[svn-commits] dvossel: trunk r248397 - in /trunk: ./ channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Feb 23 10:34:43 CST 2010
Author: dvossel
Date: Tue Feb 23 10:34:39 2010
New Revision: 248397
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=248397
Log:
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:
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://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=248397&r1=248396&r2=248397
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Feb 23 10:34:39 2010
@@ -18816,6 +18816,7 @@
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 */
@@ -18856,7 +18857,8 @@
p->invitestate = INV_COMPLETED;
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -18883,7 +18885,8 @@
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
@@ -18909,7 +18912,8 @@
*/
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;
}
}
@@ -18947,7 +18951,8 @@
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;
}
}
@@ -18964,7 +18969,8 @@
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)
@@ -18977,7 +18983,8 @@
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)
@@ -19040,6 +19047,8 @@
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 .
@@ -19077,8 +19086,10 @@
ast_channel_unlock(p->refer->refer_call->owner);
}
}
+ refer_locked = 0;
p->invitestate = INV_COMPLETED;
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -19121,7 +19132,8 @@
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 {
@@ -19142,7 +19154,8 @@
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) {
@@ -19155,7 +19168,8 @@
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 */
@@ -19172,7 +19186,8 @@
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 (p->rtp) {
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
@@ -19205,7 +19220,8 @@
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 */
change_redirecting_information(p, req, &redirecting, FALSE); /*Will return immediately if no Diversion header is present */
@@ -19232,7 +19248,8 @@
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 */
@@ -19292,7 +19309,8 @@
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -19306,7 +19324,8 @@
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -19320,7 +19339,8 @@
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
p->stimer->st_active_peer_ua = TRUE;
@@ -19350,7 +19370,8 @@
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
break;
@@ -19429,12 +19450,15 @@
ast_hangup(c);
sip_pvt_lock(p); /* pvt is expected to remain locked on return, so re-lock it */
- 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;
}
}
@@ -19558,6 +19582,16 @@
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;
}
More information about the svn-commits
mailing list