.c */
/*!
* \brief Reload asterisk modules.
Modified: trunk/include/asterisk/channel.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/channel.h?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/include/asterisk/channel.h (original)
+++ trunk/include/asterisk/channel.h Wed Jun 1 16:31:40 2011
@@ -3496,4 +3496,14 @@
}
#endif
+/*!
+ * \brief Remove a channel from the global channels container
+ *
+ * \param chan channel to remove
+ *
+ * In a case where it is desired that a channel not be available in any lookups
+ * in the global channels conatiner, use this function.
+ */
+void ast_channel_unlink(struct ast_channel *chan);
+
#endif /* _ASTERISK_CHANNEL_H */
Modified: trunk/include/asterisk/jabber.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/jabber.h?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/include/asterisk/jabber.h (original)
+++ trunk/include/asterisk/jabber.h Wed Jun 1 16:31:40 2011
@@ -157,6 +157,7 @@
char name_space[256];
char sid[10]; /* Session ID */
char mid[6]; /* Message ID */
+ char context[AST_MAX_CONTEXT];
iksid *jid;
iksparser *p;
iksfilter *f;
@@ -179,6 +180,7 @@
int message_timeout;
int authorized;
int distribute_events;
+ int send_to_dialplan;
struct ast_flags flags;
int component; /* 0 client, 1 component */
struct aji_buddy_container buddies;
Modified: trunk/main/asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/asterisk.c?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/main/asterisk.c (original)
+++ trunk/main/asterisk.c Wed Jun 1 16:31:40 2011
@@ -3750,6 +3750,11 @@
ast_xmldoc_load_documentation();
#endif
+ if (ast_msg_init()) {
+ printf("%s", term_quit());
+ exit(1);
+ }
+
/* initialize the data retrieval API */
if (ast_data_init()) {
printf ("%s", term_quit());
Modified: trunk/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/channel.c?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/main/channel.c (original)
+++ trunk/main/channel.c Wed Jun 1 16:31:40 2011
@@ -9593,3 +9593,8 @@
return result;
}
+
+void ast_channel_unlink(struct ast_channel *chan)
+{
+ ao2_unlink(channels, chan);
+}
Modified: trunk/res/res_jabber.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_jabber.c?view=diff&rev=321546&r1=321545&r2=321546
==============================================================================
--- trunk/res/res_jabber.c (original)
+++ trunk/res/res_jabber.c Wed Jun 1 16:31:40 2011
@@ -60,6 +60,7 @@
#include "asterisk/manager.h"
#include "asterisk/event.h"
#include "asterisk/devicestate.h"
+#include "asterisk/message.h"
/*** DOCUMENTATION
<application name="JabberSend" language="en_US">
@@ -373,6 +374,13 @@
static int aji_register_transport2(void *data, ikspak *pak);
*/
+static int msg_send_cb(const struct ast_msg *msg, const char *to, const char *from);
+
+static const struct ast_msg_tech msg_tech = {
+ .name = "xmpp",
+ .msg_send = msg_send_cb,
+};
+
static struct ast_cli_entry aji_cli[] = {
AST_CLI_DEFINE(aji_do_set_debug, "Enable/Disable Jabber debug"),
AST_CLI_DEFINE(aji_do_reload, "Reload Jabber configuration"),
@@ -1134,6 +1142,44 @@
ast_aji_send_chat(client, args.recipient, args.message);
}
return 0;
+}
+
+static int msg_send_cb(const struct ast_msg *msg, const char *to, const char *from)
+{
+ struct aji_client *client;
+ char *sender;
+ char *dest;
+ int res;
+
+ sender = ast_strdupa(from);
+ strsep(&sender, ":");
+ dest = ast_strdupa(to);
+ strsep(&dest, ":");
+
+ if (ast_strlen_zero(sender)) {
+ ast_log(LOG_ERROR, "MESSAGE(from) of '%s' invalid for xmpp\n", from);
+ return -1;
+ }
+
+ if (!(client = ast_aji_get_client(sender))) {
+ ast_log(LOG_WARNING, "Could not finder account to send from as '%s'\n", sender);
+ return -1;
+ }
+
+
+ ast_debug(1, "Sending message to '%s' from '%s'\n", dest, client->name);
+
+ res = ast_aji_send_chat(client, dest, ast_msg_get_body(msg));
+ if (res != IKS_OK) {
+ ast_log(LOG_WARNING, "Failed to send xmpp message (%d).\n", res);
+ }
+
+ /*
+ * XXX Reference leak here. See note with ast_aji_get_client() about the problems
+ * with that function.
