Fix formatting issue with previous patch.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266580 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index bcc3469a6e47a1e5d178872055b71fed7c997f0e..d678f7bd43c9d3405051de51b4b924a86978dd8c 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -19012,13 +19012,12 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt
                ast_mutex_unlock(&p->lock);
                return 0;
        } else {
-                       ast_log(LOG_ERROR, "Something went wrong with T.38. State is:%d on channel %s and %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>", p->t38.state, chan ? chan->name : "<none>");
-                       ast_mutex_unlock(&p->lock);
-                       return 0;
+               ast_log(LOG_ERROR, "Something went wrong with T.38. State is:%d on channel %s and %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>", p->t38.state, chan ? chan->name : "<none>");
+               ast_mutex_unlock(&p->lock);
+               return 0;
        }
 }
 
-
 /*! \brief Returns null if we can't reinvite audio (part of RTP interface) */
 static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
 {