[asterisk-commits] trunk r12494 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Mar 10 05:13:07 MST 2006
Author: oej
Date: Fri Mar 10 06:13:05 2006
New Revision: 12494
URL: http://svn.digium.com/view/asterisk?rev=12494&view=rev
Log:
Implement enum for retransmit options to various functions.
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=12494&r1=12493&r2=12494&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Mar 10 06:13:05 2006
@@ -103,6 +103,7 @@
#ifndef TRUE
#define TRUE 1
#endif
+
#define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
#ifndef IPTOS_MINCOST
@@ -170,6 +171,13 @@
submitting a patch. If these two lists do not match properly
bad things will happen.
*/
+
+enum xmittype {
+ XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
+ If it fails, it's critical and will cause a teardown of the session */
+ XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
+ XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
+};
enum subscriptiontype {
NONE = 0,
@@ -909,11 +917,11 @@
/*---------------------------- Forward declarations of functions in chan_sip.c */
static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req);
-static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
+static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable);
static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported);
-static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *rand, int reliable, const char *header, int stale);
-static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch);
-static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch);
+static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
+static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
+static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sendsdp, int init);
static int transmit_reinvite_with_sdp(struct sip_pvt *p);
static int transmit_info_with_digit(struct sip_pvt *p, char digit);
@@ -947,7 +955,7 @@
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);
+ char *uri, enum xmittype 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);
@@ -961,7 +969,7 @@
static void parse_request(struct sip_request *req);
static char *get_header(struct sip_request *req, const char *name);
static void copy_request(struct sip_request *dst,struct sip_request *src);
-static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal);
+static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req);
static int transmit_register(struct sip_registry *r, int sipmethod, char *auth, char *authheader);
static int sip_poke_peer(struct sip_peer *peer);
static int __sip_do_register(struct sip_registry *r);
@@ -1344,6 +1352,7 @@
pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
if (fatal)
ast_set_flag(pkt, FLAG_FATAL);
+
if (pkt->timer_t1)
siptimer_a = pkt->timer_t1 * 2;
@@ -1527,7 +1536,7 @@
}
/*! \brief Transmit response on SIP request*/
-static int send_response(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno)
+static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
int res;
@@ -1544,7 +1553,7 @@
append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
}
res = (reliable) ?
- __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1), req->method) :
+ __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
__sip_xmit(p, req->data, req->len);
if (res > 0)
return 0;
@@ -1552,7 +1561,7 @@
}
/*! \brief Send SIP Request to the other part of the dialogue */
-static int send_request(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno)
+static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
int res;
@@ -2569,13 +2578,13 @@
if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
if (needcancel) { /* Outgoing call, not up */
if (ast_test_flag(p, SIP_OUTGOING)) {
- transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0);
+ transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, 0);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
ast_clear_flag(&locflags, SIP_NEEDDESTROY);
- sip_scheddestroy(p, 15000);
/* stop retransmitting an INVITE that has not received a response */
__sip_pretend_ack(p);
+ sip_scheddestroy(p, 32000);
if ( p->initid != -1 ) {
/* channel still up - reverse dec of inUse counter
only if the channel is not auto-congested */
@@ -2584,14 +2593,14 @@
} else { /* Incoming call, not up */
char *res;
if (ast->hangupcause && ((res = hangup_cause2sip(ast->hangupcause)))) {
- transmit_response_reliable(p, res, &p->initreq, 1);
+ transmit_response_reliable(p, res, &p->initreq);
} else
- transmit_response_reliable(p, "603 Declined", &p->initreq, 1);
