[asterisk-commits] file: branch group/pimp_my_sip r379224 - /team/group/pimp_my_sip/channels/

[SVN commits to the Asterisk project]

Author: file
Date: Wed Jan 16 10:59:50 2013
New Revision: 379224

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=379224
Log:
Add a supplement which reacts to new sessions by creating an Asterisk channel and sending it into the dialplan.

Modified:
    team/group/pimp_my_sip/channels/chan_gulp.c

Modified: team/group/pimp_my_sip/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/channels/chan_gulp.c?view=diff&rev=379224&r1=379223&r2=379224
==============================================================================
--- team/group/pimp_my_sip/channels/chan_gulp.c (original)
+++ team/group/pimp_my_sip/channels/chan_gulp.c Wed Jan 16 10:59:50 2013
@@ -100,6 +100,14 @@
 	.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
 };
 
+/*! \brief SIP session interaction functions */
+static int gulp_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
+
+/*! \brief SIP session supplement structure */
+static struct ast_sip_session_supplement gulp_supplement = {
+	.incoming_request = gulp_incoming_request,
+};
+
 /*! \brief Function called by RTP engine to get local RTP peer */
 static enum ast_rtp_glue_result gulp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
 {
@@ -124,6 +132,32 @@
 	.get_codec = gulp_get_codec,
 	.update_peer = gulp_set_rtp_peer,
 };
+
+/*! \brief Function called to create a new Gulp Asterisk channel */
+static struct ast_channel *gulp_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const char *linkedid, const char *cid_name)
+{
+	struct ast_channel *chan;
+
+	/* TODO: Fix channel name generation */
+	if (!(chan = ast_channel_alloc(1, state, "", S_OR(cid_name, ""), "", "", "", linkedid, 0, "Gulp/%04lx", ast_random() & 0xffff))) {
+		return NULL;
+	}
+
+	ast_channel_tech_set(chan, &gulp_tech);
+	ast_channel_tech_pvt_set(chan, session);
+
+	if (state == AST_STATE_RING) {
+		ast_channel_rings_set(chan, 1);
+	}
+
+	ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
+
+	ast_channel_context_set(chan, "default");
+	ast_channel_exten_set(chan, S_OR(exten, "s"));
+	ast_channel_priority_set(chan, 1);
+
+       return chan;
+}
 
 /*! \brief Function called by core when we should answer a Gulp session */
 static int gulp_answer(struct ast_channel *ast)
@@ -411,6 +445,48 @@
 	return 0;
 }
 
+/*! \brief Function called when a request is received on the session */
+static int gulp_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+	int res = AST_PBX_FAILED;
+
+	/* We only care about new sessions */
+	if ((rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) ||
+	    (session->channel)) {
+		return 0;
+	}
+
+	if (!(session->channel = gulp_new(session, AST_STATE_DOWN, NULL, NULL, NULL, NULL))) {
+		pjsip_tx_data *packet = NULL;
+
+		if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
+			pjsip_inv_send_msg(session->inv_session, packet);
+		}
+
+		return -1;
+	}
+
+	res = ast_pbx_start(session->channel);
+
+	switch (res) {
+	case AST_PBX_FAILED:
+		ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
+		ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
+		ast_hangup(session->channel);
+		break;
+	case AST_PBX_CALL_LIMIT:
+		ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+		ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
+		ast_hangup(session->channel);
+		break;
+	case AST_PBX_SUCCESS:
+	default:
+		break;
+	}
+
+	return (res == AST_PBX_SUCCESS) ? 0 : -1;
+}
+
 /*!
  * \brief Load the module
  *
@@ -436,6 +512,11 @@
 		goto end;
 	}
 
+	if (ast_sip_session_register_supplement(&gulp_supplement)) {
+		ast_log(LOG_ERROR, "Unable to register Gulp supplement\n");
+		goto end;
+	}
+
 	return 0;
 
 end:
@@ -453,6 +534,7 @@
 /*! \brief Unload the Gulp channel from Asterisk */
 static int unload_module(void)
 {
+	ast_sip_session_unregister_supplement(&gulp_supplement);
 	ast_channel_unregister(&gulp_tech);
 	ast_rtp_glue_unregister(&gulp_rtp_glue);