Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity

whitespace changes related to previous commit

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Luigi Rizzo 20 years ago
parent e0f745b1f6
commit fde338ba2c
1 changed files with 140 additions and 140 deletions
Show Stats Download Patch File Download Diff File

280
channels/chan_sip.c
Unescape View File

@ -1703,7 +1703,7 @@ static int __sip_xmit(struct sip_pvt *p, char *data, int len)
{
int res;
const struct sockaddr_in *dst = sip_real_dst(p);
res=sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
if (res != len)
ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));
@ -2015,7 +2015,7 @@ static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod, int
(!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
if (!resp && (seqno == p->pendinginvite)) {
if (option_debug)
ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
p->pendinginvite = 0;
}
/* this is our baby */
@ -2945,79 +2945,79 @@ static int update_call_counter(struct sip_pvt *fup, int event)
call_limit = &u->call_limit;
inringing = NULL;
} else if ( (p = find_peer(fup->peername, NULL, 1) ) ) { /* Try to find peer */
inuse = &p->inUse;
call_limit = &p->call_limit;
inringing = &p->inRinging;
ast_copy_string(name, fup->peername, sizeof(name));
} else {
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
return 0;
}
inuse = &p->inUse;
call_limit = &p->call_limit;
inringing = &p->inRinging;
ast_copy_string(name, fup->peername, sizeof(name));
} else {
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
return 0;
}
switch(event) {
/* incoming and outgoing affects the inUse counter */
case DEC_CALL_LIMIT:
if ( *inuse > 0 ) {
if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT))
(*inuse)--;
} else {
*inuse = 0;
}
if (inringing) {
if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
if (*inringing > 0)
(*inringing)--;
else
ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername);
ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
}
if (option_debug > 1 || sipdebug) {
ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
/* incoming and outgoing affects the inUse counter */
case DEC_CALL_LIMIT:
if ( *inuse > 0 ) {
if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT))
(*inuse)--;
} else {
*inuse = 0;
}
if (inringing) {
if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
if (*inringing > 0)
(*inringing)--;
else
ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername);
ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
break;
}
if (option_debug > 1 || sipdebug) {
ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
}
break;
case INC_CALL_RINGING:
case INC_CALL_LIMIT:
if (*call_limit > 0 ) {
if (*inuse >= *call_limit) {
ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
if (u)
ASTOBJ_UNREF(u, sip_destroy_user);
else
ASTOBJ_UNREF(p, sip_destroy_peer);
return -1;
}
}
if (inringing && (event == INC_CALL_RINGING)) {
if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
(*inringing)++;
ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
case INC_CALL_RINGING:
case INC_CALL_LIMIT:
if (*call_limit > 0 ) {
if (*inuse >= *call_limit) {
ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
if (u)
ASTOBJ_UNREF(u, sip_destroy_user);
else
ASTOBJ_UNREF(p, sip_destroy_peer);
return -1;
}
/* Continue */
(*inuse)++;
ast_set_flag(&fup->flags[0], 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);
}
if (inringing && (event == INC_CALL_RINGING)) {
if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
(*inringing)++;
ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
break;
}
/* Continue */
(*inuse)++;
ast_set_flag(&fup->flags[0], 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);
}
break;
case DEC_CALL_RINGING:
if (inringing) {
if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
if (*inringing > 0)
(*inringing)--;
else
ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", p->name);
ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
case DEC_CALL_RINGING:
if (inringing) {
if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
if (*inringing > 0)
(*inringing)--;
else
ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", p->name);
ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
break;
}
break;
default:
ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
default:
ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
}
if (p) {
