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Diffstat (limited to 'channels/chan_sip.c')
-rw-r--r--channels/chan_sip.c185
1 files changed, 91 insertions, 94 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d9304b66e..f9cd26a23 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3136,9 +3136,9 @@ static int __sip_xmit(struct sip_pvt *p, struct ast_str *data, int len)
if (res == -1) {
switch (errno) {
- case EBADF: /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
- case EHOSTUNREACH: /* Host can't be reached */
- case ENETDOWN: /* Interface down */
+ case EBADF: /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
+ case EHOSTUNREACH: /* Host can't be reached */
+ case ENETDOWN: /* Interface down */
case ENETUNREACH: /* Network failure */
case ECONNREFUSED: /* ICMP port unreachable */
res = XMIT_ERROR; /* Don't bother with trying to transmit again */
@@ -4182,8 +4182,8 @@ static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data
if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
- }
+ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
+ }
break;
default:
ast_log(LOG_WARNING, "Don't know how to send URI when state is %d!\n", chan->_state);
@@ -4218,7 +4218,7 @@ static int sip_sendtext(struct ast_channel *ast, const char *text)
if (debug)
ast_verbose("Sending text %s on %s\n", text, ast->name);
transmit_message_with_text(dialog, text);
- return 0;
+ return 0;
}
/*! \brief Update peer object in realtime storage
@@ -4348,18 +4348,17 @@ static void sip_destroy_peer(struct sip_peer *peer)
dialog_unlink_all(peer->call, TRUE, TRUE);
peer->call = dialog_unref(peer->call, "peer->call is being unset");
}
-
if (peer->mwipvt) { /* We have an active subscription, delete it */
dialog_unlink_all(peer->mwipvt, TRUE, TRUE);
peer->mwipvt = dialog_unref(peer->mwipvt, "unreffing peer->mwipvt");
}
-
+
if (peer->chanvars) {
ast_variables_destroy(peer->chanvars);
peer->chanvars = NULL;
}
-
+
register_peer_exten(peer, FALSE);
ast_free_ha(peer->ha);
ast_free_ha(peer->directmediaha);
@@ -5296,9 +5295,9 @@ static void sip_registry_destroy(struct sip_registry *reg)
reg->call = dialog_unref(reg->call, "unref reg->call");
/* reg->call = sip_destroy(reg->call); */
}
- AST_SCHED_DEL(sched, reg->expire);
+ AST_SCHED_DEL(sched, reg->expire);
AST_SCHED_DEL(sched, reg->timeout);
-
+
ast_string_field_free_memory(reg);
ast_atomic_fetchadd_int(&regobjs, -1);
ast_dnsmgr_release(reg->dnsmgr);
@@ -5806,7 +5805,7 @@ const char *hangup_cause2sip(int cause)
return "488 Not Acceptable Here";
case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
return "500 Network error";
-
+
case AST_CAUSE_NOTDEFINED:
default:
ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause);
@@ -6055,7 +6054,7 @@ static void try_suggested_sip_codec(struct sip_pvt *p)
ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
} else
ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
- return;
+ return;
}
/*! \brief sip_answer: Answer SIP call , send 200 OK on Invite
@@ -6067,7 +6066,7 @@ static int sip_answer(struct ast_channel *ast)
sip_pvt_lock(p);
if (ast->_state != AST_STATE_UP) {
- try_suggested_sip_codec(p);
+ try_suggested_sip_codec(p);
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
@@ -6613,14 +6612,14 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
needvideo = 1;
else if (!ast_format_cap_is_empty(i->prefcaps))
needvideo = ast_format_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_VIDEO); /* Outbound call */
- else
+ else
needvideo = ast_format_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_VIDEO); /* Inbound call */
}
if (i->trtp) {
if (!ast_format_cap_is_empty(i->prefcaps))
needtext = ast_format_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_TEXT); /* Outbound call */
- else
+ else
needtext = ast_format_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_TEXT); /* Inbound call */
}
@@ -7010,7 +7009,7 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
