From: Joshua Colp <jcolp@digium.com>
Date: Fri, 6 Nov 2009 16:56:37 +0000 (+0000)
Subject: Fix a crash caused by freeing a dialog directly instead of using dialog_unref.
X-Git-Tag: 1.6.0.18-rc1~9
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=8d5fab9e52ef89e0dc318af2ceb3bb4bb8a5ffa4;p=thirdparty%2Fasterisk.git

Fix a crash caused by freeing a dialog directly instead of using dialog_unref.

(closes issue #16097)
Reported by: steinwej
Patches:
      no_RTP.diff uploaded by steinwej (license 841)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 70e2b7d0c7..9ac215491c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6218,7 +6218,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
 		return NULL;
 
 	if (ast_string_field_init(p, 512)) {
-		ast_free(p);
+		dialog_unref(p);
 		return NULL;
 	}
 
@@ -6278,27 +6278,12 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
 			p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
 			p->t38_maxdatagram = global_t38_maxdatagram;
 		}
- 		if (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp) 
+ 		if (p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp) 
 				|| (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && !p->trtp)) {
  			ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n",
  				ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "",
  				ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "and text " : "", strerror(errno));
-			if (p->rtp) {
-				ast_rtp_destroy(p->rtp);
-			}
-			if (p->vrtp) {
-				ast_rtp_destroy(p->vrtp);
-			}
-			if (p->udptl) {
-				ast_udptl_destroy(p->udptl);
-			}
-			ast_mutex_destroy(&p->pvt_lock);
-			if (p->chanvars) {
-				ast_variables_destroy(p->chanvars);
-				p->chanvars = NULL;
-			}
-			ast_string_field_free_memory(p);
-			ast_free(p);
+			dialog_unref(p);
 			return NULL;
 		}
 		ast_rtp_setqos(p->rtp, global_tos_audio, global_cos_audio, "SIP RTP");