[svn-commits] rizzo: branch rizzo/astobj2 r47825 -
/team/rizzo/astobj2/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Nov 18 12:29:12 MST 2006
Author: rizzo
Date: Sat Nov 18 13:29:11 2006
New Revision: 47825
URL: http://svn.digium.com/view/asterisk?view=rev&rev=47825
Log:
sync with version 47824
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=47825&r1=47824&r2=47825
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Sat Nov 18 13:29:11 2006
@@ -239,6 +239,21 @@
AST_FAILURE = -1,
};
+/*! \brief States for the INVITE transaction, not the dialog
+ \note this is for the INVITE that sets up the dialog
+*/
+enum invitestates {
+ INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
+ INV_CALLING, /*!< Invite sent, no answer */
+ INV_PROCEEDING, /*!< We got 1xx message */
+ INV_EARLY_MEDIA, /*!< We got 18x message with to-tag back */
+ INV_COMPLETED, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
+ INV_CONFIRMED, /*!< Confirmed response - we've got an ack (Incoming calls only) */
+ INV_TERMINATED, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
+ The only way out of this is a BYE from one side */
+ INV_CANCELLED /*!< Transaction cancelled by client or server in non-terminated state */
+};
+
/* Do _NOT_ make any changes to this enum, or the array following it;
if you think you are doing the right thing, you are probably
not doing the right thing. If you think there are changes
@@ -516,6 +531,7 @@
static int global_limitonpeers; /*!< Match call limit on peers only */
static int global_rtautoclear;
static int global_notifyringing; /*!< Send notifications on ringing */
+static int global_notifyhold; /*!< Send notifications on hold */
static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
static int global_srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
static int pedanticsipchecking; /*!< Extra checking ? Default off */
@@ -709,7 +725,7 @@
#define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
#define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
#define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */
-#define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
+#define SIP_FREE_BIT (1 << 14) /*!< ---- */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
#define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
#define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
@@ -886,6 +902,7 @@
struct sip_pvt *next; /*!< Next dialog in chain */
ast_mutex_t pvt_lock; /*!< Dialog private lock */
#endif
+ enum invitestates invitestate; /*!< Track state of SIP_INVITEs */
int method; /*!< SIP method that opened this dialog */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global CallID */
@@ -2127,9 +2144,6 @@
/* Reset schedule ID */
p->autokillid = -1;
- if (option_debug)
- ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid);
- append_history(p, "AutoDestroy", "%s", p->callid);
if (p->owner) {
ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
ast_queue_hangup(p->owner);
@@ -3039,6 +3053,8 @@
if (option_debug > 1)
ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
transmit_invite(p, SIP_INVITE, 1, 2);
+
+ p->invitestate = INV_CALLING;
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, pvt_ref(p));
@@ -3579,12 +3595,13 @@
__sip_pretend_ack(p);
/* if we can't send right now, mark it pending */
- if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE)) {
+ if (p->invitestate == INV_CALLING) {
+ /* We can't send anything in CALLING state */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
/* Do we need a timer here if we don't hear from them at all? */
} else {
/* Send a new request: CANCEL */
- transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
+ transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
/* 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. */
needdestroy = 0;
@@ -5464,7 +5481,8 @@
"Uniqueid: %s\r\n",
p->owner->name,
p->owner->uniqueid);
- sip_peer_hold(p, 0);
+ if (global_notifyhold)
+ sip_peer_hold(p, 0);
}
ast_clear_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */
} else if (!sin.sin_addr.s_addr || sendonly ) {
@@ -7697,6 +7715,9 @@
static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
{
struct sip_request resp;
+
+ if (sipmethod == SIP_ACK)
+ p->invitestate = INV_CONFIRMED;
reqprep(&resp, p, sipmethod, seqno, newbranch);
add_header_contentLength(&resp, 0);
@@ -10668,6 +10689,7 @@
ast_cli(fd, " Outbound reg. timeout: %d secs\n", global_reg_timeout);
ast_cli(fd, " Outbound reg. attempts: %d\n", global_regattempts_max);
ast_cli(fd, " Notify ringing state: %s\n", global_notifyringing ? "Yes" : "No");
+ ast_cli(fd, " Notify hold state: %s\n", global_notifyhold ? "Yes" : "No");
ast_cli(fd, " SIP Transfer mode: %s\n", transfermode2str(global_allowtransfer));
ast_cli(fd, " Max Call Bitrate: %d kbps\r\n", default_maxcallbitrate);
ast_cli(fd, " Auto-Framing: %s \r\n", global_autoframing ? "Yes" : "No");
@@ -11990,12 +12012,12 @@
{
if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
/* if we can't BYE, then this is really a pending CANCEL */
- if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE))
- transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0);
+ if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)
+ transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
/* 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. */
else
- transmit_request_with_auth(p, SIP_BYE, 0, 1, 1);
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