+ transmit_response_reliable(p, "603 Declined", &p->initreq);
}
} else { /* Call is in UP state, send BYE */
if (!p->pendinginvite) {
/* Send a hangup */
- transmit_request_with_auth(p, SIP_BYE, 0, 1, 1);
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while we have "INVITE" outstanding. */
@@ -2645,7 +2654,7 @@
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
- res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 1);
+ res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_RELIABLE);
}
ast_mutex_unlock(&p->lock);
return res;
@@ -2669,7 +2678,7 @@
if (p->rtp) {
/* If channel is not up, activate early media session */
if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
- transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
+ transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(p, SIP_PROGRESS_SENT);
}
time(&p->lastrtptx);
@@ -2684,7 +2693,7 @@
if (p->vrtp) {
/* Activate video early media */
if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
- transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
+ transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(p, SIP_PROGRESS_SENT);
}
time(&p->lastrtptx);
@@ -2813,7 +2822,7 @@
break;
case AST_CONTROL_PROGRESS:
if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
- transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
+ transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(p, SIP_PROGRESS_SENT);
break;
}
@@ -4284,7 +4293,7 @@
}
/*! \brief Base transmit response function */
-static int __transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req, int reliable)
+static int __transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable)
{
struct sip_request resp;
int seqno = 0;
@@ -4307,7 +4316,7 @@
/*! \brief Transmit response, no retransmits */
static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req)
{
- return __transmit_response(p, msg, req, 0);
+ return __transmit_response(p, msg, req, XMIT_UNRELIABLE);
}
/*! \brief Transmit response, no retransmits */
@@ -4317,13 +4326,15 @@
respprep(&resp, p, msg, req);
append_date(&resp);
add_header(&resp, "Unsupported", unsupported);
- return send_response(p, &resp, 0, 0);
-}
-
-/*! \brief Transmit response, Make sure you get a reply */
-static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal)
-{
- return __transmit_response(p, msg, req, fatal ? 2 : 1);
+ return send_response(p, &resp, XMIT_UNRELIABLE, 0);
+}
+
+/*! \brief Transmit response, Make sure you get an ACK
+ This is only used for responses to INVITEs, where we need to make sure we get an ACK
+*/
+static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req)
+{
+ return __transmit_response(p, msg, req, XMIT_CRITICAL);
}
/*! \brief Append date to SIP message */
@@ -4347,11 +4358,11 @@
append_date(&resp);
add_header_contentLength(&resp, 0);
add_blank_header(&resp);
- return send_response(p, &resp, 0, 0);
+ return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
/*! \brief Append Accept header, content length before transmitting response */
-static int transmit_response_with_allow(struct sip_pvt *p, char *msg, struct sip_request *req, int reliable)
+static int transmit_response_with_allow(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable)
{
struct sip_request resp;
respprep(&resp, p, msg, req);
@@ -4362,7 +4373,7 @@
}
/*! \brief Respond with authorization request */
-static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *randdata, int reliable, const char *header, int stale)
+static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale)
{
struct sip_request resp;
char tmp[256];
@@ -4671,7 +4682,7 @@
}
/*! \brief Used for 200 OK and 183 early media */
-static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
+static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable)
{
struct sip_request resp;
int seqno;
@@ -4686,7 +4697,7 @@
} else {
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
}
- return send_response(p, &resp, retrans, seqno);
+ return send_response(p, &resp, reliable, seqno);
}
/*! \brief Parse first line of incoming SIP request */
@@ -5718,7 +5729,7 @@
}
/*! \brief Transmit generic SIP request */
-static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch)
+static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
{
struct sip_request resp;
reqprep(&resp, p, sipmethod, seqno, newbranch);
@@ -5728,7 +5739,7 @@
}
/*! \brief Transmit SIP request, auth added */
-static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch)
+static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
