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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-09-01 17:54:22 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-09-01 17:54:22 +0000
commit99355c0ba46364c365629eaba70c0ef86227222a (patch)
treed3fe4ddb989ebc789fa0dcef20266cb9abde6e03 /channels
parentfff8a4e8f2ec72df731b6419c6a8b5e5a960bd2b (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: http://svn.digium.com/svn/asterisk/trunk@41718 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 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;
}