@@ -12042,6 +12064,15 @@
if (resp > 100 && resp < 200 && resp != 180 && resp != 183)
resp = 183;
+ /* Any response between 100 and 199 is PROCEEDING */
+ if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
+ p->invitestate = INV_PROCEEDING;
+
+ /* Final response, not 200 ? */
+ if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
+ p->invitestate = INV_COMPLETED;
+
+
switch (resp) {
case 100: /* Trying */
if (!req_ignore(req))
@@ -12059,13 +12090,13 @@
}
}
if (find_sdp(req)) {
+ p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!req_ignore(req) && p->owner) {
/* Queue a progress frame only if we have SDP in 180 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -12074,13 +12105,13 @@
sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */
if (find_sdp(req)) {
+ p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!req_ignore(req) && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -12125,8 +12156,8 @@
ast_log(LOG_WARNING, "Ooooh.. no tech! That's REALLY bad\n");
break;
}
- if (!strcasecmp(bridgepeer->tech->type,"SIP")) {
- bridgepvt = bridgepeer->tech_pvt;
+ if (bridgepeer->tech == &sip_tech) {
+ bridgepvt = (struct sip_pvt*)(bridgepeer->tech_pvt);
if (bridgepvt->udptl) {
if (p->t38.state == T38_PEER_REINVITE) {
sip_handle_t38_reinvite(bridgepeer, p, 0);
@@ -12178,8 +12209,8 @@
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a non-200 ACK only */
+ p->invitestate = INV_TERMINATED;
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -13808,6 +13839,7 @@
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s: New call is still down.... Trying... \n", c->name);
transmit_response(p, "100 Trying", req);
+ p->invitestate = INV_PROCEEDING;
ast_setstate(c, AST_STATE_RING);
if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
enum ast_pbx_result res;
@@ -13817,6 +13849,7 @@
switch(res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+ p->invitestate = INV_COMPLETED;
if (req_ignore(req))
transmit_response(p, "503 Unavailable", req);
else
@@ -13824,6 +13857,7 @@
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+ p->invitestate = INV_COMPLETED;
if (req_ignore(req))
transmit_response(p, "480 Temporarily Unavailable", req);
else
@@ -13860,6 +13894,7 @@
ast_setstate(c, AST_STATE_DOWN);
c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
}
+ p->invitestate = INV_COMPLETED;
ast_hangup(c);
sip_pvt_lock(p);
c = NULL;
@@ -13867,9 +13902,11 @@
break;
case AST_STATE_RING:
transmit_response(p, "100 Trying", req);
+ p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_RINGING:
transmit_response(p, "180 Ringing", req);
+ p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_UP:
if (option_debug > 1)
@@ -13882,8 +13919,8 @@
if ((bridgepeer = ast_bridged_channel(p->owner))) {
/* We have a bridge, and this is re-invite to switchover to T38 so we send re-invite with T38 SDP, to other side of bridge*/
/*! XXX: we should also check here does the other side supports t38 at all !!! XXX */
- if (!strcasecmp(bridgepeer->tech->type, "SIP")) { /* If we are bridged to SIP channel */
- bridgepvt = bridgepeer->tech_pvt;
+ if (bridgepeer->tech == &sip_tech) {
+ bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt;
if (bridgepvt->t38.state == T38_DISABLED) {
if (bridgepvt->udptl) { /* If everything is OK with other side's udptl struct */
/* Send re-invite to the bridged channel */
@@ -13935,8 +13972,8 @@
struct ast_channel *bridgepeer = NULL;
struct sip_pvt *bridgepvt = NULL;
if ((bridgepeer = ast_bridged_channel(p->owner))) {
- if (!strcasecmp(bridgepeer->tech->type, sip_tech.type)) {
- bridgepvt = bridgepeer->tech_pvt;
+ if (bridgepeer->tech == &sip_tech) {
+ bridgepvt = (struct sip_pvt*)bridgepeer->tech_pvt;
/* Does the bridged peer have T38 ? */
if (bridgepvt->t38.state == T38_ENABLED) {
ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n");
@@ -13954,6 +13991,7 @@
transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
}
+ p->invitestate = INV_TERMINATED;
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
@@ -13974,6 +14012,7 @@
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
@@ -14431,6 +14470,7 @@
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
@@ -14465,6 +14505,8 @@
/* If we have an INCOMING invite that we haven't answered, terminate that transaction */
if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !req_ignore(req) && !p->owner)
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
+
+ p->invitestate = INV_TERMINATED;
copy_request(&p->initreq, req);
@@ -15071,6 +15113,7 @@
case SIP_ACK:
/* Make sure we don't ignore this */
if (seqno == p->pendinginvite) {
+ p->invitestate = INV_CONFIRMED;
p->pendinginvite = 0;
__sip_ack(p, seqno, FLAG_RESPONSE, 0);
if (find_sdp(req)) {
@@ -16507,6 +16550,7 @@
expiry = DEFAULT_EXPIRY;
global_notifyringing = DEFAULT_NOTIFYRINGING;
global_limitonpeers = FALSE; /*!< Match call limit on peers only */
+ global_notifyhold = FALSE; /*!< Keep track of hold status for a peer */
global_alwaysauthreject = 0;
global_allowsubscribe = FALSE;
ast_copy_string(global_useragent, DEFAULT_USERAGENT, sizeof(global_useragent));
@@ -16600,6 +16644,7 @@
M_STR("notifymimetype", default_notifymime)
M_BOOL("notifyringing", global_notifyringing)
M_BOOL("limitpeersonly", global_limitonpeers)
+ M_BOOL("notifyhold", global_notifyhold)
M_BOOL("alwaysauthreject", global_alwaysauthreject)
M_STR("mohinterpret", default_mohinterpret)
M_STR("musicclass", default_mohinterpret)
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