{
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));
(!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 */
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);
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;
}
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;
}
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");
{
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);
}
char *hostname=NULL, *secret=NULL, *authuser=NULL;
char *porta=NULL;
char *contact=NULL;
-
+
if (!value)
return -1;
ast_copy_string(username, value, sizeof(username));
*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");
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) {
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";
}
}
}
- }
+ }
if (sdp) {
if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
ast_udptl_offered_from_local(p->udptl, 1);
}
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, '@')) {
/* 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;
}
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;
}
}
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 {
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;
}
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;
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;
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);
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) {
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;
}
} 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");
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);
}
}
/* 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 */
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
/* 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;
}
/* 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();
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;
projects / thirdparty / asterisk.git / commitdiff
