More formatting fixes and doxygen stuff
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
parent
d37002287b
commit
da55c166dc
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@ -2403,9 +2403,9 @@ static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int
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else
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p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp);
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if (!p && realtime) {
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if (!p && realtime)
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p = realtime_peer(peer, sin);
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}
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return p;
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}
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@ -2691,16 +2691,16 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
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} else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
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/* Check whether there is a variable with a name starting with SIPADDHEADER */
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p->options->addsipheaders = 1;
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} else if (!strcasecmp(ast_var_name(current),"SIPTRANSFER")) {
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} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
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/* This is a transfered call */
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p->options->transfer = 1;
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} else if (!strcasecmp(ast_var_name(current),"SIPTRANSFER_REFERER")) {
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} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
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/* This is the referer */
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referer = ast_var_value(current);
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} else if (!strcasecmp(ast_var_name(current),"SIPTRANSFER_REPLACES")) {
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} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
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/* We're replacing a call. */
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p->options->replaces = ast_var_value(current);
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} else if (!strcasecmp(ast_var_name(current),"T38CALL")) {
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} else if (!strcasecmp(ast_var_name(current), "T38CALL")) {
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p->t38.state = T38_LOCAL_DIRECT;
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if (option_debug)
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ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
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@ -3186,9 +3186,8 @@ static int sip_hangup(struct ast_channel *ast)
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else
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ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid);
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}
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if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE)) {
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if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE))
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ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n");
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}
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ast_mutex_lock(&p->lock);
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if (option_debug && sipdebug)
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@ -3338,9 +3337,8 @@ static int sip_answer(struct ast_channel *ast)
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if (option_debug > 1)
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ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
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res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
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} else {
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} else
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res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
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}
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}
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ast_mutex_unlock(&p->lock);
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return res;
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@ -5576,8 +5574,8 @@ static int add_digit(struct sip_request *req, char digit)
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return 0;
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}
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/*! \brief add XML encoded media control with update */
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/*! \note XML: The only way to turn 0 bits of information into a few hundred. */
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/*! \brief add XML encoded media control with update
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\note XML: The only way to turn 0 bits of information into a few hundred. (markster) */
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static int add_vidupdate(struct sip_request *req)
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{
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const char *xml_is_a_huge_waste_of_space =
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@ -5704,9 +5702,8 @@ static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p)
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udptldest.sin_port = udptlsin.sin_port;
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}
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if (debug) {
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if (debug)
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ast_log(LOG_DEBUG, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(udptlsin.sin_port));
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}
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/* We break with the "recommendation" and send our IP, in order that our
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peer doesn't have to ast_gethostbyname() us */
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@ -6049,7 +6046,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
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return 0;
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}
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/*--- transmit_response_with_t38_sdp: Used for 200 OK and 183 early media ---*/
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/*! \brief Used for 200 OK and 183 early media */
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static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
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{
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struct sip_request resp;
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@ -6063,9 +6060,8 @@ static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct s
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if (p->udptl) {
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ast_udptl_offered_from_local(p->udptl, 0);
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add_t38_sdp(&resp, p);
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} else {
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} else
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ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
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}
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return send_response(p, &resp, retrans, seqno);
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}
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@ -6097,9 +6093,8 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
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if (p->rtp) {
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try_suggested_sip_codec(p);
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add_sdp(&resp, p);
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} else {
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} else
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ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
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}
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return send_response(p, &resp, reliable, seqno);
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}
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@ -6405,9 +6400,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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} else if (p->options && p->options->vxml_url) {
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/* If there is a VXML URL append it to the SIP URL */
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snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
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} else {
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} else
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snprintf(to, sizeof(to), "<%s>", p->uri);
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}
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init_req(req, sipmethod, p->uri);
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snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);
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@ -6419,9 +6413,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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if (ast_test_flag(&p->flags[0], SIP_SENDRPID) && (sipmethod == SIP_INVITE)) {
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build_rpid(p);
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add_header(req, "From", p->rpid_from);
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} else {
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} else
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add_header(req, "From", from);
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}
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add_header(req, "To", to);
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ast_string_field_set(p, exten, l);
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build_contact(p);
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@ -6472,9 +6465,8 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
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add_header(&req, "Require", "replaces");
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}
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if (p->options && !ast_strlen_zero(p->options->distinctive_ring)) {
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if (p->options && !ast_strlen_zero(p->options->distinctive_ring))
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add_header(&req, "Alert-Info", p->options->distinctive_ring);
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}
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add_header(&req, "Allow", ALLOWED_METHODS);
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add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
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if (p->options && p->options->addsipheaders ) {
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@ -6522,9 +6514,8 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
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if (option_debug)
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ast_log(LOG_DEBUG, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
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add_t38_sdp(&req, p);
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} else if (p->rtp) {
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} else if (p->rtp)
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add_sdp(&req, p);
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}
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} else {
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add_header_contentLength(&req, 0);
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}
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