[asterisk-commits] oej: branch 1.4 r172169 - /branches/1.4/channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Jan 29 02:48:18 CST 2009


Author: oej
Date: Thu Jan 29 02:48:18 2009
New Revision: 172169

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=172169
Log:
Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause.
This patch implements a temporary storage in the pvt and use that instead.

The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!

(closes issue #14294)
related to issue #13385

Reported by: klaus3000 and adomjan
Patches: 
      bug14294b.diff uploaded by oej (license 306)
      Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000


Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=172169&r1=172168&r2=172169
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Thu Jan 29 02:48:18 2009
@@ -1028,6 +1028,7 @@
 	struct sip_pvt *next;			/*!< Next dialog in chain */
 	struct sip_invite_param *options;	/*!< Options for INVITE */
 	int autoframing;
+	int hangupcause;			/*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
 } *iflist = NULL;
 
 /*! Max entires in the history list for a sip_pvt */
@@ -3539,6 +3540,10 @@
 		return 0;
 	}
 
+	/* Store hangupcause locally in PVT so we still have it before disconnect */
+	if (p->owner)
+		p->hangupcause = p->owner->hangupcause;
+
 	if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
 		if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
 			if (option_debug && sipdebug)
@@ -3589,7 +3594,7 @@
 
 	stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
 
-	append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->owner->hangupcause) : "Unknown");
+	append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->hangupcause) : "Unknown");
 
 	/* Disconnect */
 	if (p->vad)
@@ -3641,7 +3646,7 @@
 				}
 			} else {	/* Incoming call, not up */
 				const char *res;
-				if (ast->hangupcause && (res = hangup_cause2sip(ast->hangupcause)))
+				if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
 					transmit_response_reliable(p, res, &p->initreq);
 				else 
 					transmit_response_reliable(p, "603 Declined", &p->initreq);
@@ -7997,11 +8002,11 @@
 	}
 	/* If we are hanging up and know a cause for that, send it in clear text to make
 		debugging easier. */
-	if (sipmethod == SIP_BYE && p->owner && p->owner->hangupcause)	{
+	if (sipmethod == SIP_BYE)	{
 		char buf[10];
 
-		add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
-		snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
+		add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->hangupcause));
+		snprintf(buf, sizeof(buf), "%d", p->hangupcause);
 		add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
 	}
 




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