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Formatting, doxygenification

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89611 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej 2007-11-26 21:12:50 +00:00
parent 812477c1c5
commit 4e62295db7
1 changed files with 10 additions and 7 deletions
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17
channels/chan_sip.c
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@ -2753,8 +2753,9 @@ static char *get_in_brackets(char *tmp)
/*! \brief * parses a URI in its components.
*
* \note
*- If scheme is specified, drop it from the top.
* - If scheme is specified, drop it from the top.
* - If a component is not requested, do not split around it.
*
* This means that if we don't have domain, we cannot split
* name:pass and domain:port.
* It is safe to call with ret_name, pass, domain, port
@ -4576,6 +4577,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
int needtext = 0;
char buf[BUFSIZ];
char *decoded_exten;
{
const char *my_name; /* pick a good name */
@ -4658,6 +4660,8 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
if (global_relaxdtmf)
ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
}
/* Set file descriptors for audio, video, realtime text and UDPTL as needed */
if (i->rtp) {
ast_channel_set_fd(tmp, 0, ast_rtp_fd(i->rtp));
ast_channel_set_fd(tmp, 1, ast_rtcp_fd(i->rtp));
@ -4666,12 +4670,11 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
ast_channel_set_fd(tmp, 2, ast_rtp_fd(i->vrtp));
ast_channel_set_fd(tmp, 3, ast_rtcp_fd(i->vrtp));
}
if (needtext && i->trtp) {
if (needtext && i->trtp)
ast_channel_set_fd(tmp, 4, ast_rtp_fd(i->trtp));
}
if (i->udptl) {
if (i->udptl)
ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));
}
if (state == AST_STATE_RING)
tmp->rings = 1;
tmp->adsicpe = AST_ADSI_UNAVAILABLE;
@ -17102,14 +17105,13 @@ static int sip_devicestate(void *data)
SIP calls initiated by the PBX arrive here */
static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
{
int oldformat;
struct sip_pvt *p;
struct ast_channel *tmpc = NULL;
char *ext, *host;
char tmp[256];
char *dest = data;
int oldformat = format;
oldformat = format;
/* mask request with some set of allowed formats.
* XXX this needs to be fixed.
* The original code uses AST_FORMAT_AUDIO_MASK, but it is
@ -17192,6 +17194,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
return tmpc;
}
/*! Parse insecure= setting in sip.conf and set flags according to setting */
static void set_insecure_flags (struct ast_flags *flags, const char *value, int lineno)
{
if (ast_strlen_zero(value))