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authormmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b>2008-08-20 15:39:19 +0000
committermmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b>2008-08-20 15:39:19 +0000
commitd6ce86419d173db1cb2c0eeeeb16e4061adca4ad (patch)
treef49e6eb22fc0e57d3f9505a5d08c5a9f75d8a0ff /channels
parentbbbf913808c14fb989e9fb0fe78ec110cf2ff3a0 (diff)
Merged revisions 139016 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug 2008) | 14 lines Merged revisions 139015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines sip_read should properly handle a NULL return from sip_rtp_read. (closes issue #13257) Reported by: travishein ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@139017 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index eaed0bc40..96a08bb4d 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5725,7 +5725,7 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
}
/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
- if (fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
+ if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
fr = &ast_null_frame;
}