Whitespace cleanup
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10271 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -186,7 +186,7 @@ static const struct cfsubscription_types {
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const char * const mediatype;
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const char * const text;
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} subscription_types[] = {
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{ NONE, "-", "unknown", "unknown" },
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{ NONE, "-", "unknown", "unknown" },
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/* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
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{ DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
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{ CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
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@ -261,10 +261,10 @@ static const struct cfalias {
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{ "Allow-Events", "u" },
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{ "Event", "o" },
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{ "Via", "v" },
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{ "Accept-Contact", "a" },
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{ "Reject-Contact", "j" },
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{ "Accept-Contact", "a" },
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{ "Reject-Contact", "j" },
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{ "Request-Disposition", "d" },
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{ "Session-Expires", "x" },
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{ "Session-Expires", "x" },
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};
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/*! Define SIP option tags, used in Require: and Supported: headers
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@ -520,10 +520,10 @@ AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in
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/*! \brief sip_auth: Creadentials for authentication to other SIP services */
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struct sip_auth {
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char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
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char username[256]; /*!< Username */
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char secret[256]; /*!< Secret */
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char md5secret[256]; /*!< MD5Secret */
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struct sip_auth *next; /*!< Next auth structure in list */
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char username[256]; /*!< Username */
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char secret[256]; /*!< Secret */
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char md5secret[256]; /*!< MD5Secret */
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struct sip_auth *next; /*!< Next auth structure in list */
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};
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/*--- Various flags for the flags field in the pvt structure
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@ -705,7 +705,7 @@ static struct sip_pvt {
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int rtpkeepalive; /*!< Send RTP packets for keepalive */
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enum subscriptiontype subscribed; /*!< Is this dialog a subscription? */
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int stateid;
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int laststate; /*!< Last known extension state */
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int laststate; /*!< Last known extension state */
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int dialogver;
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struct ast_dsp *vad; /*!< Voice Activation Detection dsp */
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@ -829,10 +829,10 @@ struct sip_peer {
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#define REG_STATE_UNREGISTERED 0 /*!< We are not registred */
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#define REG_STATE_REGSENT 1 /*!< Registration request sent */
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#define REG_STATE_AUTHSENT 2 /*!< We have tried to authenticate */
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#define REG_STATE_REGISTERED 3 /*!< Registred and done */
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#define REG_STATE_REJECTED 4 /*!< Registration rejected */
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#define REG_STATE_TIMEOUT 5 /*!< Registration timed out */
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#define REG_STATE_NOAUTH 6 /*!< We have no accepted credentials */
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#define REG_STATE_REGISTERED 3 /*!< Registred and done */
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#define REG_STATE_REJECTED 4 /*!< Registration rejected */
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#define REG_STATE_TIMEOUT 5 /*!< Registration timed out */
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#define REG_STATE_NOAUTH 6 /*!< We have no accepted credentials */
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#define REG_STATE_FAILED 7 /*!< Registration failed after several tries */
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@ -887,7 +887,7 @@ static struct ast_register_list {
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} regl;
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/*! \todo Move the sip_auth list to AST_LIST */
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static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
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static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
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/* --- Sockets and networking --------------*/
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@ -942,12 +942,12 @@ static int sip_indicate(struct ast_channel *ast, int condition);
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static int sip_transfer(struct ast_channel *ast, const char *dest);
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static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
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static int sip_senddigit(struct ast_channel *ast, char digit);
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static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
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static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */
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static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); /* Find authentication for a specific realm */
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static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
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static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */
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static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); /* Find authentication for a specific realm */
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static int check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
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const char *secret, const char *md5secret, int sipmethod,
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char *uri, int reliable, int ignore);
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const char *secret, const char *md5secret, int sipmethod,
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char *uri, int reliable, int ignore);
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static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
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static void append_date(struct sip_request *req); /* Append date to SIP packet */
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static int determine_firstline_parts(struct sip_request *req);
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@ -1140,7 +1140,7 @@ static void build_via(struct sip_pvt *p)
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/* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
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ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
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ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch, rport);
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ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch, rport);
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}
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/*! \brief NAT fix - decide which IP address to use for ASterisk server?