}
}
}
-
+
return f;
}
@@ -7292,7 +7291,7 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr,
/* Add to active dialog list */
ao2_t_link(dialogs, p, "link pvt into dialogs table");
-
+
ast_debug(1, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : p->callid, sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP");
return p;
}
@@ -8426,7 +8425,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
ast_debug(3, "Processing session-level SDP %c=%s... %s\n", type, value, (processed == TRUE)? "OK." : "UNSUPPORTED.");
- }
+ }
@@ -8888,7 +8887,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_set_read_format(p->owner, &p->owner->readformat);
ast_set_write_format(p->owner, &p->owner->writeformat);
}
-
+
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && (!ast_sockaddr_isnull(sa) || !ast_sockaddr_isnull(vsa) || !ast_sockaddr_isnull(tsa) || !ast_sockaddr_isnull(isa)) && (!sendonly || sendonly == -1)) {
ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
/* Activate a re-invite */
@@ -9821,14 +9820,14 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in
int is_outbound = ast_test_flag(&p->flags[0], SIP_OUTGOING); /* Session direction */
memset(req, 0, sizeof(struct sip_request));
-
+
snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text);
-
+
if (!seqno) {
p->ocseq++;
seqno = p->ocseq;
}
-
+
/* A CANCEL must have the same branch as the INVITE that it is canceling. */
if (sipmethod == SIP_CANCEL) {
p->branch = p->invite_branch;
@@ -9848,7 +9847,7 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in
if (sipdebug)
ast_debug(1, "Strict routing enforced for session %s\n", p->callid);
}
-
+
if (sipmethod == SIP_CANCEL)
c = REQ_OFFSET_TO_STR(&p->initreq, rlPart2); /* Use original URI */
else if (sipmethod == SIP_ACK) {
@@ -9856,8 +9855,8 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in
(we only have the contacturi on INVITEs) */
if (!ast_strlen_zero(p->okcontacturi))
c = is_strict ? p->route->hop : p->okcontacturi;
- else
- c = REQ_OFFSET_TO_STR(&p->initreq, rlPart2);
+ else
+ c = REQ_OFFSET_TO_STR(&p->initreq, rlPart2);
} else if (!ast_strlen_zero(p->okcontacturi))
c = is_strict ? p->route->hop : p->okcontacturi; /* Use for BYE or REINVITE */
else if (!ast_strlen_zero(p->uri))
@@ -11124,7 +11123,7 @@ static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct s
{
struct sip_request resp;
int seqno;
-
+
if (sscanf(get_header(req, "CSeq"), "%30d ", &seqno) != 1) {
ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq"));
return -1;
@@ -11723,17 +11722,17 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init,
/* Strip of the starting " (if it's there) */
if (*headdup == '"') {
- headdup++;
+ headdup++;
}
if ((content = strchr(headdup, ':'))) {
*content++ = '\0';
content = ast_skip_blanks(content); /* Skip white space */
/* Strip the ending " (if it's there) */
- end = content + strlen(content) -1;
+ end = content + strlen(content) -1;
if (*end == '"') {
*end = '\0';
}
-
+
add_header(&req, headdup, content);
if (sipdebug) {
ast_debug(1, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content);
@@ -12725,7 +12724,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, const char *
if (!ast_strlen_zero(global_useragent))
add_header(&req, "User-Agent", global_useragent);
- if (auth) { /* Add auth header */
+ if (auth) { /* Add auth header */
add_header(&req, authheader, auth);
} else if (!ast_strlen_zero(r->nonce)) {
char digest[1024];
@@ -12804,7 +12803,7 @@ static int sip_notify_allocate(struct sip_pvt *p)
static int transmit_refer(struct sip_pvt *p, const char *dest)
{
struct sip_request req = {
- .headers = 0,
+ .headers = 0,
};
char from[256];
const char *of;
@@ -13533,7 +13532,7 @@ static void build_route(struct sip_pvt *p, struct sip_request *req, int backward
/* We only want to create the route set the first time this is called */
p->route_persistent = 1;
-
+
/* Build a tailq, then assign it to p->route when done.
