Multiple revisions 420089-420090,420097

........
  r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
  
  ARI: Add channel technology agnostic out of call text messaging
  
  This patch adds the ability to send and receive text messages from various
  technology stacks in Asterisk through ARI. This includes chan_sip (sip),
  res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the
  endpoints resource, and can be sent directly through that resource, or to a
  particular endpoint.
  
  For example, the following would send the message "Hello there" to PJSIP
  endpoint alice with a display URI of sip:asterisk@mycooldomain.org:
  
  ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There
  
  This is equivalent to the following as well:
  
  ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There
  
  Both forms are available for message technologies that allow for arbitrary
  destinations, such as chan_sip.
  
  Inbound messages can now be received over ARI as well. An ARI application that
  subscribes to endpoints will receive messages from those endpoints:
  
  {
    "type": "TextMessageReceived",
    "timestamp": "2014-07-12T22:53:13.494-0500",
    "endpoint": {
      "technology": "PJSIP",
      "resource": "alice",
      "state": "online",
      "channel_ids": []
    },
    "message": {
      "from": "\"alice\" <sip:alice@127.0.0.1>",
      "to": "pjsip:asterisk@127.0.0.1",
      "body": "Watson, come here.",
      "variables": []
    },
    "application": "testsuite"
  }
  
  The above was made possible due to some rather major changes in the message
  core. This includes (but is not limited to):
  - Users of the message API can now register message handlers. A handler has
    two callbacks: one to determine if the handler has a destination for the
    message, and another to handle it.
  - All dialplan functionality of handling a message was moved into a message
    handler provided by the message API.
  - Messages can now have the technology/endpoint associated with them.
    Various other properties are also now more easily accessible.
  - A number of ao2 containers that weren't really needed were replaced with
    vectors. Iteration over ao2_containers is expensive and pointless when
    the lifetime of things is well defined and the number of things is very
    small.
  
  res_stasis now has a new file that makes up its structure, messaging. The
  messaging functionality implements a message handler, and passes received
  messages that match an interested endpoint over to the app for processing.
  
  Note that inadvertently while testing this, I reproduced ASTERISK-23969.
  res_pjsip_messaging was incorrectly parsing out the 'to' field, such that
  arbitrary SIP URIs mangled the endpoint lookup. This patch includes the
  fix for that as well.
  
  Review: https://reviewboard.asterisk.org/r/3726
  
  ASTERISK-23692 #close
  Reported by: Matt Jordan
  
  ASTERISK-23969 #close
  Reported by: Andrew Nagy
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  r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
  
  Remove automerge properties :-(
........
  r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines
  
  test_message: Fix strict-aliasing compilation issue
........

Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 29 insertions, 8 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 0eb2dc1df..884c2840a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -18871,6 +18871,7 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req, struct a
 	char *to;
 	char from_name[50];
 	char stripped[SIPBUFSIZE];
+	enum sip_get_dest_result dest_result;
 
 	if (strncmp(content_type, "text/plain", strlen("text/plain"))) { /* No text/plain attachment */
 		transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
@@ -18980,7 +18981,8 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req, struct a
 		ast_string_field_set(p, context, sip_cfg.messagecontext);
 	}
 
-	switch (get_destination(p, NULL, NULL)) {
+	dest_result = get_destination(p, NULL, NULL);
+	switch (dest_result) {
 	case SIP_GET_DEST_REFUSED:
 		/* Okay to send 403 since this is after auth processing */
 		transmit_response(p, "403 Forbidden", req);
@@ -18990,12 +18992,9 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req, struct a
 		transmit_response(p, "416 Unsupported URI Scheme", req);
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		return;
-	case SIP_GET_DEST_EXTEN_NOT_FOUND:
-	case SIP_GET_DEST_EXTEN_MATCHMORE:
-		transmit_response(p, "404 Not Found", req);
-		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-		return;
-	case SIP_GET_DEST_EXTEN_FOUND:
+	default:
+		/* We may have something other than dialplan who wants
+		 * the message, so defer further error handling for now */
 		break;
 	}
 
@@ -19023,7 +19022,9 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req, struct a
 	res |= ast_msg_set_context(msg, "%s", p->context);
 
 	res |= ast_msg_set_var(msg, "SIP_RECVADDR", ast_sockaddr_stringify(&p->recv));
+	res |= ast_msg_set_tech(msg, "%s", "SIP");
 	if (!ast_strlen_zero(p->peername)) {
+		res |= ast_msg_set_endpoint(msg, "%s", p->peername);
 		res |= ast_msg_set_var(msg, "SIP_PEERNAME", p->peername);
 	}
 
@@ -19036,12 +19037,32 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req, struct a
 	if (res) {
 		ast_msg_destroy(msg);
 		transmit_response(p, "500 Internal Server Error", req);
-	} else {
+		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+		return;
+	}
+
+	if (ast_msg_has_destination(msg)) {
 		ast_msg_queue(msg);
 		transmit_response(p, "202 Accepted", req);
+		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+		return;
 	}
 
+	/* Find a specific error cause to send */
+	switch (dest_result) {
+	case SIP_GET_DEST_EXTEN_NOT_FOUND:
+	case SIP_GET_DEST_EXTEN_MATCHMORE:
+		transmit_response(p, "404 Not Found", req);
+		break;
+	case SIP_GET_DEST_EXTEN_FOUND:
+	default:
+		/* We should have sent the message already! */
+		ast_assert(0);
+		transmit_response(p, "500 Internal Server Error", req);
+		break;
+	}
 	sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+	ast_msg_destroy(msg);
 }
 
 /*! \brief  CLI Command to show calls within limits set by call_limit */