Formatting, doxygenification

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89611 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f91b2de..301397f 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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))