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asterisk
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Whitespace cleanup

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Olle Johansson 21 years ago
parent 6f8cdb5c03
commit 98d0d296fb
1 changed files with 55 additions and 55 deletions
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110
channels/chan_sip.c
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@ -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");

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