[svn-commits] oej: branch oej/codename-pineapple r47211 -
/team/oej/codename-pineapple/chan...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Nov 4 15:44:39 MST 2006
Author: oej
Date: Sat Nov 4 16:44:39 2006
New Revision: 47211
URL: http://svn.digium.com/view/asterisk?rev=47211&view=rev
Log:
Updates
Modified:
team/oej/codename-pineapple/channels/chan_sip3.c
Modified: team/oej/codename-pineapple/channels/chan_sip3.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/chan_sip3.c?rev=47211&r1=47210&r2=47211&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Sat Nov 4 16:44:39 2006
@@ -1348,7 +1348,7 @@
__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->state == DIALOG_STATE_TRYING) {
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
/* Do we need a timer here if we don't hear from them at all? */
} else {
@@ -1358,6 +1358,7 @@
INVITE, but do set an autodestruct just in case we never get it. */
needdestroy = 0;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
}
if ( p->initid != -1 ) {
/* channel still up - reverse dec of inUse counter
@@ -1370,6 +1371,7 @@
transmit_response_reliable(p, res, &p->initreq);
else
transmit_response_reliable(p, "603 Declined", &p->initreq);
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
}
} else { /* Call is in UP state, send BYE */
if (!p->pendinginvite) {
@@ -1393,6 +1395,7 @@
pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos);
if (p->vrtp && oldowner)
pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos);
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while we have "INVITE" outstanding. */
@@ -1435,24 +1438,25 @@
static int sip_answer(struct ast_channel *ast)
{
int res = 0;
- struct sip_dialog *p = ast->tech_pvt;
-
- ast_mutex_lock(&p->lock);
+ struct sip_dialog *dialog = ast->tech_pvt;
+
+ ast_mutex_lock(&dialog->lock);
if (ast->_state != AST_STATE_UP) {
- try_suggested_sip_codec(p);
+ try_suggested_sip_codec(dialog);
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
- if (p->t38.state == T38_PEER_DIRECT) {
- p->t38.state = T38_ENABLED;
+ if (dialog->t38.state == T38_PEER_DIRECT) {
+ dialog->t38.state = T38_ENABLED;
if (option_debug > 1)
- ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
- res = transmit_response_with_attachment(WITH_T38_SDP, p, "200 OK", &p->initreq, XMIT_CRITICAL);
+ ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", dialog->t38.state, ast->name);
+ res = transmit_response_with_attachment(WITH_T38_SDP, dialog, "200 OK", &dialog->initreq, XMIT_CRITICAL);
} else
- res = transmit_response_with_attachment(WITH_SDP, p, "200 OK", &p->initreq, XMIT_CRITICAL);
- }
- ast_mutex_unlock(&p->lock);
+ res = transmit_response_with_attachment(WITH_SDP, dialog, "200 OK", &dialog->initreq, XMIT_CRITICAL);
+ dialogstatechange(dialog, DIALOG_STATE_CONFIRMED);
+ }
+ ast_mutex_unlock(&dialog->lock);
return res;
}
@@ -2088,6 +2092,7 @@
static int transmit_response_with_unsupported(struct sip_dialog *p, const char *msg, const struct sip_request *req, const char *unsupported)
{
struct sip_request resp;
+
respprep(&resp, p, msg, req);
append_date(&resp);
add_header(&resp, "Unsupported", unsupported);
@@ -4563,6 +4568,14 @@
unsigned int event;
const char *c = get_header(req, "Content-Type");
+ if (ast_test_flag(req, SIP_PKT_IGNORE)) {
+ transmit_response(p, "200 OK", req);
+ return;
+ }
+
+ /* All replies to INFO are final */
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
+
/* Need to check the media/type */
if (!strcasecmp(c, "application/dtmf-relay") ||
!strcasecmp(c, "application/vnd.nortelnetworks.digits")) {
@@ -5180,6 +5193,7 @@
/* Then we AUTH */
ast_string_field_free(p, theirtag); /* forget their old tag, so we don't match tags when getting response */
+ //??? dialogstatechange(p, DIALOG_STATE_TERMINATED_AUTH);
if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
@@ -5187,6 +5201,7 @@
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
}
}
break;
@@ -5196,6 +5211,7 @@
ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", get_header(&p->initreq, "From"));
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
break;
@@ -5396,6 +5412,7 @@
if (p->subscribed == NONE)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
} else if (sipmethod == SIP_REGISTER)
res = handle_response_register(p, resp, rest, req, req->seqno);
else if (sipmethod == SIP_BYE) /* Ok, we're ready to go */
@@ -5466,6 +5483,7 @@
ast_log(LOG_WARNING, "INVITE with REPLACEs failed to '%s'\n", get_header(&p->initreq, "From"));
if (owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else if (sipmethod == SIP_REFER) {
/* A transfer with Replaces did not work */
@@ -5502,8 +5520,9 @@
handle_response_invite(p, resp, rest, req);
else if (sipmethod == SIP_REFER)
handle_response_refer(p, resp, rest, req);
- else
+ else {
ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_inet_ntoa(p->sa.sin_addr), msg);
+ }
break;
case 603: /* Declined transfer */
if (sipmethod == SIP_REFER) {
@@ -7081,6 +7100,7 @@
correct according to RFC 3261 */
/* Check if this a new request in a new dialog with a totag already attached to it,
RFC 3261 - section 12.2 - and we don't want to mess with recovery */
+
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 (!ast_test_flag(req, SIP_PKT_IGNORE) && req->method == SIP_INVITE) {
@@ -7090,6 +7110,7 @@
transmit_response(p, "481 Call/Transaction Does Not Exist", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
+ dialogstatechange(p, DIALOG_STATE_TERMINATED);
return res;
}
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