[asterisk-commits] mmichelson: branch 10 r352015 - in /branches/10: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 20 18:06:28 CST 2012
Author: mmichelson
Date: Fri Jan 20 18:06:21 2012
New Revision: 352015
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=352015
Log:
Fix RTP reference leak.
If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.
This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.
(issue ASTERISK-19192)
Review: https://reviewboard.asterisk.org/r/1681/
........
Merged revisions 352014 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Modified:
branches/10/ (props changed)
branches/10/channels/chan_sip.c
Propchange: branches/10/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Modified: branches/10/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/10/channels/chan_sip.c?view=diff&rev=352015&r1=352014&r2=352015
==============================================================================
--- branches/10/channels/chan_sip.c (original)
+++ branches/10/channels/chan_sip.c Fri Jan 20 18:06:21 2012
@@ -3882,6 +3882,7 @@
ast_channel_unref(owner);
} else if (p->refer && !p->alreadygone) {
ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
+ stop_media_flows(p);
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -20769,15 +20770,22 @@
case 200: /* Notify accepted */
/* They got the notify, this is the end */
if (p->owner) {
- if (!p->refer) {
+ if (p->refer) {
+ ast_log(LOG_NOTICE, "Got OK on REFER Notify message\n");
+ } else {
ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
- ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
- } else {
- ast_debug(4, "Got OK on REFER Notify message\n");
+ /*
+ * XXX There is discrepancy on whether a hangup should be queued
+ * or not. This code used to be duplicated in two places, and the more
+ * frequently hit area had this disabled, making it the de facto
+ * "correct" way to go.
+ *
+ * ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
+ */
}
} else {
- if (p->subscribed == NONE) {
- ast_debug(4, "Got 200 accepted on NOTIFY\n");
+ if (p->subscribed == NONE && !p->refer) {
+ ast_debug(4, "Got 200 accepted on NOTIFY %s\n", p->callid);
pvt_set_needdestroy(p, "received 200 response");
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
@@ -20801,6 +20809,9 @@
ast_log(LOG_NOTICE, "Failed to authenticate on NOTIFY to '%s'\n", sip_get_header(&p->initreq, "From"));
pvt_set_needdestroy(p, "failed to authenticate NOTIFY");
}
+ break;
+ case 481: /* Call leg does not exist */
+ pvt_set_needdestroy(p, "Received 481 response for NOTIFY");
break;
}
}
@@ -21444,6 +21455,9 @@
} else if (sipmethod == SIP_MESSAGE) {
/* More good gravy! */
handle_response_message(p, resp, rest, req, seqno);
+ } else if (sipmethod == SIP_NOTIFY) {
+ /* The gravy train continues to roll */
+ handle_response_notify(p, resp, rest, req, seqno);
} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
switch(resp) {
case 100: /* 100 Trying */
@@ -21459,8 +21473,6 @@
p->authtries = 0; /* Reset authentication counter */
if (sipmethod == SIP_INVITE) {
handle_response_invite(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_NOTIFY) {
- handle_response_notify(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_REGISTER) {
handle_response_register(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_SUBSCRIBE) {
@@ -21475,8 +21487,6 @@
case 407: /* Proxy auth required */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
- else if (sipmethod == SIP_NOTIFY)
- handle_response_notify(p, resp, rest, req, seqno);
else if (sipmethod == SIP_SUBSCRIBE)
handle_response_subscribe(p, resp, rest, req, seqno);
else if (p->registry && sipmethod == SIP_REGISTER)
@@ -21552,8 +21562,6 @@
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_SUBSCRIBE) {
handle_response_subscribe(p, resp, rest, req, seqno);
- } else if (sipmethod == SIP_NOTIFY) {
- pvt_set_needdestroy(p, "received 481 response");
} else if (sipmethod == SIP_BYE) {
/* The other side has no transaction to bye,
just assume it's all right then */
@@ -21714,24 +21722,6 @@
ast_debug(1, "Got 200 OK on CANCEL\n");
/* Wait for 487, then destroy */
- } else if (sipmethod == SIP_NOTIFY) {
- /* They got the notify, this is the end */
- if (p->owner) {
- if (p->refer) {
- ast_debug(1, "Got 200 OK on NOTIFY for transfer\n");
- } else
- ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
- /* ast_queue_hangup(p->owner); Disabled */
- } else {
- if (!p->subscribed && !p->refer) {
- pvt_set_needdestroy(p, "transaction completed");
- }
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
- /* Ready to send the next state we have on queue */
- ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
- cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
- }
- }
} else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "transaction completed");
}
@@ -21752,8 +21742,6 @@
/* Re-invite failed */
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) {
- pvt_set_needdestroy(p, "received 481 response");
- } else if (sipmethod == SIP_NOTIFY) {
pvt_set_needdestroy(p, "received 481 response");
} else if (sipdebug) {
ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
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