ast_device_state_changed("SIP/%s", p->name);
@ -3189,7 +3189,7 @@ static const char *hangup_cause2sip(int cause)
case AST_CAUSE_NOTDEFINED:
default:
if (option_debug)
ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause);
ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause);
return NULL;
}
@ -3209,7 +3209,7 @@ static int sip_hangup(struct ast_channel *ast)
if (!p) {
if (option_debug)
ast_log(LOG_DEBUG, "Asked to hangup channel that was not connected\n");
ast_log(LOG_DEBUG, "Asked to hangup channel that was not connected\n");
return 0;
}
@ -3230,8 +3230,8 @@ static int sip_hangup(struct ast_channel *ast)
ast_log(LOG_DEBUG, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
else {
if (option_debug)
ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid);
}
ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid);
}
}
if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE))
ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n");
@ -3728,11 +3728,11 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
{
const char *my_name; /* pick a good name */
if (title)
if (title)
my_name = title;
else if ( (my_name = strchr(i->fromdomain,':')) )
my_name++; /* skip ':' */
else
else
my_name = i->fromdomain;
ast_string_field_build(tmp, name, "SIP/%s-%08x", my_name, (int)(long) i);
}
@ -4286,7 +4286,7 @@ static int sip_register(char *value, int lineno)
char *hostname=NULL, *secret=NULL, *authuser=NULL;
char *porta=NULL;
char *contact=NULL;
if (!value)
return -1;
ast_copy_string(username, value, sizeof(username));
@ -4317,9 +4317,9 @@ static int sip_register(char *value, int lineno)
*porta++ = '\0';
portnum = atoi(porta);
if (portnum == 0) {
ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno);
return -1;
}
ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno);
return -1;
}
}
if (!(reg = ast_calloc(1, sizeof(*reg)))) {
ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n");
@ -4809,7 +4809,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
if (framing == LONG_MIN || framing == LONG_MAX) {
framing = 0;
if (option_debug)
ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a);
ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a);
}
}
if (framing && last_rtpmap_codec) {
@ -6414,7 +6414,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
for (; *s; s++) {
if (!strchr(AST_DIGIT_ANYNUM, *s) )
break;
}
}
/* If we have only digits, add ;user=phone to the uri */
if (*s)
urioptions = ";user=phone";
@ -6594,7 +6594,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
}
}
}
}
}
if (sdp) {
if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
ast_udptl_offered_from_local(p->udptl, 1);
@ -7044,7 +7044,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, const char *
}
r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, r);
if (option_debug)
ast_log(LOG_DEBUG, "Scheduled a registration timeout for %s id #%d \n", r->hostname, r->timeout);
ast_log(LOG_DEBUG, "Scheduled a registration timeout for %s id #%d \n", r->hostname, r->timeout);
}
if (strchr(r->username, '@')) {
@ -7654,7 +7654,7 @@ static void build_route(struct sip_pvt *p, struct sip_request *req, int backward
/* Once a persistant route is set, don't fool with it */
if (p->route && p->route_persistant) {
if (option_debug)
ast_log(LOG_DEBUG, "build_route: Retaining previous route: <%s>\n", p->route->hop);
ast_log(LOG_DEBUG, "build_route: Retaining previous route: <%s>\n", p->route->hop);
return;
}
@ -8252,7 +8252,7 @@ static int get_destination(struct sip_pvt *p, struct sip_request *oreq)
if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) {
if (!allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
if (option_debug)
ast_log(LOG_DEBUG, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
ast_log(LOG_DEBUG, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
return -2;
}
}
@ -11256,7 +11256,7 @@ static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
if (e)
*e = '\0';
if (option_debug)
ast_log(LOG_DEBUG, "Found promiscuous redirection to 'SIP/%s'\n", s);
ast_log(LOG_DEBUG, "Found promiscuous redirection to 'SIP/%s'\n", s);
if (p->owner)
ast_string_field_build(p->owner, call_forward, "SIP/%s", s);
} else {
@ -11321,7 +11321,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) { /* This call is already gone */
if (option_debug)
ast_log(LOG_DEBUG, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
ast_log(LOG_DEBUG, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
