From 98d0d296fbd1d705beaebb8db080b3035df1a381 Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Thu, 16 Feb 2006 08:19:34 +0000 Subject: Whitespace cleanup git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10271 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 110 ++++++++++++++++++++++++++-------------------------- 1 file changed, 55 insertions(+), 55 deletions(-) (limited to 'channels') diff --git a/channels/chan_sip.c b/channels/chan_sip.c index a93fb943f..94fb400cd 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -186,7 +186,7 @@ static const struct cfsubscription_types { const char * const mediatype; const char * const text; } subscription_types[] = { - { NONE, "-", "unknown", "unknown" }, + { NONE, "-", "unknown", "unknown" }, /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */ { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" }, { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */ @@ -261,10 +261,10 @@ static const struct cfalias { { "Allow-Events", "u" }, { "Event", "o" }, { "Via", "v" }, - { "Accept-Contact", "a" }, - { "Reject-Contact", "j" }, + { "Accept-Contact", "a" }, + { "Reject-Contact", "j" }, { "Request-Disposition", "d" }, - { "Session-Expires", "x" }, + { "Session-Expires", "x" }, }; /*! Define SIP option tags, used in Require: and Supported: headers @@ -520,10 +520,10 @@ AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in /*! \brief sip_auth: Creadentials for authentication to other SIP services */ struct sip_auth { char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */ - char username[256]; /*!< Username */ - char secret[256]; /*!< Secret */ - char md5secret[256]; /*!< MD5Secret */ - struct sip_auth *next; /*!< Next auth structure in list */ + char username[256]; /*!< Username */ + char secret[256]; /*!< Secret */ + char md5secret[256]; /*!< MD5Secret */ + struct sip_auth *next; /*!< Next auth structure in list */ }; /*--- Various flags for the flags field in the pvt structure @@ -705,7 +705,7 @@ static struct sip_pvt { int rtpkeepalive; /*!< Send RTP packets for keepalive */ enum subscriptiontype subscribed; /*!< Is this dialog a subscription? */ int stateid; - int laststate; /*!< Last known extension state */ + int laststate; /*!< Last known extension state */ int dialogver; struct ast_dsp *vad; /*!< Voice Activation Detection dsp */ @@ -829,10 +829,10 @@ struct sip_peer { #define REG_STATE_UNREGISTERED 0 /*!< We are not registred */ #define REG_STATE_REGSENT 1 /*!< Registration request sent */ #define REG_STATE_AUTHSENT 2 /*!< We have tried to authenticate */ -#define REG_STATE_REGISTERED 3 /*!< Registred and done */ -#define REG_STATE_REJECTED 4 /*!< Registration rejected */ -#define REG_STATE_TIMEOUT 5 /*!< Registration timed out */ -#define REG_STATE_NOAUTH 6 /*!< We have no accepted credentials */ +#define REG_STATE_REGISTERED 3 /*!< Registred and done */ +#define REG_STATE_REJECTED 4 /*!< Registration rejected */ +#define REG_STATE_TIMEOUT 5 /*!< Registration timed out */ +#define REG_STATE_NOAUTH 6 /*!< We have no accepted credentials */ #define REG_STATE_FAILED 7 /*!< Registration failed after several tries */ @@ -887,7 +887,7 @@ static struct ast_register_list { } regl; /*! \todo Move the sip_auth list to AST_LIST */ -static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */ +static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */ /* --- Sockets and networking --------------*/ @@ -942,12 +942,12 @@ static int sip_indicate(struct ast_channel *ast, int condition); static int sip_transfer(struct ast_channel *ast, const char *dest); static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); static int sip_senddigit(struct ast_channel *ast, char digit); -static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */ -static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */ -static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); /* Find authentication for a specific realm */ +static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */ +static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */ +static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm); /* Find authentication for a specific realm */ static int check_auth(struct sip_pvt *p, struct sip_request *req, const char *username, - const char *secret, const char *md5secret, int sipmethod, - char *uri, int reliable, int ignore); + const char *secret, const char *md5secret, int sipmethod, + char *uri, int reliable, int ignore); static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */ static void append_date(struct sip_request *req); /* Append date to SIP packet */ static int determine_firstline_parts(struct sip_request *req); @@ -1140,7 +1140,7 @@ static void build_via(struct sip_pvt *p) /* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */ ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s", - ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch, rport); + ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ourport, p->branch, rport); } /*! \brief NAT fix - decide which IP address to use for ASterisk server? @@ -1212,15 +1212,15 @@ static void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap) /*! \brief Append to SIP dialog history with arg list */ static int append_history_full(struct sip_pvt *p, const char *fmt, ...) { - va_list ap; + va_list ap; if (!recordhistory || !p) return 0; - va_start(ap, fmt); - append_history_va(p, fmt, ap); - va_end(ap); + va_start(ap, fmt); + append_history_va(p, fmt, ap); + va_end(ap); - return 0; + return 0; } /*! \brief Retransmit SIP message if no answer */ @@ -2081,8 +2081,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) /* Check whether there is a VXML_URL variable */ if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) { p->options->vxml_url = ast_var_value(current); - } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) { - p->options->uri_options = ast_var_value(current); + } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) { + p->options->uri_options = ast_var_value(current); } else if (!p->options->distinctive_ring && !strcasecmp(ast_var_name(current), "ALERT_INFO")) { /* Check whether there is a ALERT_INFO variable */ p->options->distinctive_ring = ast_var_value(current); @@ -2307,8 +2307,8 @@ static int update_call_counter(struct sip_pvt *fup, int event) /* incoming and outgoing affects the inUse counter */ case DEC_CALL_LIMIT: if ( *inuse > 0 ) { - if (ast_test_flag(fup, SIP_INC_COUNT)) - (*inuse)--; + if (ast_test_flag(fup, SIP_INC_COUNT)) + (*inuse)--; } else { *inuse = 0; } @@ -2328,7 +2328,7 @@ static int update_call_counter(struct sip_pvt *fup, int event) } } (*inuse)++; - ast_set_flag(fup, SIP_INC_COUNT); + ast_set_flag(fup, SIP_INC_COUNT); if (option_debug > 1 || sipdebug) { ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit); } @@ -2474,9 +2474,9 @@ static char *hangup_cause2sip(int cause) case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */ case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */ return "404 Not Found"; - case AST_CAUSE_CONGESTION: /* 34 */ - case AST_CAUSE_SWITCH_CONGESTION: /* 42 */ - return "503 Service Unavailable"; + case AST_CAUSE_CONGESTION: /* 34 */ + case AST_CAUSE_SWITCH_CONGESTION: /* 42 */ + return "503 Service Unavailable"; case AST_CAUSE_NO_USER_RESPONSE: /* 18 */ return "408 Request Timeout"; case AST_CAUSE_NO_ANSWER: /* 19 */ @@ -2492,7 +2492,7 @@ static char *hangup_cause2sip(int cause) case AST_CAUSE_USER_BUSY: return "486 Busy here"; case AST_CAUSE_FAILURE: - return "500 Server internal failure"; + return "500 Server internal failure"; case AST_CAUSE_FACILITY_REJECTED: /* 29 */ return "501 Not Implemented"; case AST_CAUSE_CHAN_NOT_IMPLEMENTED: @@ -3783,8 +3783,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) } /* Manager Hold and Unhold events must be generated, if necessary */ - if (sin.sin_addr.s_addr && !sendonly) { - append_history(p, "Unhold", "%s", req->data); + if (sin.sin_addr.s_addr && !sendonly) { + append_history(p, "Unhold", "%s", req->data); if (global_callevents && ast_test_flag(p, SIP_CALL_ONHOLD)) { manager_event(EVENT_FLAG_CALL, "Unhold", @@ -3793,16 +3793,16 