From: Paul Belanger Date: Tue, 1 Jun 2010 14:57:49 +0000 (+0000) Subject: Fix formatting issue with previous patch. X-Git-Tag: 1.4.33-rc1~4 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=b3076dd0f55aaaf88a29710265f986c3f515fc79;p=thirdparty%2Fasterisk.git 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 bcc3469a6e..d678f7bd43 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) {