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@ -1212,15 +1212,15 @@ static void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
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/*! \brief Append to SIP dialog history with arg list */
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static int append_history_full(struct sip_pvt *p, const char *fmt, ...)
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{
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va_list ap;
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va_list ap;
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if (!recordhistory || !p)
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return 0;
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va_start(ap, fmt);
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append_history_va(p, fmt, ap);
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va_end(ap);
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va_start(ap, fmt);
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append_history_va(p, fmt, ap);
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va_end(ap);
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return 0;
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return 0;
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}
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/*! \brief Retransmit SIP message if no answer */
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@ -2081,8 +2081,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
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/* Check whether there is a VXML_URL variable */
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if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
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p->options->vxml_url = ast_var_value(current);
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} else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
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p->options->uri_options = ast_var_value(current);
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} else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
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p->options->uri_options = ast_var_value(current);
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} else if (!p->options->distinctive_ring && !strcasecmp(ast_var_name(current), "ALERT_INFO")) {
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/* Check whether there is a ALERT_INFO variable */
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p->options->distinctive_ring = ast_var_value(current);
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@ -2307,8 +2307,8 @@ static int update_call_counter(struct sip_pvt *fup, int event)
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/* incoming and outgoing affects the inUse counter */
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case DEC_CALL_LIMIT:
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if ( *inuse > 0 ) {
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if (ast_test_flag(fup, SIP_INC_COUNT))
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(*inuse)--;
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if (ast_test_flag(fup, SIP_INC_COUNT))
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(*inuse)--;
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} else {
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*inuse = 0;
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}
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@ -2328,7 +2328,7 @@ static int update_call_counter(struct sip_pvt *fup, int event)
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}
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}
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(*inuse)++;
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ast_set_flag(fup, SIP_INC_COUNT);
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ast_set_flag(fup, SIP_INC_COUNT);
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if (option_debug > 1 || sipdebug) {
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ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
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}
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@ -2474,9 +2474,9 @@ static char *hangup_cause2sip(int cause)
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case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
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case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
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return "404 Not Found";
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case AST_CAUSE_CONGESTION: /* 34 */
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case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
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return "503 Service Unavailable";
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case AST_CAUSE_CONGESTION: /* 34 */
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case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
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return "503 Service Unavailable";
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case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
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return "408 Request Timeout";
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case AST_CAUSE_NO_ANSWER: /* 19 */
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@ -2492,7 +2492,7 @@ static char *hangup_cause2sip(int cause)
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case AST_CAUSE_USER_BUSY:
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return "486 Busy here";
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case AST_CAUSE_FAILURE:
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return "500 Server internal failure";
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return "500 Server internal failure";
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case AST_CAUSE_FACILITY_REJECTED: /* 29 */
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return "501 Not Implemented";
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case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
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@ -3783,8 +3783,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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}
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/* Manager Hold and Unhold events must be generated, if necessary */
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if (sin.sin_addr.s_addr && !sendonly) {
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append_history(p, "Unhold", "%s", req->data);
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if (sin.sin_addr.s_addr && !sendonly) {
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append_history(p, "Unhold", "%s", req->data);
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if (global_callevents && ast_test_flag(p, SIP_CALL_ONHOLD)) {
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manager_event(EVENT_FLAG_CALL, "Unhold",
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@ -3793,16 +3793,16 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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p->owner->name,
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p->owner->uniqueid);
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}
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}
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ast_clear_flag(p, SIP_CALL_ONHOLD);
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} else {
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} else {
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/* No address for RTP, we're on hold */
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append_history(p, "Hold", "%s", req->data);
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append_history(p, "Hold", "%s", req->data);
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if (global_callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) {