{
struct sip_request resp;
@@ -6244,7 +6255,7 @@
*/
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)
+ char *uri, enum xmittype reliable, int ignore)
{
const char *response = "407 Proxy Authentication Required";
const char *reqheader = "Proxy-Authorization";
@@ -6522,7 +6533,7 @@
} else {
ast_copy_flags(p, peer, SIP_NAT);
transmit_response(p, "100 Trying", req);
- if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, 0, ignore))) {
+ if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, XMIT_UNRELIABLE, ignore))) {
sip_cancel_destroy(p);
switch (parse_register_contact(p, peer, req)) {
case PARSE_REGISTER_FAILED:
@@ -7066,7 +7077,7 @@
\return 0 on success, -1 on failure, and 1 on challenge sent
-2 on authentication error from chedck_auth()
*/
-static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore, char *mailbox, int mailboxlen)
+static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin, int ignore, char *mailbox, int mailboxlen)
{
struct sip_user *user = NULL;
struct sip_peer *peer;
@@ -7359,7 +7370,7 @@
/*! \brief Find user
If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced
*/
-static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore)
+static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin, int ignore)
{
return check_user_full(p, req, sipmethod, uri, reliable, sin, ignore, NULL, 0);
}
@@ -9560,7 +9571,7 @@
{
/* Go ahead and send bye at this point */
if (ast_test_flag(p, SIP_PENDINGBYE)) {
- transmit_request_with_auth(p, SIP_BYE, 0, 1, 1);
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
ast_set_flag(p, SIP_NEEDDESTROY);
ast_clear_flag(p, SIP_NEEDREINVITE);
} else if (ast_test_flag(p, SIP_NEEDREINVITE)) {
@@ -9653,13 +9664,13 @@
ast_set_flag(p, SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a non-200 ACK only */
- transmit_request(p, SIP_ACK, seqno, 0, 1);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 1);
check_pendings(p);
break;
case 407: /* Proxy authentication */
case 401: /* Www auth */
/* First we ACK */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
if (p->options)
p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
@@ -9679,7 +9690,7 @@
break;
case 403: /* Forbidden */
/* First we ACK */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for INVITE to '%s'\n", get_header(&p->initreq, "From"));
if (!ignore && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
@@ -9687,7 +9698,7 @@
ast_set_flag(p, SIP_ALREADYGONE);
break;
case 404: /* Not found */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
if (p->owner && !ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(p, SIP_ALREADYGONE);
@@ -9695,7 +9706,7 @@
case 481: /* Call leg does not exist */
/* Could be REFER or INVITE */
ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
break;
case 491: /* Pending */
/* we have to wait a while, then retransmit */
@@ -9869,7 +9880,7 @@
if (peer->pokeexpire > -1)
ast_sched_del(sched, peer->pokeexpire);
if (sipmethod == SIP_INVITE) /* Does this really happen? */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
ast_set_flag(p, SIP_NEEDDESTROY);
/* Try again eventually */
@@ -10019,6 +10030,7 @@
break;
default:
if ((resp >= 300) && (resp < 700)) {
+ /* Fatal response */
if ((option_verbose > 2) && (resp != 487))
ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr));
ast_set_flag(p, SIP_ALREADYGONE);
@@ -10044,7 +10056,7 @@
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_BUSY);
break;
- case 487:
+ case 487: /* Response on INVITE that has been CANCELled */
/* channel now destroyed - dec the inUse counter */
update_call_counter(p, DEC_CALL_LIMIT);
break;
@@ -10075,7 +10087,7 @@
}
/* ACK on invite */
if (sipmethod == SIP_INVITE)
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
ast_set_flag(p, SIP_ALREADYGONE);
if (!p->owner)
ast_set_flag(p, SIP_NEEDDESTROY);
@@ -10442,12 +10454,12 @@
if (!p->lastinvite && !ignore && !p->owner) {
/* Handle authentication if this is our first invite */
- res = check_user(p, req, SIP_INVITE, e, 1, sin, ignore);
+ res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin, ignore);
if (res > 0) /* We have challenged the user for auth */
return 0;
if (res < 0) { /* Something failed in authentication */
ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
- transmit_response_reliable(p, "403 Forbidden", req, 1);
+ transmit_response_reliable(p, "403 Forbidden", req);
ast_set_flag(p, SIP_NEEDDESTROY);
ast_string_field_free(p, theirtag);
return 0;
@@ -10457,7 +10469,7 @@
if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
