[svn-commits] dvossel: branch 1.4 r248396 - /branches/1.4/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Feb 23 10:26:11 CST 2010
Author: dvossel
Date: Tue Feb 23 10:26:05 2010
New Revision: 248396
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=248396
Log:
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/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=248396&r1=248395&r2=248396
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Feb 23 10:26:05 2010
@@ -14763,6 +14763,7 @@
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 */
@@ -14786,7 +14787,8 @@
p->invitestate = INV_COMPLETED;
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -14806,7 +14808,8 @@
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
@@ -14831,7 +14834,8 @@
*/
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;
}
}
@@ -14870,7 +14874,8 @@
if (option_debug)
ast_log(LOG_DEBUG, "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;
}
}
@@ -14888,7 +14893,8 @@
ast_log(LOG_DEBUG, "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 && option_debug > 2)
@@ -14902,7 +14908,8 @@
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)
@@ -14938,6 +14945,8 @@
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;
}
/* At this point, bot the pvt and the owner of the call to be replaced is locked */
@@ -14977,8 +14986,10 @@
ast_channel_unlock(p->refer->refer_call->owner);
}
}
+ refer_locked = 0;
p->invitestate = INV_COMPLETED;
- return -1;
+ res = -1;
+ goto request_invite_cleanup;
}
}
@@ -15010,7 +15021,8 @@
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 {
@@ -15040,7 +15052,8 @@
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) {
@@ -15053,7 +15066,9 @@
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_string_field_free(p, theirtag);
- return 0;
+ res = 0;
+ goto request_invite_cleanup;
+
}
/* We have a succesful authentication, process the SDP portion if there is one */
@@ -15065,7 +15080,8 @@
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
if (option_debug)
ast_log(LOG_DEBUG, "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) {
@@ -15102,7 +15118,8 @@
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 */
@@ -15129,7 +15146,8 @@
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 */
/* Basically for calling to IP/Host name only */
@@ -15169,7 +15187,10 @@
/* Go and take over the target call */
if (sipdebug && option_debug > 3)
ast_log(LOG_DEBUG, "Sending this call to the invite/replcaes handler %s\n", p->callid);
- return handle_invite_replaces(p, req, debug, ast_test_flag(req, SIP_PKT_IGNORE), seqno, sin, nounlock);
+
+ res = handle_invite_replaces(p, req, debug, ast_test_flag(req, SIP_PKT_IGNORE), seqno, sin, nounlock);
+ refer_locked = 0;
+ goto request_invite_cleanup;
}
@@ -15356,6 +15377,17 @@
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
+ return res;
+
+request_invite_cleanup:
+
+ if (refer_locked && p->refer && p->refer->refer_call) {
+ ast_mutex_unlock(&p->refer->refer_call->lock);
+ if (p->refer->refer_call->owner) {
+ ast_channel_unlock(p->refer->refer_call->owner);
+ }
+ }
+
return res;
}
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