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asterisk
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Formatting fixes

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Olle Johansson 19 years ago
parent 3aa0f7897f
commit 4b6d6a1a0b
1 changed files with 2 additions and 3 deletions
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5
channels/chan_sip.c
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@ -3887,7 +3887,7 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
{ {
struct ast_frame *fr; struct ast_frame *fr;
struct sip_pvt *p = ast->tech_pvt; struct sip_pvt *p = ast->tech_pvt;
int faxdetected = 0; int faxdetected = FALSE;
ast_mutex_lock(&p->lock); ast_mutex_lock(&p->lock);
fr = sip_rtp_read(ast, p, &faxdetected); fr = sip_rtp_read(ast, p, &faxdetected);
@ -4024,9 +4024,8 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
ast_rtp_settos(p->vrtp, global_tos_video); ast_rtp_settos(p->vrtp, global_tos_video);
ast_rtp_setdtmf(p->vrtp, 0); ast_rtp_setdtmf(p->vrtp, 0);
} }
if (p->udptl) { if (p->udptl)
ast_udptl_settos(p->udptl, global_tos_audio); ast_udptl_settos(p->udptl, global_tos_audio);
}
p->rtptimeout = global_rtptimeout; p->rtptimeout = global_rtptimeout;
p->rtpholdtimeout = global_rtpholdtimeout; p->rtpholdtimeout = global_rtpholdtimeout;
p->rtpkeepalive = global_rtpkeepalive; p->rtpkeepalive = global_rtpkeepalive;

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