* If backwards, we add entries from the head so they end up
* in reverse order. However, we do need to maintain a correct
@@ -14770,7 +14769,7 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t
}
/* Search dialogs and find the match */
-
+
sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find of dialog in dialogs table");
if (sip_pvt_ptr) {
/* Go ahead and lock it (and its owner) before returning */
@@ -14823,7 +14822,7 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t
return NULL;
}
}
-
+
if (totag)
ast_debug(4, "Matched %s call - their tag is %s Our tag is %s\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING",
@@ -14948,7 +14947,7 @@ static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoi
}
ast_copy_string(referdata->replaces_callid_totag, ptr, sizeof(referdata->replaces_callid_totag));
}
-
+
if (from) {
ptr = from + 9;
if ((to = strchr(ptr, '&'))) {
@@ -14959,14 +14958,14 @@ static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoi
}
ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag));
}
-
+
if (!sip_cfg.pedanticsipchecking) {
ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid );
} else {
ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" );
}
}
-
+
if ((ptr = strchr(refer_to, '@'))) { /* Separate domain */
char *urioption = NULL, *domain;
int bracket = 0;
@@ -16274,7 +16273,7 @@ static int dialog_needdestroy(void *dialogobj, void *arg, int flags)
sip_pvt_unlock(dialog);
return 0;
}
-
+
if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
@@ -18326,16 +18325,16 @@ static int do_register_auth(struct sip_pvt *p, struct sip_request *req, enum sip
memset(digest, 0, sizeof(digest));
if (reply_digest(p, req, header, SIP_REGISTER, digest, sizeof(digest))) {
/* There's nothing to use for authentication */
- /* No digest challenge in request */
- if (sip_debug_test_pvt(p) && p->registry)
- ast_verbose("No authentication challenge, sending blank registration to domain/host name %s\n", p->registry->hostname);
- /* No old challenge */
+ /* No digest challenge in request */
+ if (sip_debug_test_pvt(p) && p->registry)
+ ast_verbose("No authentication challenge, sending blank registration to domain/host name %s\n", p->registry->hostname);
+ /* No old challenge */
return -1;
}
if (p->do_history)
append_history(p, "RegistryAuth", "Try: %d", p->authtries);
- if (sip_debug_test_pvt(p) && p->registry)
- ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
+ if (sip_debug_test_pvt(p) && p->registry)
+ ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
return transmit_register(p->registry, SIP_REGISTER, digest, respheader);
}
@@ -18465,28 +18464,28 @@ static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int d
snprintf(cnonce, sizeof(cnonce), "%08lx", ast_random());
- /* Check if we have separate auth credentials */
- if(!(auth = find_realm_authentication(p->peerauth, p->realm))) /* Start with peer list */
- auth = find_realm_authentication(authl, p->realm); /* If not, global list */
+ /* Check if we have separate auth credentials */
+ if(!(auth = find_realm_authentication(p->peerauth, p->realm))) /* Start with peer list */
+ auth = find_realm_authentication(authl, p->realm); /* If not, global list */
- if (auth) {
+ if (auth) {
ast_debug(3, "use realm [%s] from peer [%s][%s]\n", auth->username, p->peername, p->username);
- username = auth->username;
- secret = auth->secret;
- md5secret = auth->md5secret;
+ username = auth->username;
+ secret = auth->secret;
+ md5secret = auth->md5secret;
if (sipdebug)
- ast_debug(1, "Using realm %s authentication for call %s\n", p->realm, p->callid);
- } else {
- /* No authentication, use peer or register= config */
- username = p->authname;
- secret = p->peersecret;
- md5secret = p->peermd5secret;
- }
+ ast_debug(1, "Using realm %s authentication for call %s\n", p->realm, p->callid);
+ } else {
+ /* No authentication, use peer or register= config */
+ username = p->authname;
+ secret = p->peersecret;
+ md5secret = p->peermd5secret;
+ }
if (ast_strlen_zero(username)) /* We have no authentication */
return -1;
- /* Calculate SIP digest response */
- snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
+ /* Calculate SIP digest response */
+ snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
snprintf(a2, sizeof(a2), "%s:%s", sip_methods[method].text, uri);
if (!ast_strlen_zero(md5secret))
ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
@@ -18970,7 +18969,7 @@ static void check_pendings(struct sip_pvt *p)
/* Perhaps there is an SD change INVITE outstanding */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
}
- ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
+ ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
/* if we can't REINVITE, hold it for later */
@@ -18980,7 +18979,7 @@ static void check_pendings(struct sip_pvt *p)
ast_debug(2, "Sending pending reinvite on '%s'\n", p->callid);
/* Didn't get to reinvite yet, so do it now */
transmit_reinvite_with_sdp(p, (p->t38.state == T38_LOCAL_REINVITE ? TRUE : FALSE), FALSE);
- ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
+ ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
}
@@ -19834,7 +19833,7 @@ static int handle_response_register(struct sip_pvt *p, int resp, const char *res
pvt_set_needdestroy(p, "received erroneous 200 response");
return 0;
}
-
+
r->regstate = REG_STATE_REGISTERED;
r->regtime = ast_tvnow(); /* Reset time of last successful registration */
manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate));
@@ -19850,7 +19849,7 @@ static int handle_response_register(struct sip_pvt *p, int resp, const char *res