if (process_sdp(p, req)) {
/* Unacceptable codecs */
- transmit_response_reliable(p, "488 Not acceptable here", req, 1);
+ transmit_response_reliable(p, "488 Not acceptable here", req);
ast_set_flag(p, SIP_NEEDDESTROY);
return -1;
}
@@ -10482,7 +10494,7 @@
if (res) {
if (res < 0) {
ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username);
- transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req, 1);
+ transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
ast_set_flag(p, SIP_NEEDDESTROY);
}
return 0;
@@ -10494,10 +10506,10 @@
if (gotdest) {
if (gotdest < 0) {
- transmit_response_reliable(p, "404 Not Found", req, 1);
+ transmit_response_reliable(p, "404 Not Found", req);
update_call_counter(p, DEC_CALL_LIMIT);
} else {
- transmit_response_reliable(p, "484 Address Incomplete", req, 1);
+ transmit_response_reliable(p, "484 Address Incomplete", req);
update_call_counter(p, DEC_CALL_LIMIT);
}
ast_set_flag(p, SIP_NEEDDESTROY);
@@ -10548,14 +10560,14 @@
if (ignore)
transmit_response(p, "503 Unavailable", req);
else
- transmit_response_reliable(p, "503 Unavailable", req, 1);
+ transmit_response_reliable(p, "503 Unavailable", req);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
if (ignore)
transmit_response(p, "480 Temporarily Unavailable", req);
else
- transmit_response_reliable(p, "480 Temporarily Unavailable", req, 1);
+ transmit_response_reliable(p, "480 Temporarily Unavailable", req);
break;
case AST_PBX_SUCCESS:
/* nothing to do */
@@ -10578,7 +10590,7 @@
if (ignore)
transmit_response(p, "503 Unavailable", req);
else
- transmit_response_reliable(p, "503 Unavailable", req, 1);
+ transmit_response_reliable(p, "503 Unavailable", req);
ast_set_flag(p, SIP_ALREADYGONE);
/* Unlock locks so ast_hangup can do its magic */
ast_mutex_unlock(&p->lock);
@@ -10613,14 +10625,14 @@
if (ignore)
transmit_response(p, "488 Not Acceptable Here (codec error)", req);
else
- transmit_response_reliable(p, "488 Not Acceptable Here (codec error)", req, 1);
+ transmit_response_reliable(p, "488 Not Acceptable Here (codec error)", req);
ast_set_flag(p, SIP_NEEDDESTROY);
} else {
ast_log(LOG_NOTICE, "Unable to create/find channel\n");
if (ignore)
transmit_response(p, "503 Unavailable", req);
else
- transmit_response_reliable(p, "503 Unavailable", req, 1);
+ transmit_response_reliable(p, "503 Unavailable", req);
ast_set_flag(p, SIP_NEEDDESTROY);
}
}
@@ -10688,7 +10700,7 @@
transmit_notify_with_sipfrag(p, seqno);
/* Always increment on a BYE */
if (!nobye) {
- transmit_request_with_auth(p, SIP_BYE, 0, 1, 1);
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
ast_set_flag(p, SIP_ALREADYGONE);
}
}
@@ -10715,7 +10727,7 @@
ast_set_flag(p, SIP_NEEDDESTROY);
if (p->initreq.len > 0) {
if (!ignore)
- transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 1);
+ transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
transmit_response(p, "200 OK", req);
return 1;
} else {
@@ -10733,7 +10745,7 @@
char iabuf[INET_ADDRSTRLEN];
if (p->pendinginvite && !ast_test_flag(p, SIP_OUTGOING) && !ignore)
- transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 1);
+ transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
copy_request(&p->initreq, req);
check_via(p, req);
@@ -10834,7 +10846,7 @@
mailboxsize = sizeof(mailboxbuf);
}
/* Handle authentication if this is our first subscribe */
- res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, sin, ignore, mailbox, mailboxsize);
+ res = check_user_full(p, req, SIP_SUBSCRIBE, e, XMIT_UNRELIABLE, sin, ignore, mailbox, mailboxsize);
if (res) {
if (res < 0) {
ast_log(LOG_NOTICE, "Failed to authenticate user %s for SUBSCRIBE\n", get_header(req, "From"));
@@ -11125,7 +11137,7 @@
if (!p->initreq.headers && ast_test_flag(req, SIP_PKT_WITH_TOTAG)) {
/* If this is a first request and it got a to-tag, it is not for us */
if (!ignore && req->method == SIP_INVITE) {
- transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req, 1);
+ transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
/* Will cease to exist after ACK */
} else {
transmit_response(p, "481 Call/Transaction Does Not Exist", req);
@@ -13003,7 +13015,7 @@
}
ast_string_field_build(p, our_contact, "Transfer <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : "");
- transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq, 1);
+ transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq);
/* this is all that we want to send to that SIP device */
ast_set_flag(p, SIP_ALREADYGONE);
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