From 4499cf87b7bac11dd4dbfb9b527ab5e6c6f4d8a7 Mon Sep 17 00:00:00 2001 From: pabelanger Date: Tue, 1 Jun 2010 14:57:49 +0000 Subject: Fix formatting issue with previous patch. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266580 f38db490-d61c-443f-a65b-d21fe96a405b --- channels/chan_sip.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'channels') diff --git a/channels/chan_sip.c b/channels/chan_sip.c index bcc3469a6..d678f7bd4 100644 --- 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 : "", p->t38.state, chan ? chan->name : ""); - 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 : "", p->t38.state, chan ? chan->name : ""); + 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) { -- cgit v1.2.3