[asterisk-commits] file: branch group/pimp_my_sip r379930 - /team/group/pimp_my_sip/channels/

[SVN commits to the Asterisk project]

Author: file
Date: Tue Jan 22 14:25:05 2013
New Revision: 379930

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=379930
Log:
Allow the calling party to hang up the session.

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=379930&r1=379929&r2=379930
==============================================================================
--- team/group/pimp_my_sip/channels/chan_gulp.c (original)
+++ team/group/pimp_my_sip/channels/chan_gulp.c Tue Jan 22 14:25:05 2013
@@ -95,12 +95,16 @@
 };
 
 /*! \brief SIP session interaction functions */
+static void gulp_session_end(struct ast_sip_session *session);
 static int gulp_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
+static void gulp_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
 
 /*! \brief SIP session supplement structure */
 static struct ast_sip_session_supplement gulp_supplement = {
 	.method = "INVITE",
+	.session_end = gulp_session_end,
 	.incoming_request = gulp_incoming_request,
+	.incoming_response = gulp_incoming_response,
 };
 
 /*! \brief Function called by RTP engine to get local RTP peer */
@@ -497,6 +501,105 @@
 	return 0;
 }
 
+/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
+static int hangup_sip2cause(int cause)
+{
+	/* Possible values taken from causes.h */
+
+	switch(cause) {
+	case 401:       /* Unauthorized */
+		return AST_CAUSE_CALL_REJECTED;
+	case 403:       /* Not found */
+		return AST_CAUSE_CALL_REJECTED;
+	case 404:       /* Not found */
+		return AST_CAUSE_UNALLOCATED;
+	case 405:       /* Method not allowed */
+		return AST_CAUSE_INTERWORKING;
+	case 407:       /* Proxy authentication required */
+		return AST_CAUSE_CALL_REJECTED;
+	case 408:       /* No reaction */
+		return AST_CAUSE_NO_USER_RESPONSE;
+	case 409:       /* Conflict */
+		return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
+	case 410:       /* Gone */
+		return AST_CAUSE_NUMBER_CHANGED;
+	case 411:       /* Length required */
+		return AST_CAUSE_INTERWORKING;
+	case 413:       /* Request entity too large */
+		return AST_CAUSE_INTERWORKING;
+	case 414:       /* Request URI too large */
+		return AST_CAUSE_INTERWORKING;
+	case 415:       /* Unsupported media type */
+		return AST_CAUSE_INTERWORKING;
+	case 420:       /* Bad extension */
+		return AST_CAUSE_NO_ROUTE_DESTINATION;
+	case 480:       /* No answer */
+		return AST_CAUSE_NO_ANSWER;
+	case 481:       /* No answer */
+		return AST_CAUSE_INTERWORKING;
+	case 482:       /* Loop detected */
+		return AST_CAUSE_INTERWORKING;
+	case 483:       /* Too many hops */
+		return AST_CAUSE_NO_ANSWER;
+	case 484:       /* Address incomplete */
+		return AST_CAUSE_INVALID_NUMBER_FORMAT;
+	case 485:       /* Ambiguous */
+		return AST_CAUSE_UNALLOCATED;
+	case 486:       /* Busy everywhere */
+		return AST_CAUSE_BUSY;
+	case 487:       /* Request terminated */
+		return AST_CAUSE_INTERWORKING;
+	case 488:       /* No codecs approved */
+		return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
+	case 491:       /* Request pending */
+		return AST_CAUSE_INTERWORKING;
+	case 493:       /* Undecipherable */
+		return AST_CAUSE_INTERWORKING;
+	case 500:       /* Server internal failure */
+		return AST_CAUSE_FAILURE;
+	case 501:       /* Call rejected */
+		return AST_CAUSE_FACILITY_REJECTED;
+	case 502:
+		return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
+	case 503:       /* Service unavailable */
+		return AST_CAUSE_CONGESTION;
+	case 504:       /* Gateway timeout */
+		return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
+	case 505:       /* SIP version not supported */
+		return AST_CAUSE_INTERWORKING;
+	case 600:       /* Busy everywhere */
+		return AST_CAUSE_USER_BUSY;
+	case 603:       /* Decline */
+		return AST_CAUSE_CALL_REJECTED;
+	case 604:       /* Does not exist anywhere */
+		return AST_CAUSE_UNALLOCATED;
+	case 606:       /* Not acceptable */
+		return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
+	default:
+		if (cause < 500 && cause >= 400) {
+			/* 4xx class error that is unknown - someting wrong with our request */
+			return AST_CAUSE_INTERWORKING;
+		} else if (cause < 600 && cause >= 500) {
+			/* 5xx class error - problem in the remote end */
+			return AST_CAUSE_CONGESTION;
+		} else if (cause < 700 && cause >= 600) {
+			/* 6xx - global errors in the 4xx class */
+			return AST_CAUSE_INTERWORKING;
+		}
+		return AST_CAUSE_NORMAL;
+	}
+	/* Never reached */
+	return 0;
+}
+
+/*! \brief Function called when the session ends */
+static void gulp_session_end(struct ast_sip_session *session)
+{
+	int cause = hangup_sip2cause(session->inv_session->cause);
+
+	ast_queue_hangup_with_cause(session->channel, cause);
+}
+
 /*! \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)
 {
@@ -544,6 +647,11 @@
 	}
 
 	return (res == AST_PBX_SUCCESS) ? 0 : -1;
+}
+
+/*! \brief Function called when a response is received on the session */
+static void gulp_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
 }
 
 /*!