[asterisk-commits] trunk r10271 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Thu Feb 16 01:19:37 MST 2006
Author: oej
Date: Thu Feb 16 02:19:34 2006
New Revision: 10271
URL: http://svn.digium.com/view/asterisk?rev=10271&view=rev
Log:
Whitespace cleanup
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=10271&r1=10270&r2=10271&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Feb 16 02:19:34 2006
@@ -186,7 +186,7 @@
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 @@
{ "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 @@
/*! \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 @@
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 @@
#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 @@
} 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_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 @@
/* 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 @@
/*! \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);
-
- return 0;
+ va_start(ap, fmt);
+ append_history_va(p, fmt, ap);
+ va_end(ap);
+
+ return 0;
}
/*! \brief Retransmit SIP message if no answer */
@@ -2081,8 +2081,8 @@
/* 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 @@
/* 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 @@
}
}
(*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 @@
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 @@
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 @@
}
/* 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 @@
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);
-
- if (global_callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) {
+ append_history(p, "Hold", "%s", req->data);
+
+ 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 @@
/* 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 @@
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 @@
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 @@
} 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 @@
return 0;
}
-
+
/*! \brief Receive SIP MESSAGE method messages
\note We only handle messages within current calls currently
Reference: RFC 3428 */
@@ -7502,7 +7502,7 @@
}
return res;
}
-
+
/*! \brief CLI Command 'SIP Show Users' */
static int sip_show_users(int fd, int argc, char *argv[])
{
@@ -8440,9 +8440,9 @@
}
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 @@
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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