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authorJoshua Colp <jcolp@digium.com>2006-09-01 17:54:22 +0000
committerJoshua Colp <jcolp@digium.com>2006-09-01 17:54:22 +0000
commit0be2884d80634e80285ff9ce049fb7ec102f73a1 (patch)
treed3fe4ddb989ebc789fa0dcef20266cb9abde6e03 /channels/chan_sip.c
parent449f311cdac6cb6f9e248a56bd9f1ffd572fb669 (diff)
If we are doing video and we can't reinvite, then resort to generic bridging instead of Packet2Packet since video isn't supported there yet. (reported by PCadach in #asterisk-bugs)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels/chan_sip.c')
-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 f841816df..072b41f43 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -16130,7 +16130,7 @@ static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struc
return AST_RTP_GET_FAILED;
ast_mutex_lock(&p->lock);
- if (!(p->rtp)) {
+ if (!(p->vrtp)) {
ast_mutex_unlock(&p->lock);
return AST_RTP_GET_FAILED;
}