+ */
+
+ return res == IKS_OK ? 0 : -1;
}
/*!
@@ -2218,6 +2264,7 @@
{
struct aji_message *insert;
int deleted = 0;
+ struct ast_msg *msg;
ast_debug(3, "client %s received a message\n", client->name);
@@ -2246,6 +2293,23 @@
return;
}
ast_debug(3, "message comes from %s\n", insert->from);
+ }
+
+ if ((msg = ast_msg_alloc())) {
+ int res;
+
+ res = ast_msg_set_to(msg, "xmpp:%s", client->user);
+ res |= ast_msg_set_from(msg, "xmpp:%s", insert->from);
+ res |= ast_msg_set_body(msg, "%s", insert->message);
+ res |= ast_msg_set_context(msg, "%s", client->context);
+
+ if (res) {
+ ast_msg_destroy(msg);
+ } else {
+ ast_msg_queue(msg);
+ }
+
+ msg = NULL;
}
/* remove old messages received from this JID
@@ -4248,6 +4312,7 @@
ASTOBJ_CONTAINER_MARKALL(&client->buddies);
ast_copy_string(client->name, label, sizeof(client->name));
ast_copy_string(client->mid, "aaaaa", sizeof(client->mid));
+ ast_copy_string(client->context, "default", sizeof(client->context));
/* Set default values for the client object */
client->debug = debug;
@@ -4265,6 +4330,7 @@
ast_copy_string(client->statusmessage, "Online and Available", sizeof(client->statusmessage));
client->priority = 0;
client->status = IKS_SHOW_AVAILABLE;
+ client->send_to_dialplan = 0;
if (flag) {
client->authorized = 0;
@@ -4356,6 +4422,10 @@
} else {
ast_log(LOG_WARNING, "Unknown presence status: %s\n", var->value);
}
+ } else if (!strcasecmp(var->name, "context")) {
+ ast_copy_string(client->context, var->value, sizeof(client->context));
+ } else if (!strcasecmp(var->name, "sendtodialplan")) {
+ client->send_to_dialplan = ast_true(var->value) ? 1 : 0;
}
/* no transport support in this version */
/* else if (!strcasecmp(var->name, "transport"))
@@ -4553,6 +4623,13 @@
* (without the resource string)
* \param name label or JID
* \return aji_client.
+ *
+ * XXX \bug This function leads to reference leaks all over the place.
+ * ASTOBJ_CONTAINER_FIND() returns a reference, but if the
+ * client is found via the traversal, no reference is returned.
+ * None of the calling code releases references. This code needs
+ * to be changed to always return a reference, and all of the users
+ * need to be fixed to release them.
*/
struct aji_client *ast_aji_get_client(const char *name)
{
@@ -4668,7 +4745,7 @@
*/
static int unload_module(void)
{
-
+ ast_msg_tech_unregister(&msg_tech);
ast_cli_unregister_multiple(aji_cli, ARRAY_LEN(aji_cli));
ast_unregister_application(app_ajisend);
ast_unregister_application(app_ajisendgroup);
@@ -4721,6 +4798,7 @@
ast_cli_register_multiple(aji_cli, ARRAY_LEN(aji_cli));
ast_custom_function_register(&jabberstatus_function);
ast_custom_function_register(&jabberreceive_function);
+ ast_msg_tech_register(&msg_tech);
ast_mutex_init(&messagelock);
ast_cond_init(&message_received_condition, NULL);
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