/* Let this one hang around until we have all the responses */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
/* p->needdestroy = 1; */
-
+
/* set us up for re-registering
* figure out how long we got registered for
* according to section 6.13 of RFC, contact headers override
@@ -20408,7 +20407,7 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
break;
default: /* Errors without handlers */
if ((resp >= 100) && (resp < 200)) {
- if (sipmethod == SIP_INVITE) { /* re-invite */
+ if (sipmethod == SIP_INVITE) { /* re-invite */
if (!req->ignore && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
}
@@ -20475,7 +20474,6 @@ static void *sip_park_thread(void *stuff)
}
res = ast_park_call(transferee, transferer, 0, d->parkexten, &ext);
-
#ifdef WHEN_WE_KNOW_THAT_THE_CLIENT_SUPPORTS_MESSAGE
if (!res) {
@@ -20622,14 +20620,14 @@ static void ast_quiet_chan(struct ast_channel *chan)
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target)
{
int res = 0;
- struct ast_channel *peera = NULL,
+ struct ast_channel *peera = NULL,
*peerb = NULL,
*peerc = NULL,
*peerd = NULL;
/* We will try to connect the transferee with the target and hangup
- all channels to the transferer */
+ all channels to the transferer */
ast_debug(4, "Sip transfer:--------------------\n");
if (transferer->chan1)
ast_debug(4, "-- Transferer to PBX channel: %s State %s\n", transferer->chan1->name, ast_state2str(transferer->chan1->_state));
@@ -20664,7 +20662,7 @@ static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target
if (peera && peerb && peerc && (peerb != peerc)) {
ast_quiet_chan(peera); /* Stop generators */
- ast_quiet_chan(peerb);
+ ast_quiet_chan(peerb);
ast_quiet_chan(peerc);
if (peerd)
ast_quiet_chan(peerd);
@@ -20783,7 +20781,7 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str
*sep++ = '\0';
eventid = sep;
}
-
+
if (sipdebug)
ast_debug(2, "Got NOTIFY Event: %s\n", event);
@@ -20803,7 +20801,7 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str
We are getting notifications on a call that we transfered
We should hangup when we are getting a 200 OK in a sipfrag
Check if we have an owner of this event */
-
+
/* Check the content type */
if (strncasecmp(get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) {
/* We need a sipfrag */
@@ -22427,7 +22425,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
transmit_response(p, "202 Accepted", req);
append_history(p, "Xfer", "Refer failed. Bad extension.");
transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE);
- ast_clear_flag(&p->flags[0], SIP_GOTREFER);
+ ast_clear_flag(&p->flags[0], SIP_GOTREFER);
if (req->debug)
ast_debug(1, "SIP transfer to bad extension: %s\n", p->refer->refer_to);
break;
@@ -22665,7 +22663,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
if (p->owner)
p->owner->hangupcause = AST_CAUSE_NORMAL_CLEARING;
append_history(p, "Xfer", "Refer succeeded.");
- ast_clear_flag(&p->flags[0], SIP_GOTREFER);
+ ast_clear_flag(&p->flags[0], SIP_GOTREFER);
/* Do not hangup call, the other side do that when we say 200 OK */
/* We could possibly implement a timer here, auto congestion */
res = 0;
@@ -22676,7 +22674,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
/* Failure of some kind */
p->refer->status = REFER_FAILED;
transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable", TRUE);
- ast_clear_flag(&p->flags[0], SIP_GOTREFER);
+ ast_clear_flag(&p->flags[0], SIP_GOTREFER);
res = -1;
}
@@ -23481,12 +23479,12 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
int resubscribe = (p->subscribed != NONE) && !req->ignore;
char *temp, *event;
- if (p->initreq.headers) {
+ if (p->initreq.headers) {
/* We already have a dialog */
if (p->initreq.method != SIP_SUBSCRIBE) {
/* This is a SUBSCRIBE within another SIP dialog, which we do not support */
/* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */
- transmit_response(p, "403 Forbidden (within dialog)", req);
+ transmit_response(p, "403 Forbidden (within dialog)", req);
/* Do not destroy session, since we will break the call if we do */
ast_debug(1, "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;
@@ -23502,7 +23500,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
if so, we don't have to check peer settings after auth, which saves a lot of processing
*/
if (!sip_cfg.allowsubscribe) {
- transmit_response(p, "403 Forbidden (policy)", req);
+ transmit_response(p, "403 Forbidden (policy)", req);
pvt_set_needdestroy(p, "forbidden");
return 0;
}
@@ -23543,7 +23541,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
if ( (strchr(eventheader, ';'))) {
event = ast_strdupa(eventheader); /* Since eventheader is a const, we can't change it */
- temp = strchr(event, ';');
+ temp = strchr(event, ';');
*temp = '\0'; /* Remove any options for now */
/* We might need to use them later :-) */
} else
@@ -23695,7 +23693,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
while (!found_supported && !ast_strlen_zero(acceptheader)) {
found_supported = strcmp(acceptheader, "application/simple-message-summary") ? 0 : 1;
if (!found_supported && (option_debug > 2)) {
- ast_log(LOG_DEBUG, "Received SIP mailbox subscription for unknown format: %s\n", acceptheader);
+ ast_debug(1, "Received SIP mailbox subscription for unknown format: %s\n", acceptheader);
}
acceptheader = __get_header(req, "Accept", &start);
}
@@ -24006,8 +24004,7 @@ static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct as
/* New SIP request coming in
(could be new request in existing SIP dialog as well...)
