Merged revisions 172173 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

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r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 lines

Merged revisions 172169 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines

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


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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172217 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 11 insertions, 6 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1354c37bf..99667d317 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1309,7 +1309,8 @@ struct sip_pvt {
 							(A bit unsure of this, please correct if
 							you know more) */
 	struct sip_st_dlg *stimer;		/*!< SIP Session-Timers */              
-};
+	int hangupcause;			/*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
+}; 
 
 
 /*! Max entires in the history list for a sip_pvt */
@@ -4844,6 +4845,10 @@ static int sip_hangup(struct ast_channel *ast)
 		p->answered_elsewhere = TRUE;
 	}
 
+	/* 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 (sipdebug)
@@ -4891,7 +4896,7 @@ static int sip_hangup(struct ast_channel *ast)
 
 	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)
@@ -4942,7 +4947,7 @@ static int sip_hangup(struct ast_channel *ast)
 				}
 			} 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);
@@ -9786,11 +9791,11 @@ static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqn
 	}
 	/* 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);
 	}