[asterisk-commits] file: branch file/pimp_sip_location r382336 - in /team/file/pimp_sip_location...

[SVN commits to the Asterisk project]

Author: file
Date: Fri Mar  1 06:21:34 2013
New Revision: 382336

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=382336
Log:
Fix race conditions between answering and terminating.

Modified:
    team/file/pimp_sip_location/channels/chan_gulp.c
    team/file/pimp_sip_location/res/res_sip_sdp_audio.c

Modified: team/file/pimp_sip_location/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_location/channels/chan_gulp.c?view=diff&rev=382336&r1=382335&r2=382336
==============================================================================
--- team/file/pimp_sip_location/channels/chan_gulp.c (original)
+++ team/file/pimp_sip_location/channels/chan_gulp.c Fri Mar  1 06:21:34 2013
@@ -979,6 +979,10 @@
 {
 	struct pjsip_status_line status = rdata->msg_info.msg->line.status;
 
+	if (!session->channel) {
+		return;
+	}
+
 	switch (status.code) {
 	case 180:
 		ast_queue_control(session->channel, AST_CONTROL_RINGING);

Modified: team/file/pimp_sip_location/res/res_sip_sdp_audio.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_location/res/res_sip_sdp_audio.c?view=diff&rev=382336&r1=382335&r2=382336
==============================================================================
--- team/file/pimp_sip_location/res/res_sip_sdp_audio.c (original)
+++ team/file/pimp_sip_location/res/res_sip_sdp_audio.c Fri Mar  1 06:21:34 2013
@@ -266,6 +266,10 @@
 	RAII_VAR(struct ast_format_cap *, jointcap, NULL, ast_format_cap_destroy);
 	RAII_VAR(struct ast_format_cap *, peercap, NULL, ast_format_cap_destroy);
 	struct ast_format fmt;
+
+	if (!session->channel) {
+		return 1;
+	}
 
 	/* Create an RTP instance if need be */
 	if (!session->media[AST_SIP_MEDIA_AUDIO].rtp && audio_create_rtp(session, session->endpoint->rtp_ipv6)) {