diff options
| author | Olle Johansson <oej@edvina.net> | 2007-11-26 21:12:50 +0000 |
|---|---|---|
| committer | Olle Johansson <oej@edvina.net> | 2007-11-26 21:12:50 +0000 |
| commit | 5070d10864e77b85e29a4b10fbaf0fe4efcd14d1 (patch) | |
| tree | 9c9561b709e74dcf13ddc5b97babe5d72e3032e9 /channels | |
| parent | 96ad455115e8f9eb51861a9d4c1e261e1664f112 (diff) | |
Formatting, doxygenification
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 17 |
1 files changed, 10 insertions, 7 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index f91b2deff..301397f9a 100644 --- 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)) |