[asterisk-commits] dvossel: branch 1.6.2 r248398 - in /branches/1.6.2: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Feb 23 10:37:52 CST 2010
Author: dvossel
Date: Tue Feb 23 10:37:48 2010
New Revision: 248398
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=248398
Log:
Merged revisions 248397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines
Merged revisions 248396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines
fixes invite with replaces deadlock
(closes issue #16862)
Reported by: pwalker
Patches:
replaces_deadlock_1.4 uploaded by dvossel (license 671)
Tested by: pwalker, dvossel
........
................
Modified:
branches/1.6.2/ (props changed)
branches/1.6.2/channels/chan_sip.c
Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=248398&r1=248397&r2=248398
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Tue Feb 23 10:37:48 2010
@@ -19641,6 +19641,7 @@
int gotdest;
const char *p_replaces;
char *replace_id = NULL;
+ int refer_locked = 0;
const char *required;
unsigned int required_profile = 0;
struct ast_channel *c = NULL; /* New channel */
@@ -19681,7 +19682,8 @@
p->invitestate = INV_COMPLETED;
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -19708,7 +19710,8 @@
transmit_response(p, "482 Loop Detected", req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
} else {
/*! This is a spiral. What we need to do is to just change the outgoing INVITE
* so that it now routes to the new Request URI. Since we created the INVITE ourselves
@@ -19734,7 +19737,8 @@
*/
ast_string_field_set(p->owner, call_forward, peerorhost);
ast_queue_control(p->owner, AST_CONTROL_BUSY);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
}
@@ -19772,7 +19776,8 @@
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;
+ res = 0;
+ goto request_invite_cleanup;
}
}
@@ -19789,7 +19794,8 @@
ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
/* Do not destroy existing call */
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
if (sipdebug)
@@ -19803,7 +19809,8 @@
append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
p->invitestate = INV_COMPLETED;
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
/* Todo: (When we find phones that support this)
@@ -19862,6 +19869,8 @@
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
+ } else {
+ refer_locked = 1;
}
/* The matched call is the call from the transferer to Asterisk .
@@ -19899,8 +19908,10 @@
ast_channel_unlock(p->refer->refer_call->owner);
}
}
+ refer_locked = 0;
p->invitestate = INV_COMPLETED;
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -19933,7 +19944,8 @@
transmit_response_reliable(p, "488 Not acceptable here", req);
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
} else {
@@ -19952,7 +19964,8 @@
res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
if (res == AUTH_CHALLENGE_SENT) {
p->invitestate = INV_COMPLETED; /* Needs to restart in another INVITE transaction */
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
if (res < 0) { /* Something failed in authentication */
if (res == AUTH_FAKE_AUTH) {
@@ -19965,7 +19978,8 @@
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_string_field_set(p, theirtag, NULL);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
/* If T38 is needed but not present, then make it magically appear */
@@ -19982,7 +19996,8 @@
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_debug(1, "No compatible codecs for this SIP call.\n");
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
if (p->rtp) {
@@ -20017,7 +20032,8 @@
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
p->invitestate = INV_COMPLETED;
}
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
}
gotdest = get_destination(p, NULL); /* Get destination right away */
get_rdnis(p, NULL); /* Get redirect information */
@@ -20044,7 +20060,8 @@
p->invitestate = INV_COMPLETED;
update_call_counter(p, DEC_CALL_LIMIT);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
} else {
/* If no extension was specified, use the s one */
@@ -20098,7 +20115,8 @@
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -20112,7 +20130,8 @@
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -20126,7 +20145,8 @@
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
p->stimer->st_active_peer_ua = TRUE;
@@ -20156,7 +20176,8 @@
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
break;
@@ -20230,12 +20251,15 @@
ast_hangup(c);
sip_pvt_lock(p);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
} else {
/* Go and take over the target call */
if (sipdebug)
ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid);
- return handle_invite_replaces(p, req, debug, seqno, sin, nounlock);
+ res = handle_invite_replaces(p, req, debug, seqno, sin, nounlock);
+ refer_locked = 0;
+ goto request_invite_cleanup;
}
}
@@ -20360,6 +20384,16 @@
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
+
+request_invite_cleanup:
+
+ if (refer_locked && p->refer && p->refer->refer_call) {
+ sip_pvt_unlock(p->refer->refer_call);
+ if (p->refer->refer_call->owner) {
+ ast_channel_unlock(p->refer->refer_call->owner);
+ }
+ }
+
return res;
}
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