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if (global_callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) {
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manager_event(EVENT_FLAG_CALL, "Hold",
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"Channel: %s\r\n"
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"Uniqueid: %s\r\n",
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"Uniqueid: %s\r\n",
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p->owner->name,
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p->owner->uniqueid);
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}
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@ -3973,8 +3973,8 @@ static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, struct s
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/* Add rport to first VIA header if requested */
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/* Whoo hoo! Now we can indicate port address translation too! Just
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another RFC (RFC3581). I'll leave the original comments in for
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posterity. */
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another RFC (RFC3581). I'll leave the original comments in for
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posterity. */
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snprintf(new, sizeof(new), "%s;received=%s;rport=%d", tmp, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port));
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} else {
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/* We should *always* add a received to the topmost via */
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@ -4880,10 +4880,10 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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x=0;
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/* Test p->username against allowed characters in AST_DIGIT_ANY
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If it matches the allowed characters list, then sipuser = ";user=phone"
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If not, then sipuser = ""
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*/
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/* + is allowed in first position in a tel: uri */
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If it matches the allowed characters list, then sipuser = ";user=phone"
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If not, then sipuser = ""
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*/
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/* + is allowed in first position in a tel: uri */
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if (p->username && p->username[0] == '+')
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x=1;
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@ -5048,7 +5048,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
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ast_log(LOG_WARNING,"No Headp for the channel...ooops!\n");
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else {
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AST_LIST_TRAVERSE(headp, current, entries) {
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/* SIPADDHEADER: Add SIP header to outgoing call */
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/* SIPADDHEADER: Add SIP header to outgoing call */
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if (!strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
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char *content, *end;
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const char *header = ast_var_value(current);
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@ -6853,7 +6853,7 @@ static int get_refer_info(struct sip_pvt *sip_pvt, struct sip_request *outgoing_
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} else {
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ast_log(LOG_NOTICE, "Supervised transfer requested, but unable to find callid '%s'. Both legs must reside on Asterisk box to transfer at this time.\n", replace_callid);
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/* XXX The refer_to could contain a call on an entirely different machine, requiring an
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INVITE with a replaces header -anthm XXX */
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INVITE with a replaces header -anthm XXX */
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/* The only way to find out is to use the dialplan - oej */
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}
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} else if (ast_exists_extension(NULL, sip_pvt->context, refer_to, 1, NULL) || !strcmp(refer_to, ast_parking_ext())) {
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@ -7372,7 +7372,7 @@ static int get_msg_text(char *buf, int len, struct sip_request *req)
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return 0;
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}
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/*! \brief Receive SIP MESSAGE method messages
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\note We only handle messages within current calls currently
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Reference: RFC 3428 */
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@ -7502,7 +7502,7 @@ static int peer_status(struct sip_peer *peer, char *status, int statuslen)
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}
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return res;
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}
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/*! \brief CLI Command 'SIP Show Users' */
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static int sip_show_users(int fd, int argc, char *argv[])
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{
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@ -8440,9 +8440,9 @@ static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions
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}
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if (cur->subscribed != NONE && subscriptions) {
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ast_cli(fd, FORMAT3, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr),
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ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username,
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cur->callid, cur->exten, ast_extension_state2str(cur->laststate),
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subscription_type2str(cur->subscribed));
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ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username,
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cur->callid, cur->exten, ast_extension_state2str(cur->laststate),
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subscription_type2str(cur->subscribed));
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numchans++;
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}
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cur = cur->next;
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update_call_counter(p, DEC_CALL_LIMIT);
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break;
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case 482: /* SIP is incapable of performing a hairpin call, which
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is yet another failure of not having a layer 2 (again, YAY
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is yet another failure of not having a layer 2 (again, YAY
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IETF for thinking ahead). So we treat this as a call
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forward and hope we end up at the right place... */
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ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n");
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