[svn-commits] oej: branch oej/invitestate-1.4 r48212 -
/team/oej/invitestate-1.4/channels/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Dec 2 12:05:50 MST 2006
Author: oej
Date: Sat Dec 2 13:05:49 2006
New Revision: 48212
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48212
Log:
- Adding invitestate changes on control messages
- Don't set ALREADYGONE twice on responses, might affect communication
Modified:
team/oej/invitestate-1.4/channels/chan_sip.c
Modified: team/oej/invitestate-1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/invitestate-1.4/channels/chan_sip.c?view=diff&rev=48212&r1=48211&r2=48212
==============================================================================
--- team/oej/invitestate-1.4/channels/chan_sip.c (original)
+++ team/oej/invitestate-1.4/channels/chan_sip.c Sat Dec 2 13:05:49 2006
@@ -250,7 +250,7 @@
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_PROCEEDING, /*!< We got/sent 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) */
@@ -1609,6 +1609,13 @@
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
+static void sip_alreadygone(struct sip_pvt *dialog)
+{
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
+ ast_set_flag(&dialog->flags[0], SIP_ALREADYGONE);
+}
+
/*! \brief returns true if 'name' (with optional trailing whitespace)
* matches the sip method 'id'.
@@ -1887,7 +1894,7 @@
ast_mutex_lock(&pkt->owner->lock);
}
if (pkt->owner->owner) {
- ast_set_flag(&pkt->owner->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(pkt->owner);
ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
ast_queue_hangup(pkt->owner->owner);
ast_channel_unlock(pkt->owner->owner);
@@ -3626,6 +3633,7 @@
switch(condition) {
case AST_CONTROL_RINGING:
if (ast->_state == AST_STATE_RING) {
+ p->invitestate = INV_EARLY_MEDIA;
if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
(ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {
/* Send 180 ringing if out-of-band seems reasonable */
@@ -3642,7 +3650,8 @@
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "486 Busy Here", &p->initreq);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_TERMINATED;
+ sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
@@ -3651,7 +3660,8 @@
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "503 Service Unavailable", &p->initreq);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_TERMINATED;
+ sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
@@ -3662,6 +3672,7 @@
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
transmit_response(p, "100 Trying", &p->initreq);
+ p->invitestate = INV_PROCEEDING;
break;
}
res = -1;
@@ -3670,6 +3681,7 @@
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
break;
@@ -11702,7 +11714,7 @@
if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, authenticate, authorization, SIP_INVITE, 1)) {
ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
}
@@ -11716,14 +11728,14 @@
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
break;
case 404: /* Not found */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
break;
case 481: /* Call leg does not exist */
@@ -12180,7 +12192,6 @@
/* 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(p->sa.sin_addr));
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
@@ -12239,7 +12250,7 @@
/* ACK on invite */
if (sipmethod == SIP_INVITE)
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
if (!p->owner)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
} else if ((resp >= 100) && (resp < 200)) {
@@ -13330,7 +13341,7 @@
transmit_response(p, "503 Unavailable", req); /* OEJ - Right answer? */
else
transmit_response_reliable(p, "503 Unavailable", req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
/* Unlock locks so ast_hangup can do its magic */
ast_mutex_unlock(&p->lock);
c->hangupcause = AST_CAUSE_CALL_REJECTED;
@@ -13654,7 +13665,7 @@
transmit_response(p, "603 Declined (No dialog)", req);
if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
append_history(p, "Xfer", "Refer failed. Outside of dialog.");
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
return 0;
@@ -13913,7 +13924,7 @@
{
check_via(p, req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
@@ -13954,7 +13965,7 @@
copy_request(&p->initreq, req);
check_via(p, req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
/* Get RTCP quality before end of call */
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
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