[asterisk-commits] pabelanger: branch 1.4 r266580 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: pabelanger
Date: Tue Jun  1 09:57:49 2010
New Revision: 266580

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=266580
Log:
Fix formatting issue with previous patch.

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=266580&r1=266579&r2=266580
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Jun  1 09:57:49 2010
@@ -19012,12 +19012,11 @@
 		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)