[svn-commits] oej: trunk r126517 - in /trunk: ./ channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jun 30 08:03:54 CDT 2008
Author: oej
Date: Mon Jun 30 08:03:53 2008
New Revision: 126517
URL: http://svn.digium.com/view/asterisk?view=rev&rev=126517
Log:
The following patch with some changes for trunk...
Merged revisions 126516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines
Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and
also fail if we don't get the very same precious ACK. Based on patch by tsearle, with
my own additions.
(closes issue #12951)
Reported by: tsearle
Patches:
busy_retransmit.patch uploaded by tsearle (license 373)
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=126517&r1=126516&r2=126517
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Jun 30 08:03:53 2008
@@ -5441,7 +5441,7 @@
break;
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
- transmit_response(p, "486 Busy Here", &p->initreq);
+ transmit_response_reliable(p, "486 Busy Here", &p->initreq);
p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
@@ -5451,7 +5451,7 @@
break;
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
- transmit_response(p, "503 Service Unavailable", &p->initreq);
+ transmit_response_reliable(p, "503 Service Unavailable", &p->initreq);
p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
@@ -8026,7 +8026,7 @@
*/
static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req)
{
- return __transmit_response(p, msg, req, XMIT_CRITICAL);
+ return __transmit_response(p, msg, req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL);
}
/*! \brief Append date to SIP message */
@@ -17521,7 +17521,7 @@
being able to call yourself */
/* If pedantic is on, we need to check the tags. If they're different, this is
in fact a forked call through a SIP proxy somewhere. */
- transmit_response(p, "482 Loop Detected", req);
+ transmit_response_reliable(p, "482 Loop Detected", req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
@@ -17529,7 +17529,7 @@
if (!req->ignore && p->pendinginvite) {
/* We already have a pending invite. Sorry. You are on hold. */
- transmit_response(p, "491 Request Pending", req);
+ transmit_response_reliable(p, "491 Request Pending", req);
ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
/* Don't destroy dialog here */
return 0;
@@ -17546,7 +17546,7 @@
if (p->owner) {
ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
- transmit_response(p, "400 Bad request", req); /* The best way to not not accept the transfer */
+ transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
/* Do not destroy existing call */
return -1;
}
@@ -17558,7 +17558,7 @@
ast_uri_decode(replace_id);
if (!p->refer && !sip_refer_allocate(p)) {
- transmit_response(p, "500 Server Internal Error", req);
+ transmit_response_reliable(p, "500 Server Internal Error", req);
append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
p->invitestate = INV_COMPLETED;
@@ -17596,7 +17596,7 @@
*/
if ((p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
- transmit_response(p, "481 Call Leg Does Not Exist (Replaces)", req);
+ transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
}
@@ -17609,7 +17609,7 @@
if (p->refer->refer_call == p) {
ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
- transmit_response(p, "400 Bad request", req); /* The best way to not not accept the transfer */
+ transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
error = 1;
}
@@ -17617,13 +17617,13 @@
/* Oops, someting wrong anyway, no owner, no call */
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
/* Check for better return code */
- transmit_response(p, "481 Call Leg Does Not Exist (Replace)", req);
+ transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
error = 1;
}
if (!error && p->refer->refer_call->owner->_state != AST_STATE_RINGING && p->refer->refer_call->owner->_state != AST_STATE_RING && p->refer->refer_call->owner->_state != AST_STATE_UP ) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
- transmit_response(p, "603 Declined (Replaces)", req);
+ transmit_response_reliable(p, "603 Declined (Replaces)", req);
error = 1;
}
@@ -17666,7 +17666,7 @@
/* Handle SDP here if we already have an owner */
if (find_sdp(req)) {
if (process_sdp(p, req, SDP_T38_INITIATE)) {
- transmit_response(p, "488 Not acceptable here", req);
+ transmit_response_reliable(p, "488 Not acceptable here", req);
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
@@ -17812,7 +17812,7 @@
if (!ast_strlen_zero(p_uac_se_hdr)) {
rtn = parse_session_expires(p_uac_se_hdr, &uac_max_se, &st_ref);
if (rtn != 0) {
- transmit_response(p, "400 Session-Expires Invalid Syntax", req);
+ transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -17826,7 +17826,7 @@
if (!ast_strlen_zero(p_uac_min_se)) {
rtn = parse_minse(p_uac_min_se, &uac_min_se);
if (rtn != 0) {
- transmit_response(p, "400 Min-SE Invalid Syntax", req);
+ transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -17954,18 +17954,12 @@
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
p->invitestate = INV_COMPLETED;
- if (req->ignore)
- transmit_response(p, "503 Unavailable", req);
- else
- transmit_response_reliable(p, "503 Unavailable", req);
+ 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");
p->invitestate = INV_COMPLETED;
- if (req->ignore)
- transmit_response(p, "480 Temporarily Unavailable", req);
- else
- transmit_response_reliable(p, "480 Temporarily Unavailable", req);
+ transmit_response_reliable(p, "480 Temporarily Unavailable", req);
break;
case AST_PBX_SUCCESS:
/* nothing to do */
@@ -17986,10 +17980,7 @@
*nounlock = 1;
if (ast_pickup_call(c)) {
ast_log(LOG_NOTICE, "Nothing to pick up for %s\n", p->callid);
- if (req->ignore)
- transmit_response(p, "503 Unavailable", req); /* OEJ - Right answer? */
- else
- transmit_response_reliable(p, "503 Unavailable", req);
+ transmit_response_reliable(p, "503 Unavailable", req);
sip_alreadygone(p);
/* Unlock locks so ast_hangup can do its magic */
sip_pvt_unlock(p);
@@ -18036,10 +18027,7 @@
sip_pvt_lock(bridgepvt);
change_t38_state(bridgepvt, T38_DISABLED);
sip_pvt_unlock(bridgepvt);
- if (req->ignore)
- transmit_response(p, "488 Not acceptable here", req);
- else
- transmit_response_reliable(p, "488 Not acceptable here", req);
+ transmit_response_reliable(p, "488 Not acceptable here", req);
}
} else {
@@ -18049,10 +18037,7 @@
}
} else {
/* Other side is not a SIP channel */
- if (req->ignore)
- transmit_response(p, "488 Not acceptable here", req);
- else
- transmit_response_reliable(p, "488 Not acceptable here", req);
+ transmit_response_reliable(p, "488 Not acceptable here", req);
change_t38_state(p, T38_DISABLED);
if (!p->lastinvite) /* Only destroy if this is *not* a re-invite */
@@ -18077,10 +18062,7 @@
if (bridgepvt->t38.state == T38_ENABLED) {
ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n");
/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
- if (req->ignore)
- transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
- else
- transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
+ transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
sendok = FALSE;
}
/* No bridged peer with T38 enabled*/
@@ -18109,10 +18091,7 @@
ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
msg = "503 Unavailable";
}
- if (req->ignore)
- transmit_response(p, msg, req);
- else
- transmit_response_reliable(p, msg, req);
+ transmit_response_reliable(p, msg, req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
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