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) p->owner->name, p->owner->uniqueid); - } + } ast_clear_flag(p, SIP_CALL_ONHOLD); - } else { + } else { /* No address for RTP, we're on hold */ - append_history(p, "Hold", "%s", req->data); + append_history(p, "Hold", "%s", req->data); - if (global_callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) { + if (global_callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) { manager_event(EVENT_FLAG_CALL, "Hold", "Channel: %s\r\n" - "Uniqueid: %s\r\n", + "Uniqueid: %s\r\n", p->owner->name, p->owner->uniqueid); } @@ -3973,8 +3973,8 @@ static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, struct s /* Add rport to first VIA header if requested */ /* Whoo hoo! Now we can indicate port address translation too! Just - another RFC (RFC3581). I'll leave the original comments in for - posterity. */ + another RFC (RFC3581). I'll leave the original comments in for + posterity. */ 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)); } else { /* We should *always* add a received to the topmost via */ @@ -4880,10 +4880,10 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho x=0; /* Test p->username against allowed characters in AST_DIGIT_ANY - If it matches the allowed characters list, then sipuser = ";user=phone" - If not, then sipuser = "" - */ - /* + is allowed in first position in a tel: uri */ + If it matches the allowed characters list, then sipuser = ";user=phone" + If not, then sipuser = "" + */ + /* + is allowed in first position in a tel: uri */ if (p->username && p->username[0] == '+') x=1; @@ -5048,7 +5048,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init) ast_log(LOG_WARNING,"No Headp for the channel...ooops!\n"); else { AST_LIST_TRAVERSE(headp, current, entries) { - /* SIPADDHEADER: Add SIP header to outgoing call */ + /* SIPADDHEADER: Add SIP header to outgoing call */ if (!strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) { char *content, *end; const char *header = ast_var_value(current); @@ -6853,7 +6853,7 @@ static int get_refer_info(struct sip_pvt *sip_pvt, struct sip_request *outgoing_ } else { 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); /* XXX The refer_to could contain a call on an entirely different machine, requiring an - INVITE with a replaces header -anthm XXX */ + INVITE with a replaces header -anthm XXX */ /* The only way to find out is to use the dialplan - oej */ } } else if (ast_exists_extension(NULL, sip_pvt->context, refer_to, 1, NULL) || !strcmp(refer_to, ast_parking_ext())) { @@ -7372,7 +7372,7 @@ static int get_msg_text(char *buf, int len, struct sip_request *req) return 0; } - + /*! \brief Receive SIP MESSAGE method messages \note We only handle messages within current calls currently Reference: RFC 3428 */ @@ -7502,7 +7502,7 @@ static int peer_status(struct sip_peer *peer, char *status, int statuslen) } return res; } - + /*! \brief CLI Command 'SIP Show Users' */ static int sip_show_users(int fd, int argc, char *argv[]) { @@ -8440,9 +8440,9 @@ static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions } if (cur->subscribed != NONE && subscriptions) { ast_cli(fd, FORMAT3, ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), - ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username, - cur->callid, cur->exten, ast_extension_state2str(cur->laststate), - subscription_type2str(cur->subscribed)); + ast_strlen_zero(cur->username) ? ( ast_strlen_zero(cur->cid_num) ? "(None)" : cur->cid_num ) : cur->username, + cur->callid, cur->exten, ast_extension_state2str(cur->laststate), + subscription_type2str(cur->subscribed)); numchans++; } cur = cur->next; @@ -10021,7 +10021,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_ update_call_counter(p, DEC_CALL_LIMIT); break; case 482: /* SIP is incapable of performing a hairpin call, which - is yet another failure of not having a layer 2 (again, YAY + is yet another failure of not having a layer 2 (again, YAY IETF for thinking ahead). So we treat this as a call forward and hope we end up at the right place... */ ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n"); -- cgit v1.2.3