return;
}
@ -11640,10 +11640,10 @@ static int handle_response_register(struct sip_pvt *p, int resp, char *rest, str
manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelDriver: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate));
r->regattempts = 0;
if (option_debug)
ast_log(LOG_DEBUG, "Registration successful\n");
ast_log(LOG_DEBUG, "Registration successful\n");
if (r->timeout > -1) {
if (option_debug)
ast_log(LOG_DEBUG, "Cancelling timeout %d\n", r->timeout);
ast_log(LOG_DEBUG, "Cancelling timeout %d\n", r->timeout);
ast_sched_del(sched, r->timeout);
}
r->timeout=-1;
@ -11935,7 +11935,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
handle_response_invite(p, resp, rest, req, seqno);
else {
if (option_debug)
ast_log(LOG_DEBUG, "Got 491 on %s, unspported. Call ID %s\n", sip_methods[sipmethod].text, p->callid);
ast_log(LOG_DEBUG, "Got 491 on %s, unspported. Call ID %s\n", sip_methods[sipmethod].text, p->callid);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
break;
@ -11988,7 +11988,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
IETF for thinking ahead). So we treat this as a call
forward and hope we end up at the right place... */
if (option_debug)
ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n");
ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n");
if (p->owner)
ast_string_field_build(p->owner, call_forward,
"Local/%s@%s", p->username, p->context);
@ -12055,7 +12055,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_CANCEL) {
if (option_debug)
ast_log(LOG_DEBUG, "Got 200 OK on CANCEL\n");
ast_log(LOG_DEBUG, "Got 200 OK on CANCEL\n");
/* Wait for 487, then destroy */
} else if (sipmethod == SIP_NOTIFY) {
@ -12282,11 +12282,11 @@ static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct
if (!transferer || !transferee) {
if (!transferer) {
if (option_debug)
ast_log(LOG_DEBUG, "No transferer channel, giving up parking\n");
ast_log(LOG_DEBUG, "No transferer channel, giving up parking\n");
}
if (!transferee) {
if (option_debug)
ast_log(LOG_DEBUG, "No transferee channel, giving up parking\n");
ast_log(LOG_DEBUG, "No transferee channel, giving up parking\n");
}
return -1;
}
@ -12924,7 +12924,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
} else {
p->jointcapability = p->capability;
if (option_debug)
ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
}
if (recordhistory) /* This is a response, note what it was for */
append_history(p, "ReInv", "Re-invite received");
@ -13326,7 +13326,7 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
transferer->refer->status = REFER_200OK;
if (targetcall_pvt->owner) {
if (option_debug)
ast_log(LOG_DEBUG, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name);
ast_log(LOG_DEBUG, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name);
ast_channel_unlock(targetcall_pvt->owner);
}
}
@ -13724,20 +13724,20 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
/* Get RTCP quality before end of call */
if (recordhistory || p->owner) {
char *audioqos, *videoqos;
if (p->rtp) {
if (p->rtp) {
audioqos = ast_rtp_get_quality(p->rtp);
if (recordhistory)
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
}
if (p->vrtp) {
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
}
if (p->vrtp) {
videoqos = ast_rtp_get_quality(p->vrtp);
if (recordhistory)
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
}
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
}
}
stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
@ -13809,17 +13809,17 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
transmit_response(p, "403 Forbidden (within dialog)", req);
/* Do not destroy session, since we will break the call if we do */
if (option_debug)
ast_log(LOG_DEBUG, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text);
ast_log(LOG_DEBUG, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text);
return 0;
} else if (ast_test_flag(req, SIP_PKT_DEBUG)) {
if (option_debug) {
if (resubscribe)
ast_log(LOG_DEBUG, "Got a re-subscribe on existing subscription %s\n", p->callid);
else
ast_log(LOG_DEBUG, "Got a new subscription %s (possibly with auth)\n", p->callid);
if (resubscribe)
ast_log(LOG_DEBUG, "Got a re-subscribe on existing subscription %s\n", p->callid);
else
ast_log(LOG_DEBUG, "Got a new subscription %s (possibly with auth)\n", p->callid);
}
}
}
}
/* Check if we have a global disallow setting on subscriptions.