- */
-
+ */
p->method = req->method; /* Find out which SIP method they are using */
ast_debug(4, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
@@ -24402,7 +24399,7 @@ static int handle_request_do(struct sip_request *req, struct ast_sockaddr *addr)
/* Request failed */
ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
}
-
+
if (recount)
ast_update_use_count();
@@ -25352,7 +25349,7 @@ static int sip_devicestate(void *data)
if ((p = find_peer(host, NULL, FALSE, FINDALLDEVICES, TRUE, 0))) {
if (!(ast_sockaddr_isnull(&p->addr) && ast_sockaddr_isnull(&p->defaddr))) {
/* we have an address for the peer */
-
+
/* Check status in this order
- Hold
- Ringing
@@ -25565,7 +25562,7 @@ static struct ast_channel *sip_request_call(const char *type, struct ast_format_
ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
build_callid_pvt(p);
ao2_t_link(dialogs, p, "Linking in under new name");
-
+
/* We have an extension to call, don't use the full contact here */
/* This to enable dialing registered peers with extension dialling,
like SIP/peername/extension
@@ -25850,7 +25847,7 @@ static int add_sip_domain(const char *domain, const enum domain_mode mode, const
AST_LIST_INSERT_TAIL(&domain_list, d, list);
AST_LIST_UNLOCK(&domain_list);
- if (sipdebug)
+ if (sipdebug)
ast_debug(1, "Added local SIP domain '%s'\n", domain);
return 1;
@@ -25869,7 +25866,7 @@ static int check_sip_domain(const char *domain, char *context, size_t len)
if (len && !ast_strlen_zero(d->context))
ast_copy_string(context, d->context, len);
-
+
result = 1;
break;
}
@@ -27093,7 +27090,7 @@ static int reload_config(enum channelreloadreason reason)
sip_cfg.allowguest = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "realm")) {
ast_copy_string(sip_cfg.realm, v->value, sizeof(sip_cfg.realm));
- } else if (!strcasecmp(v->name, "domainsasrealm")) {
+ } else if (!strcasecmp(v->name, "domainsasrealm")) {
sip_cfg.domainsasrealm = ast_true(v->value);
} else if (!strcasecmp(v->name, "useragent")) {
ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
@@ -27685,7 +27682,7 @@ static int reload_config(enum channelreloadreason reason)
STANDARD_TLS_PORT);
}
ast_tcptls_server_start(&sip_tls_desc);
- if (default_tls_cfg.enabled && sip_tls_desc.accept_fd == -1) {
+ if (default_tls_cfg.enabled && sip_tls_desc.accept_fd == -1) {
ast_log(LOG_ERROR, "TLS Server start failed. Not listening on TLS socket.\n");
sip_tls_desc.tls_cfg = NULL;
}
@@ -27911,7 +27908,7 @@ static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
{
struct sip_pvt *p;
-
+
p = chan->tech_pvt;
if (!p) {
return -1;
@@ -28231,7 +28228,7 @@ static int sip_removeheader(struct ast_channel *chan, const char *data)
{
struct ast_var_t *newvariable;
struct varshead *headp;
- int removeall = 0;
+ int removeall = 0;
char *inbuf = (char *) data;
if (ast_strlen_zero(inbuf)) {
@@ -28450,7 +28447,7 @@ static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struc
static int sip_do_reload(enum channelreloadreason reason)
{
time_t start_poke, end_poke;
-
+
reload_config(reason);
ast_sched_dump(sched);
@@ -28469,7 +28466,7 @@ static int sip_do_reload(enum channelreloadreason reason)
sip_send_all_mwi_subscriptions();
end_poke = time(0);
-
+
ast_debug(4, "do_reload finished. peer poke/prune reg contact time = %d sec.\n", (int)(end_poke-start_poke));
ast_debug(4, "--------------- SIP reload done\n");
@@ -28480,7 +28477,7 @@ static int sip_do_reload(enum channelreloadreason reason)
/*! \brief Force reload of module from cli */
static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
-
+
switch (cmd) {
case CLI_INIT:
e->command = "sip reload";
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-29 13:27:18 +0000