if so, we don't have to check peer/user settings after auth, which saves a lot of processing
@ -14338,13 +14338,13 @@ static int sipsock_read(int *id, int fd, short events, void *ignore)
/* Process request, with netlock held, and with usual deadlock avoidance */
for (lockretry = 100; lockretry > 0; lockretry--) {
ast_mutex_lock(&netlock);
ast_mutex_lock(&netlock);
/* Find the active SIP dialog or create a new one */
p = find_call(&req, &sin, req.method); /* returns p locked */
if (p == NULL) {
if (option_debug)
ast_log(LOG_DEBUG, "Invalid SIP message - rejected , no callid, len %d\n", req.len);
/* Find the active SIP dialog or create a new one */
p = find_call(&req, &sin, req.method); /* returns p locked */
if (p == NULL) {
if (option_debug)
ast_log(LOG_DEBUG, "Invalid SIP message - rejected , no callid, len %d\n", req.len);
ast_mutex_unlock(&netlock);
return 1;
}
@ -14352,36 +14352,36 @@ static int sipsock_read(int *id, int fd, short events, void *ignore)
/* becaues this is deadlock-prone, we need to try and unlock if failed */
if (!p->owner || !ast_channel_trylock(p->owner))
break; /* locking succeeded */
if (option_debug)
ast_log(LOG_DEBUG, "Failed to grab owner channel lock, trying again. (SIP call %s)\n", p->callid);
ast_mutex_unlock(&p->lock);
ast_mutex_unlock(&netlock);
/* Sleep for a very short amount of time */
usleep(1);
}
p->recv = sin;
if (option_debug)
ast_log(LOG_DEBUG, "Failed to grab owner channel lock, trying again. (SIP call %s)\n", p->callid);
ast_mutex_unlock(&p->lock);
ast_mutex_unlock(&netlock);
/* Sleep for a very short amount of time */
usleep(1);
}
p->recv = sin;
if (recordhistory) /* This is a request or response, note what it was for */
append_history(p, "Rx", "%s / %s / %s", req.data, get_header(&req, "CSeq"), req.rlPart2);
if (recordhistory) /* This is a request or response, note what it was for */
append_history(p, "Rx", "%s / %s / %s", req.data, get_header(&req, "CSeq"), req.rlPart2);
if (!lockretry) {
if (!lockretry) {
ast_log(LOG_ERROR, "We could NOT get the channel lock for %s! \n", S_OR(p->owner->name, "- no channel name ??? - "));
ast_log(LOG_ERROR, "SIP transaction failed: %s \n", p->callid);
transmit_response(p, "503 Server error", &req); /* We must respond according to RFC 3261 sec 12.2 */
/* XXX We could add retry-after to make sure they come back */
append_history(p, "LockFail", "Owner lock failed, transaction failed.");
return 1;
}
nounlock = 0;
if (handle_request(p, &req, &sin, &recount, &nounlock) == -1) {
/* Request failed */
if (option_debug)
ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
}
ast_log(LOG_ERROR, "SIP transaction failed: %s \n", p->callid);
transmit_response(p, "503 Server error", &req); /* We must respond according to RFC 3261 sec 12.2 */
/* XXX We could add retry-after to make sure they come back */
append_history(p, "LockFail", "Owner lock failed, transaction failed.");
return 1;
}
nounlock = 0;
if (handle_request(p, &req, &sin, &recount, &nounlock) == -1) {
/* Request failed */
if (option_debug)
ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
}
if (p->owner && !nounlock)
ast_channel_unlock(p->owner);
ast_mutex_unlock(&p->lock);
if (p->owner && !nounlock)
ast_channel_unlock(p->owner);
ast_mutex_unlock(&p->lock);
ast_mutex_unlock(&netlock);
if (recount)
ast_update_use_count();
@ -15097,7 +15097,7 @@ static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char
return authlist;
if (option_debug)
ast_log(LOG_DEBUG, "Auth config :: %s\n", configuration);
ast_log(LOG_DEBUG, "Auth config :: %s\n", configuration);
ast_copy_string(authcopy, configuration, sizeof(authcopy));
stringp = authcopy;

Write Preview
Loading…
Cancel
Save